A Bit Stream Scalable Speech/Audio Coder Combining Enhanced Regular Pulse Excitation and Parametric Coding

Directory of Open Access Journals (Sweden)

Albertus C. den Brinker

2007-01-01

Full Text Available This paper introduces a new audio and speech broadband coding technique based on the combination of a pulse excitation coder and a standardized parametric coder, namely, MPEG-4 high-quality parametric coder. After presenting a series of enhancements to regular pulse excitation (RPE to make it suitable for the modeling of broadband signals, it is shown how pulse and parametric codings complement each other and how they can be merged to yield a layered bit stream scalable coder able to operate at different points in the quality bit rate plane. The performance of the proposed coder is evaluated in a listening test. The major result is that the extra functionality of the bit stream scalability does not come at the price of a reduced performance since the coder is competitive with standardized coders (MP3, AAC, SSC.

A Bit Stream Scalable Speech/Audio Coder Combining Enhanced Regular Pulse Excitation and Parametric Coding

Science.gov (United States)

Riera-Palou, Felip; den Brinker, Albertus C.

2007-12-01

This paper introduces a new audio and speech broadband coding technique based on the combination of a pulse excitation coder and a standardized parametric coder, namely, MPEG-4 high-quality parametric coder. After presenting a series of enhancements to regular pulse excitation (RPE) to make it suitable for the modeling of broadband signals, it is shown how pulse and parametric codings complement each other and how they can be merged to yield a layered bit stream scalable coder able to operate at different points in the quality bit rate plane. The performance of the proposed coder is evaluated in a listening test. The major result is that the extra functionality of the bit stream scalability does not come at the price of a reduced performance since the coder is competitive with standardized coders (MP3, AAC, SSC).

Tune in the Net with RealAudio.

Science.gov (United States)

Buchanan, Larry

1997-01-01

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)

Estudi i implementació del protocol de streaming http live streaming per un client i-phone

OpenAIRE

Núñez Vera, Jordi

2013-01-01

[ANGLÈS] The aim of this project is, on the one hand, the analysis of Apple's HTTP Live Streaming protocol, which is an adaptative video and audio streaming protocol able to change the streams' bit rate according to the capacity of the media through which it is being transmitted. On the other hand, the project shows a client development of this protocol for the iPhone mobile device describing this platform from scratch. I trace here the necessary steps for developing applications on iOS and I...

Real-Time Transmission and Storage of Video, Audio, and Health Data in Emergency and Home Care Situations

Directory of Open Access Journals (Sweden)

Riccardo Stagnaro

2007-01-01

Full Text Available The increase in the availability of bandwidth for wireless links, network integration, and the computational power on fixed and mobile platforms at affordable costs allows nowadays for the handling of audio and video data, their quality making them suitable for medical application. These information streams can support both continuous monitoring and emergency situations. According to this scenario, the authors have developed and implemented the mobile communication system which is described in this paper. The system is based on ITU-T H.323 multimedia terminal recommendation, suitable for real-time data/video/audio and telemedical applications. The audio and video codecs, respectively, H.264 and G723.1, were implemented and optimized in order to obtain high performance on the system target processors. Offline media streaming storage and retrieval functionalities were supported by integrating a relational database in the hospital central system. The system is based on low-cost consumer technologies such as general packet radio service (GPRS and wireless local area network (WLAN or WiFi for lowband data/video transmission. Implementation and testing were carried out for medical emergency and telemedicine application. In this paper, the emergency case study is described.

Audio Conferencing Enhancements

OpenAIRE

VESTERINEN, LEENA

2006-01-01

Audio conferencing allows multiple people in distant locations to interact in a single voice call. Whilst it can be very useful service it also has several key disadvantages. This thesis study investigated the options for improving the user experience of the mobile teleconferencing applications. In particular, the use of 3D, spatial audio and visualinteractive functionality was investigated as the means of improving the intelligibility and audio perception during the audio...

Huffman coding in advanced audio coding standard

Science.gov (United States)

Brzuchalski, Grzegorz

2012-05-01

This article presents several hardware architectures of Advanced Audio Coding (AAC) Huffman noiseless encoder, its optimisations and working implementation. Much attention has been paid to optimise the demand of hardware resources especially memory size. The aim of design was to get as short binary stream as possible in this standard. The Huffman encoder with whole audio-video system has been implemented in FPGA devices.

Intelligent audio analysis

CERN Document Server

Schuller, Björn W

2013-01-01

This book provides the reader with the knowledge necessary for comprehension of the field of Intelligent Audio Analysis. It firstly introduces standard methods and discusses the typical Intelligent Audio Analysis chain going from audio data to audio features to audio recognition.  Further, an introduction to audio source separation, and enhancement and robustness are given. After the introductory parts, the book shows several applications for the three types of audio: speech, music, and general sound. Each task is shortly introduced, followed by a description of the specific data and methods applied, experiments and results, and a conclusion for this specific task. The books provides benchmark results and standardized test-beds for a broader range of audio analysis tasks. The main focus thereby lies on the parallel advancement of realism in audio analysis, as too often today’s results are overly optimistic owing to idealized testing conditions, and it serves to stimulate synergies arising from transfer of ...

106-17 Telemetry Standards Digitized Audio Telemetry Standard Chapter 5

Science.gov (United States)

2017-07-01

Digitized Audio Telemetry Standard 5.1 General This chapter defines continuously variable slope delta (CVSD) modulation as the standard for digitizing...audio signal. The CVSD modulator is, in essence , a 1-bit analog-to-digital converter. The output of this 1-bit encoder is a serial bit stream, where

Analysis of sound data streamed over the network

Directory of Open Access Journals (Sweden)

Jiří Fejfar

2013-01-01

Full Text Available In this paper we inspect a difference between original sound recording and signal captured after streaming this original recording over a network loaded with a heavy traffic. There are several kinds of failures occurring in the captured recording caused by network congestion. We try to find a method how to evaluate correctness of streamed audio. Usually there are metrics based on a human perception of a signal such as “signal is clear, without audible failures”, “signal is having some failures but it is understandable”, or “signal is inarticulate”. These approaches need to be statistically evaluated on a broad set of respondents, which is time and resource consuming. We try to propose some metrics based on signal properties allowing us to compare the original and captured recording. We use algorithm called Dynamic Time Warping (Müller, 2007 commonly used for time series comparison in this paper. Some other time series exploration approaches can be found in (Fejfar, 2011 and (Fejfar, 2012. The data was acquired in our network laboratory simulating network traffic by downloading files, streaming audio and video simultaneously. Our former experiment inspected Quality of Service (QoS and its impact on failures of received audio data stream. This experiment is focused on the comparison of sound recordings rather than network mechanism.We focus, in this paper, on a real time audio stream such as a telephone call, where it is not possible to stream audio in advance to a “pool”. Instead it is necessary to achieve as small delay as possible (between speaker voice recording and listener voice replay. We are using RTP protocol for streaming audio.

Interpolation Filter Design for Hearing-Aid Audio Class-D Output Stage Application

DEFF Research Database (Denmark)

Pracný, Peter; Bruun, Erik; Llimos Muntal, Pere

2012-01-01

This paper deals with a design of a digital interpolation filter for a 3rd order multi-bit ΣΔ modulator with over-sampling ratio OSR = 64. The interpolation filter and the ΣΔ modulator are part of the back-end of an audio signal processing system in a hearing-aid application. The aim in this paper...... is to compare this design to designs presented in other state-of-the-art works ranging from hi-fi audio to hearing-aids. By performing comparison, trends and tradeoffs in interpolation filter design are indentified and hearing-aid specifications are derived. The possibilities for hardware reduction...... in the interpolation filter are investigated. Proposed design simplifications presented here result in the least hardware demanding combination of oversampling ratio, number of stages and number of filter taps among a number of filters reported for audio applications....

Local Control of Audio Environment: A Review of Methods and Applications

Directory of Open Access Journals (Sweden)

Jussi Kuutti

2014-02-01

Full Text Available The concept of a local audio environment is to have sound playback locally restricted such that, ideally, adjacent regions of an indoor or outdoor space could exhibit their own individual audio content without interfering with each other. This would enable people to listen to their content of choice without disturbing others next to them, yet, without any headphones to block conversation. In practice, perfect sound containment in free air cannot be attained, but a local audio environment can still be satisfactorily approximated using directional speakers. Directional speakers may be based on regular audible frequencies or they may employ modulated ultrasound. Planar, parabolic, and array form factors are commonly used. The directivity of a speaker improves as its surface area and sound frequency increases, making these the main design factors for directional audio systems. Even directional speakers radiate some sound outside the main beam, and sound can also reflect from objects. Therefore, directional speaker systems perform best when there is enough ambient noise to mask the leaking sound. Possible areas of application for local audio include information and advertisement audio feed in commercial facilities, guiding and narration in museums and exhibitions, office space personalization, control room messaging, rehabilitation environments, and entertainment audio systems.

STREAM2016: Streaming Requirements, Experience, Applications and Middleware Workshop

Energy Technology Data Exchange (ETDEWEB)

Fox, Geoffrey [Indiana Univ., Bloomington, IN (United States); Jha, Shantenu [Rutgers Univ., New Brunswick, NJ (United States); Ramakrishnan, Lavanya [Lawrence Berkeley National Lab. (LBNL), Berkeley, CA (United States)

2016-10-01

The Department of Energy (DOE) Office of Science (SC) facilities including accelerators, light sources and neutron sources and sensors that study, the environment, and the atmosphere, are producing streaming data that needs to be analyzed for next-generation scientific discoveries. There has been an explosion of new research and technologies for stream analytics arising from the academic and private sectors. However, there has been no corresponding effort in either documenting the critical research opportunities or building a community that can create and foster productive collaborations. The two-part workshop series, STREAM: Streaming Requirements, Experience, Applications and Middleware Workshop (STREAM2015 and STREAM2016), were conducted to bring the community together and identify gaps and future efforts needed by both NSF and DOE. This report describes the discussions, outcomes and conclusions from STREAM2016: Streaming Requirements, Experience, Applications and Middleware Workshop, the second of these workshops held on March 22-23, 2016 in Tysons, VA. STREAM2016 focused on the Department of Energy (DOE) applications, computational and experimental facilities, as well software systems. Thus, the role of “streaming and steering” as a critical mode of connecting the experimental and computing facilities was pervasive through the workshop. Given the overlap in interests and challenges with industry, the workshop had significant presence from several innovative companies and major contributors. The requirements that drive the proposed research directions, identified in this report, show an important opportunity for building competitive research and development program around streaming data. These findings and recommendations are consistent with vision outlined in NRC Frontiers of Data and National Strategic Computing Initiative (NCSI) [1, 2]. The discussions from the workshop are captured as topic areas covered in this report's sections. The report

Reconfigurable Multicore Architectures for Streaming Applications

NARCIS (Netherlands)

Smit, Gerardus Johannes Maria; Kokkeler, Andre B.J.; Rauwerda, G.K.; Jacobs, J.W.M.; Nicolescu, G.; Mosterman, P.J.

2009-01-01

This chapter addresses reconfigurable heterogenous and homogeneous multicore system-on-chip (SoC) platforms for streaming digital signal processing applications, also called DSP applications. In streaming DSP applications, computations can be specified as a data flow graph with streams of data items

Speech and audio processing for coding, enhancement and recognition

CERN Document Server

Togneri, Roberto; Narasimha, Madihally

2015-01-01

This book describes the basic principles underlying the generation, coding, transmission and enhancement of speech and audio signals, including advanced statistical and machine learning techniques for speech and speaker recognition with an overview of the key innovations in these areas. Key research undertaken in speech coding, speech enhancement, speech recognition, emotion recognition and speaker diarization are also presented, along with recent advances and new paradigms in these areas. ·         Offers readers a single-source reference on the significant applications of speech and audio processing to speech coding, speech enhancement and speech/speaker recognition. Enables readers involved in algorithm development and implementation issues for speech coding to understand the historical development and future challenges in speech coding research; ·         Discusses speech coding methods yielding bit-streams that are multi-rate and scalable for Voice-over-IP (VoIP) Networks; ·     �...

Securing Digital Audio using Complex Quadratic Map

Science.gov (United States)

Suryadi, MT; Satria Gunawan, Tjandra; Satria, Yudi

2018-03-01

In This digital era, exchanging data are common and easy to do, therefore it is vulnerable to be attacked and manipulated from unauthorized parties. One data type that is vulnerable to attack is digital audio. So, we need data securing method that is not vulnerable and fast. One of the methods that match all of those criteria is securing the data using chaos function. Chaos function that is used in this research is complex quadratic map (CQM). There are some parameter value that causing the key stream that is generated by CQM function to pass all 15 NIST test, this means that the key stream that is generated using this CQM is proven to be random. In addition, samples of encrypted digital sound when tested using goodness of fit test are proven to be uniform, so securing digital audio using this method is not vulnerable to frequency analysis attack. The key space is very huge about 8.1×l031 possible keys and the key sensitivity is very small about 10-10, therefore this method is also not vulnerable against brute-force attack. And finally, the processing speed for both encryption and decryption process on average about 450 times faster that its digital audio duration.

Application Layer Multicast

Science.gov (United States)

Allani, Mouna; Garbinato, Benoît; Pedone, Fernando

An increasing number of Peer-to-Peer (P2P) Internet applications rely today on data dissemination as their cornerstone, e.g., audio or video streaming, multi-party games. These applications typically depend on some support for multicast communication, where peers interested in a given data stream can join a corresponding multicast group. As a consequence, the efficiency, scalability, and reliability guarantees of these applications are tightly coupled with that of the underlying multicast mechanism.

Streaming Media Seminar--Effective Development and Distribution of Streaming Multimedia in Education

Science.gov (United States)

Mainhart, Robert; Gerraughty, James; Anderson, Kristine M.

2004-01-01

Concisely defined, "streaming media" is moving video and/or audio transmitted over the Internet for immediate viewing/listening by an end user. However, at Saint Francis University's Center of Excellence for Remote and Medically Under-Served Areas (CERMUSA), streaming media is approached from a broader perspective. The working definition includes…

Introduction to audio analysis a MATLAB approach

CERN Document Server

Giannakopoulos, Theodoros

2014-01-01

Introduction to Audio Analysis serves as a standalone introduction to audio analysis, providing theoretical background to many state-of-the-art techniques. It covers the essential theory necessary to develop audio engineering applications, but also uses programming techniques, notably MATLAB®, to take a more applied approach to the topic. Basic theory and reproducible experiments are combined to demonstrate theoretical concepts from a practical point of view and provide a solid foundation in the field of audio analysis. Audio feature extraction, audio classification, audio segmentation, au

The use of ambient audio to increase safety and immersion in location-based games

Science.gov (United States)

Kurczak, John Jason

The purpose of this thesis is to propose an alternative type of interface for mobile software being used while walking or running. Our work addresses the problem of visual user interfaces for mobile software be- ing potentially unsafe for pedestrians, and not being very immersive when used for location-based games. In addition, location-based games and applications can be dif- ficult to develop when directly interfacing with the sensors used to track the user's location. These problems need to be addressed because portable computing devices are be- coming a popular tool for navigation, playing games, and accessing the internet while walking. This poses a safety problem for mobile users, who may be paying too much attention to their device to notice and react to hazards in their environment. The difficulty of developing location-based games and other location-aware applications may significantly hinder the prevalence of applications that explore new interaction techniques for ubiquitous computing. We created the TREC toolkit to address the issues with tracking sensors while developing location-based games and applications. We have developed functional location-based applications with TREC to demonstrate the amount of work that can be saved by using this toolkit. In order to have a safer and more immersive alternative to visual interfaces, we have developed ambient audio interfaces for use with mobile applications. Ambient audio uses continuous streams of sound over headphones to present information to mobile users without distracting them from walking safely. In order to test the effectiveness of ambient audio, we ran a study to compare ambient audio with handheld visual interfaces in a location-based game. We compared players' ability to safely navigate the environment, their sense of immersion in the game, and their performance at the in-game tasks. We found that ambient audio was able to significantly increase players' safety and sense of immersion compared to a

Modified BTC Algorithm for Audio Signal Coding

Directory of Open Access Journals (Sweden)

TOMIC, S.

2016-11-01

Full Text Available This paper describes modification of a well-known image coding algorithm, named Block Truncation Coding (BTC and its application in audio signal coding. BTC algorithm was originally designed for black and white image coding. Since black and white images and audio signals have different statistical characteristics, the application of this image coding algorithm to audio signal presents a novelty and a challenge. Several implementation modifications are described in this paper, while the original idea of the algorithm is preserved. The main modifications are performed in the area of signal quantization, by designing more adequate quantizers for audio signal processing. The result is a novel audio coding algorithm, whose performance is presented and analyzed in this research. The performance analysis indicates that this novel algorithm can be successfully applied in audio signal coding.

Parametric time-frequency domain spatial audio

CERN Document Server

Delikaris-Manias, Symeon; Politis, Archontis

2018-01-01

This book provides readers with the principles and best practices in spatial audio signal processing. It describes how sound fields and their perceptual attributes are captured and analyzed within the time-frequency domain, how essential representation parameters are coded, and how such signals are efficiently reproduced for practical applications. The book is split into four parts starting with an overview of the fundamentals. It then goes on to explain the reproduction of spatial sound before offering an examination of signal-dependent spatial filtering. The book finishes with coverage of both current and future applications and the direction that spatial audio research is heading in. Parametric Time-frequency Domain Spatial Audio focuses on applications in entertainment audio, including music, home cinema, and gaming--covering the capturing and reproduction of spatial sound as well as its generation, transduction, representation, transmission, and perception. This book will teach readers the tools needed...

Perancangan Radio Streaming Edukasi (Studi Kasus Balai Pengembangan Media Radio YOGYAKARTA)

OpenAIRE

Nurwulan, Ayu Isni; Paputungan, Irving Vitra

2009-01-01

Pendidikan berkualitas sudah sewajarnya bisa dinikmati secara merata oleh semua orang. Mediapembelajaran secara audio yang selama ini disampaikan masih memiliki banyak keterbatasan, terutama padalingkup wilayah penyampaian. Dalam makalah ini, sebuah media pendidikan berbasis audio dengan cara laindiusulkan. Media tersebut bernama radio streaming. Pembuatan radio streaming memerlukan banyak analisissehingga perancangannya tepat. Hasil analisis dan perancangan yang disampaikan dalam makalah ini...

Audio-Visual Speech Recognition Using MPEG-4 Compliant Visual Features

Directory of Open Access Journals (Sweden)

Petar S. Aleksic

2002-11-01

Full Text Available We describe an audio-visual automatic continuous speech recognition system, which significantly improves speech recognition performance over a wide range of acoustic noise levels, as well as under clean audio conditions. The system utilizes facial animation parameters (FAPs supported by the MPEG-4 standard for the visual representation of speech. We also describe a robust and automatic algorithm we have developed to extract FAPs from visual data, which does not require hand labeling or extensive training procedures. The principal component analysis (PCA was performed on the FAPs in order to decrease the dimensionality of the visual feature vectors, and the derived projection weights were used as visual features in the audio-visual automatic speech recognition (ASR experiments. Both single-stream and multistream hidden Markov models (HMMs were used to model the ASR system, integrate audio and visual information, and perform a relatively large vocabulary (approximately 1000 words speech recognition experiments. The experiments performed use clean audio data and audio data corrupted by stationary white Gaussian noise at various SNRs. The proposed system reduces the word error rate (WER by 20% to 23% relatively to audio-only speech recognition WERs, at various SNRs (0–30 dB with additive white Gaussian noise, and by 19% relatively to audio-only speech recognition WER under clean audio conditions.

Big Data Analytics: Challenges And Applications For Text, Audio, Video, And Social Media Data

OpenAIRE

Jai Prakash Verma; Smita Agrawal; Bankim Patel; Atul Patel

2016-01-01

All types of machine automated systems are generating large amount of data in different forms like statistical, text, audio, video, sensor, and bio-metric data that emerges the term Big Data. In this paper we are discussing issues, challenges, and application of these types of Big Data with the consideration of big data dimensions. Here we are discussing social media data analytics, content based analytics, text data analytics, audio, and video data analytics their issues and expected applica...

Coexistence issues for a 2.4 GHz wireless audio streaming in presence of bluetooth paging and WLAN

Science.gov (United States)

Pfeiffer, F.; Rashwan, M.; Biebl, E.; Napholz, B.

2015-11-01

Nowadays, customers expect to integrate their mobile electronic devices (smartphones and laptops) in a vehicle to form a wireless network. Typically, IEEE 802.11 is used to provide a high-speed wireless local area network (WLAN) and Bluetooth is used for cable replacement applications in a wireless personal area network (PAN). In addition, Daimler uses KLEER as third wireless technology in the unlicensed (UL) 2.4 GHz-ISM-band to transmit full CD-quality digital audio. As Bluetooth, IEEE 802.11 and KLEER are operating in the same frequency band, it has to be ensured that all three technologies can be used simultaneously without interference. In this paper, we focus on the impact of Bluetooth and IEEE 802.11 as interferer in presence of a KLEER audio transmission.

STEGANOGRAPHY USAGE TO CONTROL MULTIMEDIA STREAM

Directory of Open Access Journals (Sweden)

Grzegorz Koziel

2014-03-01

Full Text Available In the paper, a proposal of new application for steganography is presented. It is possible to use steganographic techniques to control multimedia stream playback. Special control markers can be included in the sound signal and the player can detect markers and modify the playback parameters according to the hidden instructions. This solution allows for remembering user preferences within the audio track as well as allowing for preparation of various versions of the same content at the production level.

High-Fidelity Piezoelectric Audio Device

Science.gov (United States)

Woodward, Stanley E.; Fox, Robert L.; Bryant, Robert G.

2003-01-01

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

Smartphone audio port data collection cookbook

Directory of Open Access Journals (Sweden)

Kyle Forinash

2018-06-01

Full Text Available The audio port of a smartphone is designed to send and receive audio but can be harnessed for portable, economical, and accurate data collection from a variety of sources. While smartphones have internal sensors to measure a number of physical phenomena such as acceleration, magnetism and illumination levels, measurement of other phenomena such as voltage, external temperature, or accurate timing of moving objects are excluded. The audio port cannot be only employed to sense external phenomena. It has the additional advantage of timing precision; because audio is recorded or played at a controlled rate separated from other smartphone activities, timings based on audio can be highly accurate. The following outlines unpublished details of the audio port technical elements for data collection, a general data collection recipe and an example timing application for Android devices.

Audio Recording of Children with Dyslalia

OpenAIRE

Stefan Gheorghe Pentiuc; Maria D. Schipor; Ovidiu A. Schipor

2008-01-01

In this paper we present our researches regarding automat parsing of audio recordings. These recordings are obtained from children with dyslalia and are necessary for an accurate identification of speech problems. We develop a software application that helps parsing audio, real time, recordings.

AUDIO CRYPTANALYSIS- AN APPLICATION OF SYMMETRIC KEY CRYPTOGRAPHY AND AUDIO STEGANOGRAPHY

Directory of Open Access Journals (Sweden)

Smita Paira

2016-09-01

Full Text Available In the recent trend of network and technology, “Cryptography” and “Steganography” have emerged out as the essential elements of providing network security. Although Cryptography plays a major role in the fabrication and modification of the secret message into an encrypted version yet it has certain drawbacks. Steganography is the art that meets one of the basic limitations of Cryptography. In this paper, a new algorithm has been proposed based on both Symmetric Key Cryptography and Audio Steganography. The combination of a randomly generated Symmetric Key along with LSB technique of Audio Steganography sends a secret message unrecognizable through an insecure medium. The Stego File generated is almost lossless giving a 100 percent recovery of the original message. This paper also presents a detailed experimental analysis of the algorithm with a brief comparison with other existing algorithms and a future scope. The experimental verification and security issues are promising.

Audio Recording of Children with Dyslalia

Directory of Open Access Journals (Sweden)

Stefan Gheorghe Pentiuc

2008-01-01

Full Text Available In this paper we present our researches regarding automat parsing of audio recordings. These recordings are obtained from children with dyslalia and are necessary for an accurate identification of speech problems. We develop a software application that helps parsing audio, real time, recordings.

Wavelet-based audio embedding and audio/video compression

Science.gov (United States)

Mendenhall, Michael J.; Claypoole, Roger L., Jr.

2001-12-01

Watermarking, traditionally used for copyright protection, is used in a new and exciting way. An efficient wavelet-based watermarking technique embeds audio information into a video signal. Several effective compression techniques are applied to compress the resulting audio/video signal in an embedded fashion. This wavelet-based compression algorithm incorporates bit-plane coding, index coding, and Huffman coding. To demonstrate the potential of this audio embedding and audio/video compression algorithm, we embed an audio signal into a video signal and then compress. Results show that overall compression rates of 15:1 can be achieved. The video signal is reconstructed with a median PSNR of nearly 33 dB. Finally, the audio signal is extracted from the compressed audio/video signal without error.

Aurally Aided Visual Search Performance Comparing Virtual Audio Systems

DEFF Research Database (Denmark)

Larsen, Camilla Horne; Lauritsen, David Skødt; Larsen, Jacob Junker

2014-01-01

Due to increased computational power, reproducing binaural hearing in real-time applications, through usage of head-related transfer functions (HRTFs), is now possible. This paper addresses the differences in aurally-aided visual search performance between a HRTF enhanced audio system (3D) and an...... with white dots. The results indicate that 3D audio yields faster search latencies than panning audio, especially with larger amounts of distractors. The applications of this research could fit virtual environments such as video games or virtual simulations.......Due to increased computational power, reproducing binaural hearing in real-time applications, through usage of head-related transfer functions (HRTFs), is now possible. This paper addresses the differences in aurally-aided visual search performance between a HRTF enhanced audio system (3D...

Aurally Aided Visual Search Performance Comparing Virtual Audio Systems

DEFF Research Database (Denmark)

Larsen, Camilla Horne; Lauritsen, David Skødt; Larsen, Jacob Junker

2014-01-01

Due to increased computational power reproducing binaural hearing in real-time applications, through usage of head-related transfer functions (HRTFs), is now possible. This paper addresses the differences in aurally-aided visual search performance between an HRTF enhanced audio system (3D) and an...... with white dots. The results indicate that 3D audio yields faster search latencies than panning audio, especially with larger amounts of distractors. The applications of this research could fit virtual environments such as video games or virtual simulations.......Due to increased computational power reproducing binaural hearing in real-time applications, through usage of head-related transfer functions (HRTFs), is now possible. This paper addresses the differences in aurally-aided visual search performance between an HRTF enhanced audio system (3D...

Audio production principles practical studio applications

CERN Document Server

Elmosnino, Stephane

2018-01-01

A new and fully practical guide to all of the key topics in audio production, this book covers the entire workflow from pre-production, to recording all kinds of instruments, to mixing theories and tools, and finally to mastering.

Book review: An Introduction to Audio Content Analysis: Applications in Signal Processing and Music Informatics by Alexander Lerch

DEFF Research Database (Denmark)

Sturm, Bob L.

2013-01-01

A critical review of the book: An Introduction to Audio Content Analysis: Applications in Signal Processing and Music Informatics, by Alexander Lerch, October 2012, Wiley-IEEE Press. ISBN: 978-1-118-26682-3, Hardcover, 272 pages, 503 references. List price $125.00......A critical review of the book: An Introduction to Audio Content Analysis: Applications in Signal Processing and Music Informatics, by Alexander Lerch, October 2012, Wiley-IEEE Press. ISBN: 978-1-118-26682-3, Hardcover, 272 pages, 503 references. List price $125.00...

EVALUASI KEPUASAN PENGGUNA TERHADAP APLIKASI AUDIO BOOKS

Directory of Open Access Journals (Sweden)

Raditya Maulana Anuraga

2017-02-01

Full Text Available Listeno is the first application audio books in Indonesia so that the users can get the book in audio form like listen to music, Listeno have problems in a feature request Listeno offline mode that have not been released, a security problem mp3 files that must be considered, and the target Listeno not yet reached 100,000 active users. This research has the objective to evaluate user satisfaction to Audio Books with research method approach, Nielsen. The analysis in this study using Importance Performance Analysis (IPA is combined with the index of User Satisfaction (IKP based on the indicators used are: Benefit (Usefulness, Utility (Utility, Usability (Usability, easy to understand (Learnability, Efficient (efficiency , Easy to remember (Memorability, Error (Error, and satisfaction (satisfaction. The results showed Applications User Satisfaction Audio books are quite satisfied with the results of the calculation IKP 69.58%..

Audio Papers

DEFF Research Database (Denmark)

Groth, Sanne Krogh; Samson, Kristine

2016-01-01

With this special issue of Seismograf we are happy to present a new format of articles: Audio Papers. Audio papers resemble the regular essay or the academic text in that they deal with a certain topic of interest, but presented in the form of an audio production. The audio paper is an extension...

DAFX Digital Audio Effects

CERN Document Server

Zö

2011-01-01

The rapid development in various fields of Digital Audio Effects, or DAFX, has led to new algorithms and this second edition of the popular book, DAFX: Digital Audio Effects has been updated throughout to reflect progress in the field. It maintains a unique approach to DAFX with a lecture-style introduction into the basics of effect processing. Each effect description begins with the presentation of the physical and acoustical phenomena, an explanation of the signal processing techniques to achieve the effect, followed by a discussion of musical applications and the control of effect parameter

A Smart Audio on Demand Application on Android Systems

Directory of Open Access Journals (Sweden)

Ing-Jr Ding

2015-05-01

Full Text Available This paper describes a study of the realization of intelligent Audio on Demand (AOD processing in the embedded system environment. This study describes the development of innovative Android software that will enhance user experience of the increasingly popular number of smart mobile devices now available on the market. The application we developed can accumulate records of the songs that are played and automatically analyze the favorite song types of a user. The application can also select sound control playback functions to make operation more convenient. A large number of different types of music genre were collected to create a sound database and build an intelligent AOD processing mechanism. Formant analysis was used to extract voice features and the K-means clustering method and acoustic modeling technology of the Gaussian mixture model (GMM were used to study and develop the application mechanism. The processes we developed run smoothly in the embedded Android platform.

[Intermodal timing cues for audio-visual speech recognition].

Science.gov (United States)

Hashimoto, Masahiro; Kumashiro, Masaharu

2004-06-01

The purpose of this study was to investigate the limitations of lip-reading advantages for Japanese young adults by desynchronizing visual and auditory information in speech. In the experiment, audio-visual speech stimuli were presented under the six test conditions: audio-alone, and audio-visually with either 0, 60, 120, 240 or 480 ms of audio delay. The stimuli were the video recordings of a face of a female Japanese speaking long and short Japanese sentences. The intelligibility of the audio-visual stimuli was measured as a function of audio delays in sixteen untrained young subjects. Speech intelligibility under the audio-delay condition of less than 120 ms was significantly better than that under the audio-alone condition. On the other hand, the delay of 120 ms corresponded to the mean mora duration measured for the audio stimuli. The results implied that audio delays of up to 120 ms would not disrupt lip-reading advantage, because visual and auditory information in speech seemed to be integrated on a syllabic time scale. Potential applications of this research include noisy workplace in which a worker must extract relevant speech from all the other competing noises.

Parametric Audio Based Decoder and Music Synthesizer for Mobile Applications

NARCIS (Netherlands)

Oomen, A.W.J.; Szczerba, M.Z.; Therssen, D.

2011-01-01

This paper reviews parametric audio coders and discusses novel technologies introduced in a low-complexity, low-power consumption audiodecoder and music synthesizer platform developed by the authors. Thedecoder uses parametric coding scheme based on the MPEG-4 Parametric Audio standard. In order to

Improvements of ModalMax High-Fidelity Piezoelectric Audio Device

Science.gov (United States)

Woodard, Stanley E.

2005-01-01

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

All About Audio Equalization: Solutions and Frontiers

Directory of Open Access Journals (Sweden)

Vesa Välimäki

2016-05-01

Full Text Available Audio equalization is a vast and active research area. The extent of research means that one often cannot identify the preferred technique for a particular problem. This review paper bridges those gaps, systemically providing a deep understanding of the problems and approaches in audio equalization, their relative merits and applications. Digital signal processing techniques for modifying the spectral balance in audio signals and applications of these techniques are reviewed, ranging from classic equalizers to emerging designs based on new advances in signal processing and machine learning. Emphasis is placed on putting the range of approaches within a common mathematical and conceptual framework. The application areas discussed herein are diverse, and include well-defined, solvable problems of filter design subject to constraints, as well as newly emerging challenges that touch on problems in semantics, perception and human computer interaction. Case studies are given in order to illustrate key concepts and how they are applied in practice. We also recommend preferred signal processing approaches for important audio equalization problems. Finally, we discuss current challenges and the uncharted frontiers in this field. The source code for methods discussed in this paper is made available at https://code.soundsoftware.ac.uk/projects/allaboutaudioeq.

An introduction to audio content analysis applications in signal processing and music informatics

CERN Document Server

Lerch, Alexander

2012-01-01

"With the proliferation of digital audio distribution over digital media, audio content analysis is fast becoming a requirement for designers of intelligent signal-adaptive audio processing systems. Written by a well-known expert in the field, this book provides quick access to different analysis algorithms and allows comparison between different approaches to the same task, making it useful for newcomers to audio signal processing and industry experts alike. A review of relevant fundamentals in audio signal processing, psychoacoustics, and music theory, as well as downloadable MATLAB files are also included"--

AudioMUD: a multiuser virtual environment for blind people.

Science.gov (United States)

Sánchez, Jaime; Hassler, Tiago

2007-03-01

A number of virtual environments have been developed during the last years. Among them there are some applications for blind people based on different type of audio, from simple sounds to 3-D audio. In this study, we pursued a different approach. We designed AudioMUD by using spoken text to describe the environment, navigation, and interaction. We have also introduced some collaborative features into the interaction between blind users. The core of a multiuser MUD game is a networked textual virtual environment. We developed AudioMUD by adding some collaborative features to the basic idea of a MUD and placed a simulated virtual environment inside the human body. This paper presents the design and usability evaluation of AudioMUD. Blind learners were motivated when interacted with AudioMUD and helped to improve the interaction through audio and interface design elements.

Warped Linear Prediction of Physical Model Excitations with Applications in Audio Compression and Instrument Synthesis

Science.gov (United States)

Glass, Alexis; Fukudome, Kimitoshi

2004-12-01

A sound recording of a plucked string instrument is encoded and resynthesized using two stages of prediction. In the first stage of prediction, a simple physical model of a plucked string is estimated and the instrument excitation is obtained. The second stage of prediction compensates for the simplicity of the model in the first stage by encoding either the instrument excitation or the model error using warped linear prediction. These two methods of compensation are compared with each other, and to the case of single-stage warped linear prediction, adjustments are introduced, and their applications to instrument synthesis and MPEG4's audio compression within the structured audio format are discussed.

AUTOMATIC SEGMENTATION OF BROADCAST AUDIO SIGNALS USING AUTO ASSOCIATIVE NEURAL NETWORKS

Directory of Open Access Journals (Sweden)

P. Dhanalakshmi

2010-12-01

Full Text Available In this paper, we describe automatic segmentation methods for audio broadcast data. Today, digital audio applications are part of our everyday lives. Since there are more and more digital audio databases in place these days, the importance of effective management for audio databases have become prominent. Broadcast audio data is recorded from the Television which comprises of various categories of audio signals. Efficient algorithms for segmenting the audio broadcast data into predefined categories are proposed. Audio features namely Linear prediction coefficients (LPC, Linear prediction cepstral coefficients, and Mel frequency cepstral coefficients (MFCC are extracted to characterize the audio data. Auto Associative Neural Networks are used to segment the audio data into predefined categories using the extracted features. Experimental results indicate that the proposed algorithms can produce satisfactory results.

Acoustofluidics 14: Applications of acoustic streaming in microfluidic devices.

Science.gov (United States)

Wiklund, Martin; Green, Roy; Ohlin, Mathias

2012-07-21

In part 14 of the tutorial series "Acoustofluidics--exploiting ultrasonic standing wave forces and acoustic streaming in microfluidic systems for cell and particle manipulation", we provide a qualitative description of acoustic streaming and review its applications in lab-on-a-chip devices. The paper covers boundary layer driven streaming, including Schlichting and Rayleigh streaming, Eckart streaming in the bulk fluid, cavitation microstreaming and surface-acoustic-wave-driven streaming.

StirMark Benchmark: audio watermarking attacks based on lossy compression

Science.gov (United States)

Steinebach, Martin; Lang, Andreas; Dittmann, Jana

2002-04-01

StirMark Benchmark is a well-known evaluation tool for watermarking robustness. Additional attacks are added to it continuously. To enable application based evaluation, in our paper we address attacks against audio watermarks based on lossy audio compression algorithms to be included in the test environment. We discuss the effect of different lossy compression algorithms like MPEG-2 audio Layer 3, Ogg or VQF on a selection of audio test data. Our focus is on changes regarding the basic characteristics of the audio data like spectrum or average power and on removal of embedded watermarks. Furthermore we compare results of different watermarking algorithms and show that lossy compression is still a challenge for most of them. There are two strategies for adding evaluation of robustness against lossy compression to StirMark Benchmark: (a) use of existing free compression algorithms (b) implementation of a generic lossy compression simulation. We discuss how such a model can be implemented based on the results of our tests. This method is less complex, as no real psycho acoustic model has to be applied. Our model can be used for audio watermarking evaluation of numerous application fields. As an example, we describe its importance for e-commerce applications with watermarking security.

Multiple frequency audio signal communication as a mechanism for neurophysiology and video data synchronization.

Science.gov (United States)

Topper, Nicholas C; Burke, Sara N; Maurer, Andrew Porter

2014-12-30

Current methods for aligning neurophysiology and video data are either prepackaged, requiring the additional purchase of a software suite, or use a blinking LED with a stationary pulse-width and frequency. These methods lack significant user interface for adaptation, are expensive, or risk a misalignment of the two data streams. A cost-effective means to obtain high-precision alignment of behavioral and neurophysiological data is obtained by generating an audio-pulse embedded with two domains of information, a low-frequency binary-counting signal and a high, randomly changing frequency. This enabled the derivation of temporal information while maintaining enough entropy in the system for algorithmic alignment. The sample to frame index constructed using the audio input correlation method described in this paper enables video and data acquisition to be aligned at a sub-frame level of precision. Traditionally, a synchrony pulse is recorded on-screen via a flashing diode. The higher sampling rate of the audio input of the camcorder enables the timing of an event to be detected with greater precision. While on-line analysis and synchronization using specialized equipment may be the ideal situation in some cases, the method presented in the current paper presents a viable, low cost alternative, and gives the flexibility to interface with custom off-line analysis tools. Moreover, the ease of constructing and implements this set-up presented in the current paper makes it applicable to a wide variety of applications that require video recording. Copyright © 2014 Elsevier B.V. All rights reserved.

The audio expert everything you need to know about audio

CERN Document Server

Winer, Ethan

2012-01-01

The Audio Expert is a comprehensive reference that covers all aspects of audio, with many practical, as well as theoretical, explanations. Providing in-depth descriptions of how audio really works, using common sense plain-English explanations and mechanical analogies with minimal math, the book is written for people who want to understand audio at the deepest, most technical level, without needing an engineering degree. It's presented in an easy-to-read, conversational tone, and includes more than 400 figures and photos augmenting the text.The Audio Expert takes th

Applying the EBU R128 loudness standard in live-streaming sound sculptures

DEFF Research Database (Denmark)

Højlund, Marie Koldkjær; Riis, Morten S.; Rothmann, Daniel

2017-01-01

to preserve a natural sounding dynamic image from the varying sound sources that can be played back under varying conditions, an adaptation of the EBU R128 loudness measurement recommendation, originally developed for levelling non-real-time broadcast material, has been applied. The paper describes the Pure......This paper describes the development of a loudness-based compressor for live audio streams. The need for this device arose while developing the public sound art project The Overheard, which involves mixing together several live audio streams through a web based mixing interface. In order...

Delivering Instruction via Streaming Media: A Higher Education Perspective.

Science.gov (United States)

Mortensen, Mark; Schlieve, Paul; Young, Jon

2000-01-01

Describes streaming media, an audio/video presentation that is delivered across a network so that it is viewed while being downloaded onto the user's computer, including a continuous stream of video that can be pre-recorded or live. Discusses its use for nontraditional students in higher education and reports on implementation experiences. (LRW)

Spatial audio reproduction with primary ambient extraction

CERN Document Server

He, JianJun

2017-01-01

This book first introduces the background of spatial audio reproduction, with different types of audio content and for different types of playback systems. A literature study on the classical and emerging Primary Ambient Extraction (PAE) techniques is presented. The emerging techniques aim to improve the extraction performance and also enhance the robustness of PAE approaches in dealing with more complex signals encountered in practice. The in-depth theoretical study helps readers to understand the rationales behind these approaches. Extensive objective and subjective experiments validate the feasibility of applying PAE in spatial audio reproduction systems. These experimental results, together with some representative audio examples and MATLAB codes of the key algorithms, illustrate clearly the differences among various approaches and also help readers gain insights on selecting different approaches for different applications.

Audio-Visual Tibetan Speech Recognition Based on a Deep Dynamic Bayesian Network for Natural Human Robot Interaction

Directory of Open Access Journals (Sweden)

Yue Zhao

2012-12-01

Full Text Available Audio-visual speech recognition is a natural and robust approach to improving human-robot interaction in noisy environments. Although multi-stream Dynamic Bayesian Network and coupled HMM are widely used for audio-visual speech recognition, they fail to learn the shared features between modalities and ignore the dependency of features among the frames within each discrete state. In this paper, we propose a Deep Dynamic Bayesian Network (DDBN to perform unsupervised extraction of spatial-temporal multimodal features from Tibetan audio-visual speech data and build an accurate audio-visual speech recognition model under a no frame-independency assumption. The experiment results on Tibetan speech data from some real-world environments showed the proposed DDBN outperforms the state-of-art methods in word recognition accuracy.

Near-field Localization of Audio

DEFF Research Database (Denmark)

Jensen, Jesper Rindom; Christensen, Mads Græsbøll

2014-01-01

Localization of audio sources using microphone arrays has been an important research problem for more than two decades. Many traditional methods for solving the problem are based on a two-stage procedure: first, information about the audio source, such as time differences-of-arrival (TDOAs......) and gain ratios-of-arrival (GROAs) between microphones is estimated, and, second, this knowledge is used to localize the audio source. These methods often have a low computational complexity, but this comes at the cost of a limited estimation accuracy. Therefore, we propose a new localization approach......, where the desired signal is modeled using TDOAs and GROAs, which are determined by the source location. This facilitates the derivation of one-stage, maximum likelihood methods under a white Gaussian noise assumption that is applicable in both near- and far-field scenarios. Simulations show...

Virtual Microphones for Multichannel Audio Resynthesis

Directory of Open Access Journals (Sweden)

Athanasios Mouchtaris

2003-09-01

Full Text Available Multichannel audio offers significant advantages for music reproduction, including the ability to provide better localization and envelopment, as well as reduced imaging distortion. On the other hand, multichannel audio is a demanding media type in terms of transmission requirements. Often, bandwidth limitations prohibit transmission of multiple audio channels. In such cases, an alternative is to transmit only one or two reference channels and recreate the rest of the channels at the receiving end. Here, we propose a system capable of synthesizing the required signals from a smaller set of signals recorded in a particular venue. These synthesized “virtual” microphone signals can be used to produce multichannel recordings that accurately capture the acoustics of that venue. Applications of the proposed system include transmission of multichannel audio over the current Internet infrastructure and, as an extension of the methods proposed here, remastering existing monophonic and stereophonic recordings for multichannel rendering.

Abstractions for aperiodic multiprocessor scheduling of real-time stream processing applications

NARCIS (Netherlands)

Hausmans, J.P.H.M.

2015-01-01

Embedded multiprocessor systems are often used in the domain of real-time stream processing applications to keep up with increasing power and performance requirements. Examples of such real-time stream processing applications are digital radio baseband processing and WLAN transceivers. These stream

Multimodal indexing of digital audio-visual documents: A case study for cultural heritage data

NARCIS (Netherlands)

Carmichael, J.; Larson, M.; Marlow, J.; Newman, E.; Clough, P.; Oomen, J.; Sav, S.

2008-01-01

This paper describes a multimedia multimodal information access sub-system (MIAS) for digital audio-visual documents, typically presented in streaming media format. The system is designed to provide both professional and general users with entry points into video documents that are relevant to their

Analysis and Implementation of Gossip-Based P2P Streaming with Distributed Incentive Mechanisms for Peer Cooperation

Directory of Open Access Journals (Sweden)

Sachin Agarwal

2007-10-01

Full Text Available Peer-to-peer (P2P systems are becoming a popular means of streaming audio and video content but they are prone to bandwidth starvation if selfish peers do not contribute bandwidth to other peers. We prove that an incentive mechanism can be created for a live streaming P2P protocol while preserving the asymptotic properties of randomized gossip-based streaming. In order to show the utility of our result, we adapt a distributed incentive scheme from P2P file storage literature to the live streaming scenario. We provide simulation results that confirm the ability to achieve a constant download rate (in time, per peer that is needed for streaming applications on peers. The incentive scheme fairly differentiates peers' download rates according to the amount of useful bandwidth they contribute back to the P2P system, thus creating a powerful quality-of-service incentive for peers to contribute bandwidth to other peers. We propose a functional architecture and protocol format for a gossip-based streaming system with incentive mechanisms, and present evaluation data from a real implementation of a P2P streaming application.

Audio Teleconferencing: Low Cost Technology for External Studies Networking.

Science.gov (United States)

Robertson, Bill

1987-01-01

This discussion of the benefits of audio teleconferencing for distance education programs and for business and government applications focuses on the recent experience of Canadian educational users. Four successful operating models and their costs are reviewed, and it is concluded that audio teleconferencing is cost efficient and educationally…

Musical examination to bridge audio data and sheet music

Science.gov (United States)

Pan, Xunyu; Cross, Timothy J.; Xiao, Liangliang; Hei, Xiali

2015-03-01

The digitalization of audio is commonly implemented for the purpose of convenient storage and transmission of music and songs in today's digital age. Analyzing digital audio for an insightful look at a specific musical characteristic, however, can be quite challenging for various types of applications. Many existing musical analysis techniques can examine a particular piece of audio data. For example, the frequency of digital sound can be easily read and identified at a specific section in an audio file. Based on this information, we could determine the musical note being played at that instant, but what if you want to see a list of all the notes played in a song? While most existing methods help to provide information about a single piece of the audio data at a time, few of them can analyze the available audio file on a larger scale. The research conducted in this work considers how to further utilize the examination of audio data by storing more information from the original audio file. In practice, we develop a novel musical analysis system Musicians Aid to process musical representation and examination of audio data. Musicians Aid solves the previous problem by storing and analyzing the audio information as it reads it rather than tossing it aside. The system can provide professional musicians with an insightful look at the music they created and advance their understanding of their work. Amateur musicians could also benefit from using it solely for the purpose of obtaining feedback about a song they were attempting to play. By comparing our system's interpretation of traditional sheet music with their own playing, a musician could ensure what they played was correct. More specifically, the system could show them exactly where they went wrong and how to adjust their mistakes. In addition, the application could be extended over the Internet to allow users to play music with one another and then review the audio data they produced. This would be particularly

AKTIVITAS SEKUNDER AUDIO UNTUK MENJAGA KEWASPADAAN PENGEMUDI MOBIL INDONESIA

Directory of Open Access Journals (Sweden)

Iftikar Zahedi Sutalaksana

2013-03-01

Full Text Available Tingkat kecelakaan lalu lintas yang melibatkan mobil di Indonesia semakin mengkhawatirkan. Tingginya peran faktor manusia sebagai penyebab utama kejadian kecelakaan patut diperhatikan. Penurunan kewaspadaan saat mengemudi akibat kantuk atau kelelahan merupakan salah satu kondisi yang mendorong terjadinya kecelakaan. Tulisan ini memaparkan aplikasi audio response test sebagai aktivitas sekunder dalam mengemudikan mobil. Response test yang dimaksud merupakan seperangkat aplikasi pada dashboard mobil yang menuntut respon pengemudi setiap stimulus suara bekerja. Audio response test ini diusulkan sebagai pemantau tingkat kewaspadaan pengemudi selama berkendara. Kewaspadaan pengemudi merupakan kondisi selama berkendara yang terjaga, awas, dan mampu memproses semua stimulus dengan baik. Hasil studi ini menghasilkan suatu bentuk audio response test yang terintegrasi dengan sistem berkendara di dalam mobil. Sumber bunyi diperdengarkan dengan intensitas konstan antara 80-85 dB. Bunyi akan berhenti jika pengemudi memberikan respon atas stimulus suara tersebut. Response test ini dirancang untuk mampu memantau tingkat kewaspadaan pengemudi selama berkendara. Penerapannya diharapkan mampu membantu menekan tingkat kecelakaan lalu lintas di Indonesia. Kata kunci: mengemudi, aktivitas sekunder, audio, kewaspadaan, response test   Abstract   The level of traffic accidents involving cars in Indonesia increasingly alarming. The high role of the human factor as the main cause of accident noteworthy. Decreased alertness while driving due to sleepiness or fatigue is one of the conditions that led to the accident. This paper describes an audio application response test as a secondary activity of driving a car. Response test is a set of applications on the dashboard of a car that demands a response driver each stimulus voice work. Audio response was proposed as test monitors the driver's level of alertness while driving. Vigilance driver was driving conditions during

Modified DCTNet for audio signals classification

Science.gov (United States)

Xian, Yin; Pu, Yunchen; Gan, Zhe; Lu, Liang; Thompson, Andrew

2016-10-01

In this paper, we investigate DCTNet for audio signal classification. Its output feature is related to Cohen's class of time-frequency distributions. We introduce the use of adaptive DCTNet (A-DCTNet) for audio signals feature extraction. The A-DCTNet applies the idea of constant-Q transform, with its center frequencies of filterbanks geometrically spaced. The A-DCTNet is adaptive to different acoustic scales, and it can better capture low frequency acoustic information that is sensitive to human audio perception than features such as Mel-frequency spectral coefficients (MFSC). We use features extracted by the A-DCTNet as input for classifiers. Experimental results show that the A-DCTNet and Recurrent Neural Networks (RNN) achieve state-of-the-art performance in bird song classification rate, and improve artist identification accuracy in music data. They demonstrate A-DCTNet's applicability to signal processing problems.

StreamWorks: the live and on-demand audio/video server and its applications in medical information systems

Science.gov (United States)

Akrout, Nabil M.; Gordon, Howard; Palisson, Patrice M.; Prost, Remy; Goutte, Robert

1996-05-01

Facing a world undergoing fundamental and rapid change, healthcare organizations are seeking ways to increase innovation, quality, productivity, and patient value, keys to more effective care. Individual clinics acting alone can respond in only a limited way, so re- engineering the process key which services are delivered demands real-time collaborative technology that provides immediate information sharing, improving the management and coordination of information in cross-functional teams. StreamWorks is a development stage architecture that uses a distribution technique to deliver an advanced information management system for telemedicine. The challenge of StreamWorks in telemedicine is to enable equity of the quality of Health Care of Telecommunications and Information Technology also to patients in less favored regions, like India or China, where the quality of medical care varies greatly by region, but where there are some very current communications facilities.

Open soundcard as a platform for practical, laboratory study of digital audio

DEFF Research Database (Denmark)

Dimitrov, Smilen; Serafin, Stefania

2014-01-01

This article investigates how lacking suitable platforms for laboratory exercises becomes a learning problem, limiting the practical experience students gain. In engineering education, laboratory demonstration difficulty of issues like real-time streaming in digital signal and audio processing...... afforded by such laboratories, and their open nature, could testably improve the diversity of demonstrated practical topics, while maintaining engineering students' motivation....

Video Streaming in Online Learning

Science.gov (United States)

Hartsell, Taralynn; Yuen, Steve Chi-Yin

2006-01-01

The use of video in teaching and learning is a common practice in education today. As learning online becomes more of a common practice in education, streaming video and audio will play a bigger role in delivering course materials to online learners. This form of technology brings courses alive by allowing online learners to use their visual and…

Routing Optimization of AVB Streams in TSN Networks

DEFF Research Database (Denmark)

Laursen, Sune Mølgaard; Pop, Paul; Steiner, Wilfried

2016-01-01

In this paper we are interested in safety-critical real-time applications implemented on distributed architectures using the Time-Sensitive Networking (TSN) standard. The ongoing standardization of TSN is an IEEE effort to bring deterministic real-time capabilities into the IEEE 802.1 Ethernet...... standard supporting safety-critical systems and guaranteed Quality-of-Service. TSN will support Time-Triggered (TT) communication based on schedule tables, Audio-Video-Bridging (AVB) streams with bounded end-to-end latency as well as Best-Effort messages. We consider that we know the topology...... Procedure (GRASP)-based heuristic for this routing optimization problem. The proposed approaches has been evaluated using several test cases....

Automatic Detection and Classification of Audio Events for Road Surveillance Applications

Directory of Open Access Journals (Sweden)

Noor Almaadeed

2018-06-01

Full Text Available This work investigates the problem of detecting hazardous events on roads by designing an audio surveillance system that automatically detects perilous situations such as car crashes and tire skidding. In recent years, research has shown several visual surveillance systems that have been proposed for road monitoring to detect accidents with an aim to improve safety procedures in emergency cases. However, the visual information alone cannot detect certain events such as car crashes and tire skidding, especially under adverse and visually cluttered weather conditions such as snowfall, rain, and fog. Consequently, the incorporation of microphones and audio event detectors based on audio processing can significantly enhance the detection accuracy of such surveillance systems. This paper proposes to combine time-domain, frequency-domain, and joint time-frequency features extracted from a class of quadratic time-frequency distributions (QTFDs to detect events on roads through audio analysis and processing. Experiments were carried out using a publicly available dataset. The experimental results conform the effectiveness of the proposed approach for detecting hazardous events on roads as demonstrated by 7% improvement of accuracy rate when compared against methods that use individual temporal and spectral features.

StreamStats: A water resources web application

Science.gov (United States)

Ries, Kernell G.; Guthrie, John G.; Rea, Alan H.; Steeves, Peter A.; Stewart, David W.

2008-01-01

. Streamflow measurements are collected systematically over a period of years at partial-record stations to estimate peak-flow or low-flow statistics. Streamflow measurements usually are collected at miscellaneous-measurement stations for specific hydrologic studies with various objectives.StreamStats is a Web-based Geographic Information System (GIS) application that was created by the USGS, in cooperation with Environmental Systems Research Institute, Inc. (ESRI)1, to provide users with access to an assortment of analytical tools that are useful for water-resources planning and management. StreamStats functionality is based on ESRI’s ArcHydro Data Model and Tools, described on the Web at http://resources.arcgis.com/en/communities/hydro/01vn0000000s000000.htm. StreamStats allows users to easily obtain streamflow statistics, basin characteristics, and descriptive information for USGS data-collection stations and user-selected ungaged sites. It also allows users to identify stream reaches that are upstream and downstream from user-selected sites, and to identify and obtain information for locations along the streams where activities that may affect streamflow conditions are occurring. This functionality can be accessed through a map-based user interface that appears in the user’s Web browser, or individual functions can be requested remotely as Web services by other Web or desktop computer applications. StreamStats can perform these analyses much faster than historically used manual techniques.StreamStats was designed so that each state would be implemented as a separate application, with a reliance on local partnerships to fund the individual applications, and a goal of eventual full national implementation. Idaho became the first state to implement StreamStats in 2003. By mid-2008, 14 states had applications available to the public, and 18 other states were in various stages of implementation.

The Chameleon Architecture for Streaming DSP Applications

NARCIS (Netherlands)

Bergmann, N.; Smit, Gerardus Johannes Maria; Kokkeler, Andre B.J.; Platzner, M.; Wolkotte, P.T.; Teich, J.; Holzenspies, P.K.F.; van de Burgwal, M.D.; Heysters, P.M.

2007-01-01

We focus on architectures for streaming DSP applications such as wireless baseband processing and image processing. We aim at a single generic architecture that is capable of dealing with different DSP applications. This architecture has to be energy efficient and fault tolerant. We introduce a

Audio Twister

DEFF Research Database (Denmark)

Cermak, Daniel; Moreno Garcia, Rodrigo; Monastiridis, Stefanos

2015-01-01

Daniel Cermak-Sassenrath, Rodrigo Moreno Garcia, Stefanos Monastiridis. Audio Twister. Installation. P-Hack Copenhagen 2015, Copenhagen, DK, Apr 24, 2015.......Daniel Cermak-Sassenrath, Rodrigo Moreno Garcia, Stefanos Monastiridis. Audio Twister. Installation. P-Hack Copenhagen 2015, Copenhagen, DK, Apr 24, 2015....

Back to basics audio

CERN Document Server

Nathan, Julian

1998-01-01

Back to Basics Audio is a thorough, yet approachable handbook on audio electronics theory and equipment. The first part of the book discusses electrical and audio principles. Those principles form a basis for understanding the operation of equipment and systems, covered in the second section. Finally, the author addresses planning and installation of a home audio system.Julian Nathan joined the audio service and manufacturing industry in 1954 and moved into motion picture engineering and production in 1960. He installed and operated recording theaters in Sydney, Austra

Identification of Sparse Audio Tampering Using Distributed Source Coding and Compressive Sensing Techniques

Directory of Open Access Journals (Sweden)

Valenzise G

2009-01-01

Full Text Available In the past few years, a large amount of techniques have been proposed to identify whether a multimedia content has been illegally tampered or not. Nevertheless, very few efforts have been devoted to identifying which kind of attack has been carried out, especially due to the large data required for this task. We propose a novel hashing scheme which exploits the paradigms of compressive sensing and distributed source coding to generate a compact hash signature, and we apply it to the case of audio content protection. The audio content provider produces a small hash signature by computing a limited number of random projections of a perceptual, time-frequency representation of the original audio stream; the audio hash is given by the syndrome bits of an LDPC code applied to the projections. At the content user side, the hash is decoded using distributed source coding tools. If the tampering is sparsifiable or compressible in some orthonormal basis or redundant dictionary, it is possible to identify the time-frequency position of the attack, with a hash size as small as 200 bits/second; the bit saving obtained by introducing distributed source coding ranges between 20% to 70%.

The application of controlled source audio frequency magnetotelluric method to prospecting for uranium and gold

International Nuclear Information System (INIS)

Guan Taiping

1992-01-01

The controlled source audio frequency magnetotelluric method is a new geophysical method which is rapid, effective and economical and can be used for studying the structural pattern of underground strata (rock bodies). This method provides the basis for the determination of the deeper part and structures within the unconformity-related uranium deposit in North China Platform and the result of application is optimal

Multilevel inverter based class D audio amplifier for capacitive transducers

DEFF Research Database (Denmark)

Nielsen, Dennis; Knott, Arnold; Andersen, Michael A. E.

2014-01-01

The reduced semiconductor voltage stress makes the multilevel inverters especially interesting, when driving capacitive transducers for audio applications. A ± 300 V flying capacitor class D audio amplifier driving a 100 nF load in the midrange region of 0.1-3.5 kHz with Total Harmonic Distortion...

Fall Detection Using Smartphone Audio Features.

Science.gov (United States)

Cheffena, Michael

2016-07-01

An automated fall detection system based on smartphone audio features is developed. The spectrogram, mel frequency cepstral coefficents (MFCCs), linear predictive coding (LPC), and matching pursuit (MP) features of different fall and no-fall sound events are extracted from experimental data. Based on the extracted audio features, four different machine learning classifiers: k-nearest neighbor classifier (k-NN), support vector machine (SVM), least squares method (LSM), and artificial neural network (ANN) are investigated for distinguishing between fall and no-fall events. For each audio feature, the performance of each classifier in terms of sensitivity, specificity, accuracy, and computational complexity is evaluated. The best performance is achieved using spectrogram features with ANN classifier with sensitivity, specificity, and accuracy all above 98%. The classifier also has acceptable computational requirement for training and testing. The system is applicable in home environments where the phone is placed in the vicinity of the user.

Reduction in time-to-sleep through EEG based brain state detection and audio stimulation.

Science.gov (United States)

Zhuo Zhang; Cuntai Guan; Ti Eu Chan; Juanhong Yu; Aung Aung Phyo Wai; Chuanchu Wang; Haihong Zhang

2015-08-01

We developed an EEG- and audio-based sleep sensing and enhancing system, called iSleep (interactive Sleep enhancement apparatus). The system adopts a closed-loop approach which optimizes the audio recording selection based on user's sleep status detected through our online EEG computing algorithm. The iSleep prototype comprises two major parts: 1) a sleeping mask integrated with a single channel EEG electrode and amplifier, a pair of stereo earphones and a microcontroller with wireless circuit for control and data streaming; 2) a mobile app to receive EEG signals for online sleep monitoring and audio playback control. In this study we attempt to validate our hypothesis that appropriate audio stimulation in relation to brain state can induce faster onset of sleep and improve the quality of a nap. We conduct experiments on 28 healthy subjects, each undergoing two nap sessions - one with a quiet background and one with our audio-stimulation. We compare the time-to-sleep in both sessions between two groups of subjects, e.g., fast and slow sleep onset groups. The p-value obtained from Wilcoxon Signed Rank Test is 1.22e-04 for slow onset group, which demonstrates that iSleep can significantly reduce the time-to-sleep for people with difficulty in falling sleep.

Digital Augmented Reality Audio Headset

Directory of Open Access Journals (Sweden)

Jussi Rämö

2012-01-01

Full Text Available Augmented reality audio (ARA combines virtual sound sources with the real sonic environment of the user. An ARA system can be realized with a headset containing binaural microphones. Ideally, the ARA headset should be acoustically transparent, that is, it should not cause audible modification to the surrounding sound. A practical implementation of an ARA mixer requires a low-latency headphone reproduction system with additional equalization to compensate for the attenuation and the modified ear canal resonances caused by the headphones. This paper proposes digital IIR filters to realize the required equalization and evaluates a real-time prototype ARA system. Measurements show that the throughput latency of the digital prototype ARA system can be less than 1.4 ms, which is sufficiently small in practice. When the direct and processed sounds are combined in the ear, a comb filtering effect is brought about and appears as notches in the frequency response. The comb filter effect in speech and music signals was studied in a listening test and it was found to be inaudible when the attenuation is 20 dB. Insert ARA headphones have a sufficient attenuation at frequencies above about 1 kHz. The proposed digital ARA system enables several immersive audio applications, such as a virtual audio tourist guide and audio teleconferencing.

Audio Quality Assurance : An Application of Cross Correlation

DEFF Research Database (Denmark)

Jurik, Bolette Ammitzbøll; Nielsen, Jesper Asbjørn Sindahl

2012-01-01

We describe algorithms for automated quality assurance on content of audio files in context of preservation actions and access. The algorithms use cross correlation to compare the sound waves. They are used to do overlap analysis in an access scenario, where preserved radio broadcasts are used in...

A Perceptual Model for Sinusoidal Audio Coding Based on Spectral Integration

NARCIS (Netherlands)

Van de Par, S.; Kohlrausch, A.; Heusdens, R.; Jensen, J.; Holdt Jensen, S.

2005-01-01

Psychoacoustical models have been used extensively within audio coding applications over the past decades. Recently, parametric coding techniques have been applied to general audio and this has created the need for a psychoacoustical model that is specifically suited for sinusoidal modelling of

A perceptual model for sinusoidal audio coding based on spectral integration

NARCIS (Netherlands)

Van de Par, S.; Kohlrauch, A.; Heusdens, R.; Jensen, J.; Jensen, S.H.

2005-01-01

Psychoacoustical models have been used extensively within audio coding applications over the past decades. Recently, parametric coding techniques have been applied to general audio and this has created the need for a psychoacoustical model that is specifically suited for sinusoidal modelling of

ATLAS Live: Collaborative Information Streams

CERN Document Server

Goldfarb, S; The ATLAS collaboration

2011-01-01

I report on a pilot project launched in 2010 focusing on facilitating communication and information exchange within the ATLAS Collaboration, through the combination of digital signage software and webcasting. The project, called ATLAS Live, implements video streams of information, ranging from detailed detector and data status to educational and outreach material. The content, including text, images, video and audio, is collected, visualised and scheduled using digital signage software. The system is robust and flexible, utilizing scripts to input data from remote sources, such as the CERN Document Server, Indico, or any available URL, and to integrate these sources into professional-quality streams, including text scrolling, transition effects, inter and intra-screen divisibility. Information is published via the encoding and webcasting of standard video streams, viewable on all common platforms, using a web browser or other common video tool. Authorisation is enforced at the level of the streaming and at th...

Specification and Compilation of Real-Time Stream Processing Applications

NARCIS (Netherlands)

Geuns, S.J.

2015-01-01

This thesis is concerned with the specification, compilation and corresponding temporal analysis of real-time stream processing applications that are executed on embedded multiprocessor systems. An example of such applications are software defined radio applications. These applications typically

Portable audio electronics for impedance-based measurements in microfluidics

International Nuclear Information System (INIS)

Wood, Paul; Sinton, David

2010-01-01

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. (technical note)

Optimal bus and buffer allocation for a set of leaky-bucket-controlled streams

NARCIS (Netherlands)

Boef, den E.; Korst, J.H.M.; Verhaegh, W.F.J.; De Souza, J.N.; Dini, P.; Lorenz, P.

2004-01-01

In an in-home digital network (IHDN) it may be expected that several variable-bit-rate streams (audio, video) run simultaneously over a shared communication device, e.g. a bus. The data supply and demand of most of these streams will not be exactly known in advance, but only a coarse traffic

An Interactive Concert Program Based on Infrared Watermark and Audio Synthesis

Science.gov (United States)

Wang, Hsi-Chun; Lee, Wen-Pin Hope; Liang, Feng-Ju

The objective of this research is to propose a video/audio system which allows the user to listen the typical music notes in the concert program under infrared detection. The system synthesizes audio with different pitches and tempi in accordance with the encoded data in a 2-D barcode embedded in the infrared watermark. The digital halftoning technique has been used to fabricate the infrared watermark composed of halftone dots by both amplitude modulation (AM) and frequency modulation (FM). The results show that this interactive system successfully recognizes the barcode and synthesizes audio under infrared detection of a concert program which is also valid for human observation of the contents. This interactive video/audio system has greatly expanded the capability of the printout paper to audio display and also has many potential value-added applications.

Applications of Scalable Multipoint Video and Audio Using the Public Internet

Directory of Open Access Journals (Sweden)

Robert D. Gaglianello

2000-01-01

Full Text Available This paper describes a scalable multipoint video system, designed for efficient generation and display of high quality, multiple resolution, multiple compressed video streams over IP-based networks. We present our experiences using the system over the public Internet for several “real-world” applications, including distance learning, virtual theater, and virtual collaboration. The trials were a combined effort of Bell Laboratories and the Gertrude Stein Repertory Theatre (TGSRT. We also present current advances in the conferencing system since the trials, new areas for application and future applications.

Online Class Review: Using Streaming-Media Technology

Science.gov (United States)

Loudon, Marc; Sharp, Mark

2006-01-01

We present an automated system that allows students to replay both audio and video from a large nonmajors' organic chemistry class as streaming RealMedia. Once established, this system requires no technical intervention and is virtually transparent to the instructor. This gives students access to online class review at any time. Assessment has…

A heterogeneous multiprocessor architecture for low-power audio signal processing applications

DEFF Research Database (Denmark)

Paker, Ozgun; Sparsø, Jens; Haandbæk, Niels

2001-01-01

. The processors are tailored for different classes of filtering algorithms (FIR, IIR, N-LMS etc.), and in a typical system the communication among processors occurs at the sampling rate only. The processors are parameterized in word-size, memory-size, etc. and can be instantiated according to the needs...... of the application at hand using a normal synthesis based ASIC design flow. To give an impression of the size of a processor we mention that one of the FIR processors in a prototype design has 16 instructions, a 32 word×16 bit program memory, a 64 word×16 bit data memory and a 25 word×16 bit coefficient memory....... Early results obtained from the design of a prototype chip containing filter processors for a hearing aid application, indicate a power consumption that is an order of magnitude better than current state of the art low-power audio DSPs implemented using full-custom techniques. This is due to: (1...

Detection and characterization of lightning-based sources using continuous wavelet transform: application to audio-magnetotellurics

Science.gov (United States)

Larnier, H.; Sailhac, P.; Chambodut, A.

2018-01-01

Atmospheric electromagnetic waves created by global lightning activity contain information about electrical processes of the inner and the outer Earth. Large signal-to-noise ratio events are particularly interesting because they convey information about electromagnetic properties along their path. We introduce a new methodology to automatically detect and characterize lightning-based waves using a time-frequency decomposition obtained through the application of continuous wavelet transform. We focus specifically on three types of sources, namely, atmospherics, slow tails and whistlers, that cover the frequency range 10 Hz to 10 kHz. Each wave has distinguishable characteristics in the time-frequency domain due to source shape and dispersion processes. Our methodology allows automatic detection of each type of event in the time-frequency decomposition thanks to their specific signature. Horizontal polarization attributes are also recovered in the time-frequency domain. This procedure is first applied to synthetic extremely low frequency time-series with different signal-to-noise ratios to test for robustness. We then apply it on real data: three stations of audio-magnetotelluric data acquired in Guadeloupe, oversea French territories. Most of analysed atmospherics and slow tails display linear polarization, whereas analysed whistlers are elliptically polarized. The diversity of lightning activity is finally analysed in an audio-magnetotelluric data processing framework, as used in subsurface prospecting, through estimation of the impedance response functions. We show that audio-magnetotelluric processing results depend mainly on the frequency content of electromagnetic waves observed in processed time-series, with an emphasis on the difference between morning and afternoon acquisition. Our new methodology based on the time-frequency signature of lightning-induced electromagnetic waves allows automatic detection and characterization of events in audio

ATLAS Live: Collaborative Information Streams

Energy Technology Data Exchange (ETDEWEB)

Goldfarb, Steven [Department of Physics, University of Michigan, Ann Arbor, MI 48109 (United States); Collaboration: ATLAS Collaboration

2011-12-23

I report on a pilot project launched in 2010 focusing on facilitating communication and information exchange within the ATLAS Collaboration, through the combination of digital signage software and webcasting. The project, called ATLAS Live, implements video streams of information, ranging from detailed detector and data status to educational and outreach material. The content, including text, images, video and audio, is collected, visualised and scheduled using digital signage software. The system is robust and flexible, utilizing scripts to input data from remote sources, such as the CERN Document Server, Indico, or any available URL, and to integrate these sources into professional-quality streams, including text scrolling, transition effects, inter and intra-screen divisibility. Information is published via the encoding and webcasting of standard video streams, viewable on all common platforms, using a web browser or other common video tool. Authorisation is enforced at the level of the streaming and at the web portals, using the CERN SSO system.

ATLAS Live: Collaborative Information Streams

International Nuclear Information System (INIS)

Goldfarb, Steven

2011-01-01

I report on a pilot project launched in 2010 focusing on facilitating communication and information exchange within the ATLAS Collaboration, through the combination of digital signage software and webcasting. The project, called ATLAS Live, implements video streams of information, ranging from detailed detector and data status to educational and outreach material. The content, including text, images, video and audio, is collected, visualised and scheduled using digital signage software. The system is robust and flexible, utilizing scripts to input data from remote sources, such as the CERN Document Server, Indico, or any available URL, and to integrate these sources into professional-quality streams, including text scrolling, transition effects, inter and intra-screen divisibility. Information is published via the encoding and webcasting of standard video streams, viewable on all common platforms, using a web browser or other common video tool. Authorisation is enforced at the level of the streaming and at the web portals, using the CERN SSO system.

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

International Nuclear Information System (INIS)

George, Rohini; Chung, Theodore D.; Vedam, Sastry S.; Ramakrishnan, Viswanathan; Mohan, Radhe; Weiss, Elisabeth; Keall, Paul J.

2006-01-01

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

Sequential specification of time-aware stream processing applications

NARCIS (Netherlands)

Geuns, S.J.; Hausmans, J.P.H.M.; Bekooij, Marco Jan Gerrit

Automatic parallelization of Nested Loop Programs (NLPs) is an attractive method to create embedded real-time stream processing applications for multi-core systems. However, the description and parallelization of applications with a time dependent functional behavior has not been considered in NLPs.

Method for Reading Sensors and Controlling Actuators Using Audio Interfaces of Mobile Devices

Science.gov (United States)

Aroca, Rafael V.; Burlamaqui, Aquiles F.; Gonçalves, Luiz M. G.

2012-01-01

This article presents a novel closed loop control architecture based on audio channels of several types of computing devices, such as mobile phones and tablet computers, but not restricted to them. The communication is based on an audio interface that relies on the exchange of audio tones, allowing sensors to be read and actuators to be controlled. As an application example, the presented technique is used to build a low cost mobile robot, but the system can also be used in a variety of mechatronics applications and sensor networks, where smartphones are the basic building blocks. PMID:22438726

Method for reading sensors and controlling actuators using audio interfaces of mobile devices.

Science.gov (United States)

Aroca, Rafael V; Burlamaqui, Aquiles F; Gonçalves, Luiz M G

2012-01-01

This article presents a novel closed loop control architecture based on audio channels of several types of computing devices, such as mobile phones and tablet computers, but not restricted to them. The communication is based on an audio interface that relies on the exchange of audio tones, allowing sensors to be read and actuators to be controlled. As an application example, the presented technique is used to build a low cost mobile robot, but the system can also be used in a variety of mechatronics applications and sensor networks, where smartphones are the basic building blocks.

Real-Time Audio Processing on the T-CREST Multicore Platform

DEFF Research Database (Denmark)

Ausin, Daniel Sanz; Pezzarossa, Luca; Schoeberl, Martin

2017-01-01

of the audio signal. This paper presents a real-time multicore audio processing system based on the T-CREST platform. T-CREST is a time-predictable multicore processor for real-time embedded systems. Multiple audio effect tasks have been implemented, which can be connected together in different configurations...... forming sequential and parallel effect chains, and using a network-onchip for intercommunication between processors. The evaluation of the system shows that real-time processing of multiple effect configurations is possible, and that the estimation and control of latency ensures real-time behavior.......Multicore platforms are nowadays widely used for audio processing applications, due to the improvement of computational power that they provide. However, some of these systems are not optimized for temporally constrained environments, which often leads to an undesired increase in the latency...

Four-quadrant flyback converter for direct audio power amplification

Energy Technology Data Exchange (ETDEWEB)

Ljusev, P.; Andersen, Michael A.E.

2005-07-01

This paper presents a bidirectional, four-quadrant yback converter for use in direct audio power amplication. When compared to the standard Class-D switching-mode audio power amplier with separate power supply, the proposed four-quadrant flyback converter provides simple and compact solution with high efciency, higher level of integration, lower component count, less board space and eventually lower cost. Both peak and average current-mode control for use with 4Q flyback power converters are described and compared. Integrated magnetics is presented which simplies the construction of the auxiliary power supplies for control biasing and isolated gate drives. The feasibility of the approach is proven on audio power amplier prototype for subwoofer applications. (au)

The Single- and Multichannel Audio Recordings Database (SMARD)

DEFF Research Database (Denmark)

Nielsen, Jesper Kjær; Jensen, Jesper Rindom; Jensen, Søren Holdt

2014-01-01

A new single- and multichannel audio recordings database (SMARD) is presented in this paper. The database contains recordings from a box-shaped listening room for various loudspeaker and array types. The recordings were made for 48 different configurations of three different loudspeakers and four...... different microphone arrays. In each configuration, 20 different audio segments were played and recorded ranging from simple artificial sounds to polyphonic music. SMARD can be used for testing algorithms developed for numerous application, and we give examples of source localisation results....

Digital signal processor for silicon audio playback devices; Silicon audio saisei kikiyo digital signal processor

Energy Technology Data Exchange (ETDEWEB)

NONE

2000-03-01

The digital audio signal processor (DSP) TC9446F series has been developed silicon audio playback devices with a memory medium of, e.g., flash memory, DVD players, and AV devices, e.g., TV sets. It corresponds to AAC (advanced audio coding) (2ch) and MP3 (MPEG1 Layer3), as the audio compressing techniques being used for transmitting music through an internet. It also corresponds to compressed types, e.g., Dolby Digital, DTS (digital theater system) and MPEG2 audio, being adopted for, e.g., DVDs. It can carry a built-in audio signal processing program, e.g., Dolby ProLogic, equalizer, sound field controlling, and 3D sound. TC9446XB has been lined up anew. It adopts an FBGA (fine pitch ball grid array) package for portable audio devices. (translated by NEDO)

SP@CE - An SP-based programming model for consumer electronics streaming applications

NARCIS (Netherlands)

Varbanescu, Ana Lucia; Nijhuis, Maik; Escribano, Arturo González; Sips, Henk; Bos, Herbert; Bal, Henri

2007-01-01

Efficient programming of multimedia streaming applications for Consumer Electronics (CE) devices is not trivial. As a solution for this problem, we present SP@CE, a novel programming model designed to balance the specific requirements of CE streaming applications with the simplicity and efficiency

Using online handwriting and audio streams for mathematical expressions recognition: a bimodal approach

Science.gov (United States)

Medjkoune, Sofiane; Mouchère, Harold; Petitrenaud, Simon; Viard-Gaudin, Christian

2013-01-01

The work reported in this paper concerns the problem of mathematical expressions recognition. This task is known to be a very hard one. We propose to alleviate the difficulties by taking into account two complementary modalities. The modalities referred to are handwriting and audio ones. To combine the signals coming from both modalities, various fusion methods are explored. Performances evaluated on the HAMEX dataset show a significant improvement compared to a single modality (handwriting) based system.

Automatic processing of CERN video, audio and photo archives

International Nuclear Information System (INIS)

Kwiatek, M

2008-01-01

The digitalization of CERN audio-visual archives, a major task currently in progress, will generate over 40 TB of video, audio and photo files. Storing these files is one issue, but a far more important challenge is to provide long-time coherence of the archive and to make these files available on-line with minimum manpower investment. An infrastructure, based on standard CERN services, has been implemented, whereby master files, stored in the CERN Distributed File System (DFS), are discovered and scheduled for encoding into lightweight web formats based on predefined profiles. Changes in master files, conversion profiles or in the metadata database (read from CDS, the CERN Document Server) are automatically detected and the media re-encoded whenever necessary. The encoding processes are run on virtual servers provided on-demand by the CERN Server Self Service Centre, so that new servers can be easily configured to adapt to higher load. Finally, the generated files are made available from the CERN standard web servers with streaming implemented using Windows Media Services

Automatic processing of CERN video, audio and photo archives

Energy Technology Data Exchange (ETDEWEB)

Kwiatek, M [CERN, Geneva (Switzerland)], E-mail: Michal.Kwiatek@cem.ch

2008-07-15

The digitalization of CERN audio-visual archives, a major task currently in progress, will generate over 40 TB of video, audio and photo files. Storing these files is one issue, but a far more important challenge is to provide long-time coherence of the archive and to make these files available on-line with minimum manpower investment. An infrastructure, based on standard CERN services, has been implemented, whereby master files, stored in the CERN Distributed File System (DFS), are discovered and scheduled for encoding into lightweight web formats based on predefined profiles. Changes in master files, conversion profiles or in the metadata database (read from CDS, the CERN Document Server) are automatically detected and the media re-encoded whenever necessary. The encoding processes are run on virtual servers provided on-demand by the CERN Server Self Service Centre, so that new servers can be easily configured to adapt to higher load. Finally, the generated files are made available from the CERN standard web servers with streaming implemented using Windows Media Services.

Design of low power and low area passive sigma delta modulators for audio applications

CERN Document Server

Fouto, David

2017-01-01

This book presents the study, design, modulation, optimization and implementation of low power, passive DT-ΣΔMs for use in audio applications. The high gain and bandwidth amplifier normally used for integration in ΣΔ modulation, is replaced by passive, switched-capacitor branches working under the Ultra Incomplete Settling (UIS) condition, leading to a reduction of the consumed power. The authors describe a design process that uses high level models and an optimization process based in genetic algorithms to achieve the desired performance.

Low-cost synchronization of high-speed audio and video recordings in bio-acoustic experiments.

Science.gov (United States)

Laurijssen, Dennis; Verreycken, Erik; Geipel, Inga; Daems, Walter; Peremans, Herbert; Steckel, Jan

2018-02-27

In this paper, we present a method for synchronizing high-speed audio and video recordings of bio-acoustic experiments. By embedding a random signal into the recorded video and audio data, robust synchronization of a diverse set of sensor streams can be performed without the need to keep detailed records. The synchronization can be performed using recording devices without dedicated synchronization inputs. We demonstrate the efficacy of the approach in two sets of experiments: behavioral experiments on different species of echolocating bats and the recordings of field crickets. We present the general operating principle of the synchronization method, discuss its synchronization strength and provide insights into how to construct such a device using off-the-shelf components. © 2018. Published by The Company of Biologists Ltd.

Tensorial dynamic time warping with articulation index representation for efficient audio-template learning.

Science.gov (United States)

Le, Long N; Jones, Douglas L

2018-03-01

Audio classification techniques often depend on the availability of a large labeled training dataset for successful performance. However, in many application domains of audio classification (e.g., wildlife monitoring), obtaining labeled data is still a costly and laborious process. Motivated by this observation, a technique is proposed to efficiently learn a clean template from a few labeled, but likely corrupted (by noise and interferences), data samples. This learning can be done efficiently via tensorial dynamic time warping on the articulation index-based time-frequency representations of audio data. The learned template can then be used in audio classification following the standard template-based approach. Experimental results show that the proposed approach outperforms both (1) the recurrent neural network approach and (2) the state-of-the-art in the template-based approach on a wildlife detection application with few training samples.

Advances in audio watermarking based on singular value decomposition

CERN Document Server

Dhar, Pranab Kumar

2015-01-01

This book introduces audio watermarking methods for copyright protection, which has drawn extensive attention for securing digital data from unauthorized copying. The book is divided into two parts. First, an audio watermarking method in discrete wavelet transform (DWT) and discrete cosine transform (DCT) domains using singular value decomposition (SVD) and quantization is introduced. This method is robust against various attacks and provides good imperceptible watermarked sounds. Then, an audio watermarking method in fast Fourier transform (FFT) domain using SVD and Cartesian-polar transformation (CPT) is presented. This method has high imperceptibility and high data payload and it provides good robustness against various attacks. These techniques allow media owners to protect copyright and to show authenticity and ownership of their material in a variety of applications.   ·         Features new methods of audio watermarking for copyright protection and ownership protection ·         Outl...

Perancangan Sistem Audio Mobil Berbasiskan Sistem Pakar dan Web

Directory of Open Access Journals (Sweden)

Djunaidi Santoso

2011-12-01

Full Text Available Designing car audio that fits user’s needs is a fun activity. However, the design often consumes more time and costly since it should be consulted to the experts several times. For easy access to information in designing a car audio system as well as error prevention, an car audio system based on expert system and web is designed for those who do not have sufficient time and expense to consult directly to experts. This system consists of tutorial modules designed using the HyperText Preprocessor (PHP and MySQL as database. This car audio system design is evaluated uses black box testing method which focuses on the functional needs of the application. Tests are performed by providing inputs and produce outputs corresponding to the function of each module. The test results prove the correspondence between input and output, which means that the program meet the initial goals of the design. 

Precision Scaling of Neural Networks for Efficient Audio Processing

OpenAIRE

Ko, Jong Hwan; Fromm, Josh; Philipose, Matthai; Tashev, Ivan; Zarar, Shuayb

2017-01-01

While deep neural networks have shown powerful performance in many audio applications, their large computation and memory demand has been a challenge for real-time processing. In this paper, we study the impact of scaling the precision of neural networks on the performance of two common audio processing tasks, namely, voice-activity detection and single-channel speech enhancement. We determine the optimal pair of weight/neuron bit precision by exploring its impact on both the performance and ...

Categorizing Video Game Audio

DEFF Research Database (Denmark)

Westerberg, Andreas Rytter; Schoenau-Fog, Henrik

2015-01-01

they can use audio in video games. The conclusion of this study is that the current models' view of the diegetic spaces, used to categorize video game audio, is not t to categorize all sounds. This can however possibly be changed though a rethinking of how the player interprets audio.......This paper dives into the subject of video game audio and how it can be categorized in order to deliver a message to a player in the most precise way. A new categorization, with a new take on the diegetic spaces, can be used a tool of inspiration for sound- and game-designers to rethink how...

ATLAS Live: Collaborative Information Streams

CERN Document Server

Goldfarb, S; The ATLAS collaboration

2010-01-01

I report on a pilot project launched in 2010 focusing on facilitating communication and information exchange within the ATLAS Collaboration, through the combination of digital signage software and webcasting. The project, called ATLAS Live, implements video streams of information, ranging from detailed detector and data status to educational and outreach material. The content, including text, images, video and audio, is collected, visualised and scheduled using the SCALA digital signage software system. The system is robust and flexible, allowing for the usage of scripts to input data from remote sources, such as the CERN Document Server, Indico, or any available URL, and to integrate these sources into professional-quality streams, including text scrolling, transition effects, inter and intrascreen divisibility. The video is made available to the collaboration or public through the encoding and webcasting of standard video streams, viewable on all common platforms, using a web browser or other common video t...

Approaches to building single-stage AC/AC conversion switch-mode audio power amplifiers

DEFF Research Database (Denmark)

Ljusev, Petar; Andersen, Michael Andreas E.

2004-01-01

This paper discusses the possible topologies and promising approaches towards direct single-phase AC-AC conversion of the mains voltage for audio applications. When compared to standard Class-D switching audio power amplifiers with a separate power supply, it is expected that direct conversion...

A Joint Audio-Visual Approach to Audio Localization

DEFF Research Database (Denmark)

Jensen, Jesper Rindom; Christensen, Mads Græsbøll

2015-01-01

Localization of audio sources is an important research problem, e.g., to facilitate noise reduction. In the recent years, the problem has been tackled using distributed microphone arrays (DMA). A common approach is to apply direction-of-arrival (DOA) estimation on each array (denoted as nodes), a...... time-of-flight cameras. Moreover, we propose an optimal method for weighting such DOA and range information for audio localization. Our experiments on both synthetic and real data show that there is a clear, potential advantage of using the joint audiovisual localization framework....

Modeling Audio Fingerprints : Structure, Distortion, Capacity

NARCIS (Netherlands)

Doets, P.J.O.

2010-01-01

An audio fingerprint is a compact low-level representation of a multimedia signal. An audio fingerprint can be used to identify audio files or fragments in a reliable way. The use of audio fingerprints for identification consists of two phases. In the enrollment phase known content is fingerprinted,

Theory and Application of Audio-Based Assessment of Cough

Directory of Open Access Journals (Sweden)

Yan Shi

2018-01-01

Full Text Available Cough is a common symptom of many respiratory diseases. Many medical literatures underline that a system for the automatic, objective, and reliable detection of cough events is important and very promising to detect pathology severity in chronic cough disease. In order to track the development status of an audio-based cough monitoring system, we briefly described the history of objective cough detection and then illustrated the cough sound generating principle. The probable endpoints of cough clinical studies, including cough frequency, intensity of coughing, and acoustic properties of cough sound, were analyzed in this paper. Finally, we introduce some successful cough monitoring equipment and their recognition algorithm in detail. It can be obtained that, firstly, acoustic variability of cough sounds within and between individuals makes it difficult to assess the intensity of coughing. Furthermore, now great progress in audio-based cough detection is being made. Moreover, accurate portable objective monitoring systems will be available and widely used in home care and clinical trials in the near future.

Advances in audio source seperation and multisource audio content retrieval

Science.gov (United States)

Vincent, Emmanuel

2012-06-01

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.

Modular Sensor Environment : Audio Visual Industry Monitoring Applications

OpenAIRE

Guillot, Calvin

2017-01-01

This work was made for Electro Waves Oy. The company specializes in Audio-visual services and interactive systems. The purpose of this work is to design and implement a modular sensor environment for the company, which will be used for developing automated systems. This thesis begins with an introduction to sensor systems and their different topologies. It is followed by an introduction to the technologies used in this project. The system is divided in three parts. The client, tha...

Roundtable Audio Discussion

Directory of Open Access Journals (Sweden)

Chris Bigum

2007-01-01

Full Text Available RoundTable on Technology, Teaching and Tools. This is a roundtable audio interview conducted by James Farmer, founder of Edublogs, with Anne Bartlett-Bragg (University of Technology Sydney and Chris Bigum (Deakin University. Skype was used to make and record the audio conference and the resulting sound file was edited by Andrew McLauchlan.

Acoustic Heritage and Audio Creativity: the Creative Application of Sound in the Representation, Understanding and Experience of Past Environments

Directory of Open Access Journals (Sweden)

Damian Murphy

2017-06-01

Full Text Available Acoustic Heritage is one aspect of archaeoacoustics, and refers more specifically to the quantifiable acoustic properties of buildings, sites and landscapes from our architectural and archaeological past, forming an important aspect of our intangible cultural heritage. Auralisation, the audio equivalent of 3D visualisation, enables these acoustic properties, captured via the process of measurement and survey, or computer-based modelling, to form the basis of an audio reconstruction and presentation of the studied space. This article examines the application of auralisation and audio creativity as a means to explore our acoustic heritage, thereby diversifying and enhancing the toolset available to the digital heritage or humanities researcher. The Open Acoustic Impulse Response (OpenAIR library is an online repository for acoustic impulse response and auralisation data, with a significant part having been gathered from a broad range of heritage sites. The methodology used to gather this acoustic data is discussed, together with the processes used in generating and calibrating a comparable computer model, and how the data generated might be analysed and presented. The creative use of this acoustic data is also considered, in the context of music production, mixed media artwork and audio for gaming. More relevant to digital heritage is how these data can be used to create new experiences of past environments, as information, interpretation, guide or artwork and ultimately help to articulate new research questions and explorations of our acoustic heritage.

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

Science.gov (United States)

Wood, Paul; Sinton, David

2010-08-01

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.

Design of batch audio/video conversion platform based on JavaEE

Science.gov (United States)

Cui, Yansong; Jiang, Lianpin

2018-03-01

With the rapid development of digital publishing industry, the direction of audio / video publishing shows the diversity of coding standards for audio and video files, massive data and other significant features. Faced with massive and diverse data, how to quickly and efficiently convert to a unified code format has brought great difficulties to the digital publishing organization. In view of this demand and present situation in this paper, basing on the development architecture of Sptring+SpringMVC+Mybatis, and combined with the open source FFMPEG format conversion tool, a distributed online audio and video format conversion platform with a B/S structure is proposed. Based on the Java language, the key technologies and strategies designed in the design of platform architecture are analyzed emphatically in this paper, designing and developing a efficient audio and video format conversion system, which is composed of “Front display system”, "core scheduling server " and " conversion server ". The test results show that, compared with the ordinary audio and video conversion scheme, the use of batch audio and video format conversion platform can effectively improve the conversion efficiency of audio and video files, and reduce the complexity of the work. Practice has proved that the key technology discussed in this paper can be applied in the field of large batch file processing, and has certain practical application value.

Benchmarking Distributed Stream Processing Platforms for IoT Applications

OpenAIRE

Shukla, Anshu; Simmhan, Yogesh

2016-01-01

Internet of Things (IoT) is a technology paradigm where millions of sensors monitor, and help inform or manage, physical, envi- ronmental and human systems in real-time. The inherent closed-loop re- sponsiveness and decision making of IoT applications makes them ideal candidates for using low latency and scalable stream processing plat- forms. Distributed Stream Processing Systems (DSPS) are becoming es- sential components of any IoT stack, but the efficacy and performance of contemporary DSP...

Audio Restoration

Science.gov (United States)

Esquef, Paulo A. A.

The first reproducible recording of human voice was made in 1877 on a tinfoil cylinder phonograph devised by Thomas A. Edison. Since then, much effort has been expended to find better ways to record and reproduce sounds. By the mid-1920s, the first electrical recordings appeared and gradually took over purely acoustic recordings. The development of electronic computers, in conjunction with the ability to record data onto magnetic or optical media, culminated in the standardization of compact disc format in 1980. Nowadays, digital technology is applied to several audio applications, not only to improve the quality of modern and old recording/reproduction techniques, but also to trade off sound quality for less storage space and less taxing transmission capacity requirements.

Application of the Hydroecological Integrity Assessment Process for Missouri Streams

Science.gov (United States)

Kennen, Jonathan G.; Henriksen, James A.; Heasley, John; Cade, Brian S.; Terrell, James W.

2009-01-01

Natural flow regime concepts and theories have established the justification for maintaining or restoring the range of natural hydrologic variability so that physiochemical processes, native biodiversity, and the evolutionary potential of aquatic and riparian assemblages can be sustained. A synthesis of recent research advances in hydroecology, coupled with stream classification using hydroecologically relevant indices, has produced the Hydroecological Integrity Assessment Process (HIP). HIP consists of (1) a regional classification of streams into hydrologic stream types based on flow data from long-term gaging-station records for relatively unmodified streams, (2) an identification of stream-type specific indices that address 11 subcomponents of the flow regime, (3) an ability to establish environmental flow standards, (4) an evaluation of hydrologic alteration, and (5) a capacity to conduct alternative analyses. The process starts with the identification of a hydrologic baseline (reference condition) for selected locations, uses flow data from a stream-gage network, and proceeds to classify streams into hydrologic stream types. Concurrently, the analysis identifies a set of non-redundant and ecologically relevant hydrologic indices for 11 subcomponents of flow for each stream type. Furthermore, regional hydrologic models for synthesizing flow conditions across a region and the development of flow-ecology response relations for each stream type can be added to further enhance the process. The application of HIP to Missouri streams identified five stream types ((1) intermittent, (2) perennial runoff-flashy, (3) perennial runoff-moderate baseflow, (4) perennial groundwater-stable, and (5) perennial groundwater-super stable). Two Missouri-specific computer software programs were developed: (1) a Missouri Hydrologic Assessment Tool (MOHAT) which is used to establish a hydrologic baseline, provide options for setting environmental flow standards, and compare past and

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

Science.gov (United States)

Romero-Fresco, Pablo; Fryer, Louise

2013-01-01

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…

Bringing Legacy Visualization Software to Modern Computing Devices via Application Streaming

Science.gov (United States)

Fisher, Ward

2014-05-01

Planning software compatibility across forthcoming generations of computing platforms is a problem commonly encountered in software engineering and development. While this problem can affect any class of software, data analysis and visualization programs are particularly vulnerable. This is due in part to their inherent dependency on specialized hardware and computing environments. A number of strategies and tools have been designed to aid software engineers with this task. While generally embraced by developers at 'traditional' software companies, these methodologies are often dismissed by the scientific software community as unwieldy, inefficient and unnecessary. As a result, many important and storied scientific software packages can struggle to adapt to a new computing environment; for example, one in which much work is carried out on sub-laptop devices (such as tablets and smartphones). Rewriting these packages for a new platform often requires significant investment in terms of development time and developer expertise. In many cases, porting older software to modern devices is neither practical nor possible. As a result, replacement software must be developed from scratch, wasting resources better spent on other projects. Enabled largely by the rapid rise and adoption of cloud computing platforms, 'Application Streaming' technologies allow legacy visualization and analysis software to be operated wholly from a client device (be it laptop, tablet or smartphone) while retaining full functionality and interactivity. It mitigates much of the developer effort required by other more traditional methods while simultaneously reducing the time it takes to bring the software to a new platform. This work will provide an overview of Application Streaming and how it compares against other technologies which allow scientific visualization software to be executed from a remote computer. We will discuss the functionality and limitations of existing application streaming

Location audio simplified capturing your audio and your audience

CERN Document Server

Miles, Dean

2014-01-01

From the basics of using camera, handheld, lavalier, and shotgun microphones to camera calibration and mixer set-ups, Location Audio Simplified unlocks the secrets to clean and clear broadcast quality audio no matter what challenges you face. Author Dean Miles applies his twenty-plus years of experience as a professional location operator to teach the skills, techniques, tips, and secrets needed to produce high-quality production sound on location. Humorous and thoroughly practical, the book covers a wide array of topics, such as:* location selection* field mixing* boo

Efficient Estimation of Dynamic Density Functions with Applications in Streaming Data

KAUST Repository

Qahtan, Abdulhakim

2016-05-11

Recent advances in computing technology allow for collecting vast amount of data that arrive continuously in the form of streams. Mining data streams is challenged by the speed and volume of the arriving data. Furthermore, the underlying distribution of the data changes over the time in unpredicted scenarios. To reduce the computational cost, data streams are often studied in forms of condensed representation, e.g., Probability Density Function (PDF). This thesis aims at developing an online density estimator that builds a model called KDE-Track for characterizing the dynamic density of the data streams. KDE-Track estimates the PDF of the stream at a set of resampling points and uses interpolation to estimate the density at any given point. To reduce the interpolation error and computational complexity, we introduce adaptive resampling where more/less resampling points are used in high/low curved regions of the PDF. The PDF values at the resampling points are updated online to provide up-to-date model of the data stream. Comparing with other existing online density estimators, KDE-Track is often more accurate (as reflected by smaller error values) and more computationally efficient (as reflected by shorter running time). The anytime available PDF estimated by KDE-Track can be applied for visualizing the dynamic density of data streams, outlier detection and change detection in data streams. In this thesis work, the first application is to visualize the taxi traffic volume in New York city. Utilizing KDE-Track allows for visualizing and monitoring the traffic flow on real time without extra overhead and provides insight analysis of the pick up demand that can be utilized by service providers to improve service availability. The second application is to detect outliers in data streams from sensor networks based on the estimated PDF. The method detects outliers accurately and outperforms baseline methods designed for detecting and cleaning outliers in sensor data. The

Structure Learning in Audio

DEFF Research Database (Denmark)

Nielsen, Andreas Brinch

By having information about the setting a user is in, a computer is able to make decisions proactively to facilitate tasks for the user. Two approaches are taken in this thesis to achieve more information about an audio environment. One approach is that of classifying audio, and a new approach...... investigated. A fast and computationally simple approach that compares recordings and classifies if they are from the same audio environment have been developed, and shows very high accuracy and the ability to synchronize recordings in the case of recording devices which are not connected. A more general model...

Migrating Home Computer Audio Waveforms to Digital Objects: A Case Study on Digital Archaeology

Directory of Open Access Journals (Sweden)

Mark Guttenbrunner

2011-03-01

Full Text Available Rescuing data from inaccessible or damaged storage media for the purpose of preserving the digital data for the long term is one of the dimensions of digital archaeology. With the current pace of technological development, any system can become obsolete in a matter of years and hence the data stored in a specific storage media might not be accessible anymore due to the unavailability of the system to access the media. In order to preserve digital records residing in such storage media, it is necessary to extract the data stored in those media by some means.One early storage medium for home computers in the 1980s was audio tape. The first home computer systems allowed the use of standard cassette players to record and replay data. Audio cassettes are more durable than old home computers when properly stored. Devices playing this medium (i.e. tape recorders can be found in working condition or can be repaired, as they are usually made out of standard components. By re-engineering the format of the waveform and the file formats, the data on such media can then be extracted from a digitised audio stream and migrated to a non-obsolete format.In this paper we present a case study on extracting the data stored on an audio tape by an early home computer system, namely the Philips Videopac+ G7400. The original data formats were re-engineered and an application was written to support the migration of the data stored on tapes without using the original system. This eliminates the necessity of keeping an obsolete system alive for enabling access to the data on the storage media meant for this system. Two different methods to interpret the data and eliminate possible errors in the tape were implemented and evaluated on original tapes, which were recorded 20 years ago. Results show that with some error correction methods, parts of the tapes are still readable even without the original system. It also implies that it is easier to build solutions while original

Sequential Specification of Time-aware Stream Processing Applications (Extended Abstract)

NARCIS (Netherlands)

Geuns, S.J.; Hausmans, J.P.H.M.; Bekooij, Marco Jan Gerrit

2012-01-01

Automatic parallelization of Nested Loop Programs (NLPs) is an attractive method to create embedded real-time stream processing applications for multi-core systems. However, the description and parallelization of applications with a time dependent functional behavior has not been considered in NLPs.

Instrumental Landing Using Audio Indication

Science.gov (United States)

Burlak, E. A.; Nabatchikov, A. M.; Korsun, O. N.

2018-02-01

The paper proposes an audio indication method for presenting to a pilot the information regarding the relative positions of an aircraft in the tasks of precision piloting. The implementation of the method is presented, the use of such parameters of audio signal as loudness, frequency and modulation are discussed. To confirm the operability of the audio indication channel the experiments using modern aircraft simulation facility were carried out. The simulated performed the instrument landing using the proposed audio method to indicate the aircraft deviations in relation to the slide path. The results proved compatible with the simulated instrumental landings using the traditional glidescope pointers. It inspires to develop the method in order to solve other precision piloting tasks.

Bit rates in audio source coding

NARCIS (Netherlands)

Veldhuis, Raymond N.J.

1992-01-01

The goal is to introduce and solve the audio coding optimization problem. Psychoacoustic results such as masking and excitation pattern models are combined with results from rate distortion theory to formulate the audio coding optimization problem. The solution of the audio optimization problem is a

Integration of Geographical Information Systems and Geophysical Applications with Distributed Computing Technologies.

Science.gov (United States)

Pierce, M. E.; Aktas, M. S.; Aydin, G.; Fox, G. C.; Gadgil, H.; Sayar, A.

2005-12-01

We examine the application of Web Service Architectures and Grid-based distributed computing technologies to geophysics and geo-informatics. We are particularly interested in the integration of Geographical Information System (GIS) services with distributed data mining applications. GIS services provide the general purpose framework for building archival data services, real time streaming data services, and map-based visualization services that may be integrated with data mining and other applications through the use of distributed messaging systems and Web Service orchestration tools. Building upon on our previous work in these areas, we present our current research efforts. These include fundamental investigations into increasing XML-based Web service performance, supporting real time data streams, and integrating GIS mapping tools with audio/video collaboration systems for shared display and annotation.

Implementing Audio-CASI on Windows’ Platforms

Science.gov (United States)

Cooley, Philip C.; Turner, Charles F.

2011-01-01

Audio computer-assisted self interviewing (Audio-CASI) technologies have recently been shown to provide important and sometimes dramatic improvements in the quality of survey measurements. This is particularly true for measurements requiring respondents to divulge highly sensitive information such as their sexual, drug use, or other sensitive behaviors. However, DOS-based Audio-CASI systems that were designed and adopted in the early 1990s have important limitations. Most salient is the poor control they provide for manipulating the video presentation of survey questions. This article reports our experiences adapting Audio-CASI to Microsoft Windows 3.1 and Windows 95 platforms. Overall, our Windows-based system provided the desired control over video presentation and afforded other advantages including compatibility with a much wider array of audio devices than our DOS-based Audio-CASI technologies. These advantages came at the cost of increased system requirements --including the need for both more RAM and larger hard disks. While these costs will be an issue for organizations converting large inventories of PCS to Windows Audio-CASI today, this will not be a serious constraint for organizations and individuals with small inventories of machines to upgrade or those purchasing new machines today. PMID:22081743

Stream processing health card application.

Science.gov (United States)

Polat, Seda; Gündem, Taflan Imre

2012-10-01

In this paper, we propose a data stream management system embedded to a smart card for handling and storing user specific summaries of streaming data coming from medical sensor measurements and/or other medical measurements. The data stream management system that we propose for a health card can handle the stream data rates of commonly known medical devices and sensors. It incorporates a type of context awareness feature that acts according to user specific information. The proposed system is cheap and provides security for private data by enhancing the capabilities of smart health cards. The stream data management system is tested on a real smart card using both synthetic and real data.

Estudio del streaming de audio y vídeo sobre redes heterogéneas

OpenAIRE

Gómez Cruz, María del Carmen

2009-01-01

Las operadoras han encontrado nichos de negocio en la integración de múltiples servicios avanzados, como pueden ser Voz sobre IP, Vídeos bajo demanda, datos de alta capacidad, distribución de televisión de alta definición, etc. Esto supone una adaptación constante de sus redes de comunicación de banda ancha para soportar mayores anchos de banda, con mayor alcance, con menores pérdidas, en definitiva con mayores prestaciones; y promueve el desarrollo por parte de los fabricantes de audio y víd...

∑∆ Modulator System-Level Considerations for Hearing-Aid Audio Class-D Output Stage Application

DEFF Research Database (Denmark)

Pracný, Peter; Bruun, Erik

2012-01-01

This paper deals with a system-level design of a digital sigma-delta (∑∆) modulator for hearing-aid audio Class D output stage application. The aim of this paper is to provide a thorough discussion on various possibilities and tradeoffs of ∑∆ modulator system-level design parameter combinations...... - order, oversampling ratio (OSR) and number of bits in the quantizer - including their impact on interpolation filter design as well. The system is kept in digital domain up to the input of the Class D power stage including the digital pulse width modulation (DPWM) block. Notes on the impact of the DPWM...

The application and testing of diatom-based indices of stream water ...

African Journals Online (AJOL)

The application and testing of diatom-based indices of stream water quality in Chinhoyi Town, Zimbabwe. ... PROMOTING ACCESS TO AFRICAN RESEARCH ... test the applicability of foreign diatom-based water quality assessment indices to ...

Audio wiring guide how to wire the most popular audio and video connectors

CERN Document Server

Hechtman, John

2012-01-01

Whether you're a pro or an amateur, a musician or into multimedia, you can't afford to guess about audio wiring. The Audio Wiring Guide is a comprehensive, easy-to-use guide that explains exactly what you need to know. No matter the size of your wiring project or installation, this handy tool provides you with the essential information you need and the techniques to use it. Using The Audio Wiring Guide is like having an expert at your side. By following the clear, step-by-step directions, you can do professional-level work at a fraction of the cost.

Automatic dataflow model extraction from modal real-time stream processing applications

NARCIS (Netherlands)

Geuns, S.J.; Hausmans, J.P.H.M.; Bekooij, Marco Jan Gerrit

2013-01-01

Many real-time stream processing applications are initially described as a sequential application containing while-loops, which execute for an unknown number of iterations. These modal applications have to be executed in parallel on an MPSoC system in order to meet their real-time throughput

Comparative evaluation of audio and audio - tactile methods to improve oral hygiene status of visually impaired school children

OpenAIRE

R Krishnakumar; Swarna Swathi Silla; Sugumaran K Durai; Mohan Govindarajan; Syed Shaheed Ahamed; Logeshwari Mathivanan

2016-01-01

Background: Visually impaired children are unable to maintain good oral hygiene, as their tactile abilities are often underdeveloped owing to their visual disturbances. Conventional brushing techniques are often poorly comprehended by these children and hence, it was decided to evaluate the effectiveness of audio and audio-tactile methods in improving the oral hygiene of these children. Objective: To evaluate and compare the effectiveness of audio and audio-tactile methods in improving oral h...

Efficient Estimation of Dynamic Density Functions with Applications in Streaming Data

KAUST Repository

Qahtan, Abdulhakim Ali Ali

2016-01-01

application is to detect outliers in data streams from sensor networks based on the estimated PDF. The method detects outliers accurately and outperforms baseline methods designed for detecting and cleaning outliers in sensor data. The third application

Audio Frequency Analysis in Mobile Phones

Science.gov (United States)

Aguilar, Horacio Munguía

2016-01-01

A new experiment using mobile phones is proposed in which its audio frequency response is analyzed using the audio port for inputting external signal and getting a measurable output. This experiment shows how the limited audio bandwidth used in mobile telephony is the main cause of the poor speech quality in this service. A brief discussion is…

Presence and the utility of audio spatialization

DEFF Research Database (Denmark)

Bormann, Karsten

2005-01-01

The primary concern of this paper is whether the utility of audio spatialization, as opposed to the fidelity of audio spatialization, impacts presence. An experiment is reported that investigates the presence-performance relationship by decoupling spatial audio fidelity (realism) from task...... performance by varying the spatial fidelity of the audio independently of its relevance to performance on the search task that subjects were to perform. This was achieved by having conditions in which subjects searched for a music-playing radio (an active sound source) and having conditions in which...... supplied only nonattenuated audio was detrimental to performance. Even so, this group of subjects consistently had the largest increase in presence scores over the baseline experiment. Further, the Witmer and Singer (1998) presence questionnaire was more sensitive to whether the audio source was active...

Audio scene segmentation for video with generic content

Science.gov (United States)

Niu, Feng; Goela, Naveen; Divakaran, Ajay; Abdel-Mottaleb, Mohamed

2008-01-01

In this paper, we present a content-adaptive audio texture based method to segment video into audio scenes. The audio scene is modeled as a semantically consistent chunk of audio data. Our algorithm is based on "semantic audio texture analysis." At first, we train GMM models for basic audio classes such as speech, music, etc. Then we define the semantic audio texture based on those classes. We study and present two types of scene changes, those corresponding to an overall audio texture change and those corresponding to a special "transition marker" used by the content creator, such as a short stretch of music in a sitcom or silence in dramatic content. Unlike prior work using genre specific heuristics, such as some methods presented for detecting commercials, we adaptively find out if such special transition markers are being used and if so, which of the base classes are being used as markers without any prior knowledge about the content. Our experimental results show that our proposed audio scene segmentation works well across a wide variety of broadcast content genres.

New musical organology : the audio-games

OpenAIRE

Zénouda , Hervé

2012-01-01

International audience; This article aims to shed light on a new and emerging creative field: " Audio Games, " a crossroad between video games and computer music. Today, a plethora of tiny applications, which propose entertaining audiovisual experiences with a preponderant sound dimension, are available for game consoles, computers, and mobile phones. These experiences represent a new universe where the gameplay of video games is applied to musical composition, hence creating new links betwee...

Design of a WAV audio player based on K20

Directory of Open Access Journals (Sweden)

Xu Yu

2016-01-01

Full Text Available The designed player uses the Freescale Company’s MK20DX128VLH7 as the core control ship, and its hardware platform is equipped with VS1003 audio decoder, OLED display interface, USB interface and SD card slot. The player uses the open source embedded real-time operating system μC/OS-II, Freescale USB Stack V4.1.1 and FATFS, and a graphical user interface is developed to improve the user experience based on CGUI. In general, the designed WAV audio player has a strong applicability and a good practical value.

Audio segmentation of broadcast news in the Albayzin-2010 evaluation: overview, results, and discussion

Directory of Open Access Journals (Sweden)

Butko Taras

2011-01-01

Full Text Available Abstract Recently, audio segmentation has attracted research interest because of its usefulness in several applications like audio indexing and retrieval, subtitling, monitoring of acoustic scenes, etc. Moreover, a previous audio segmentation stage may be useful to improve the robustness of speech technologies like automatic speech recognition and speaker diarization. In this article, we present the evaluation of broadcast news audio segmentation systems carried out in the context of the Albayzín-2010 evaluation campaign. That evaluation consisted of segmenting audio from the 3/24 Catalan TV channel into five acoustic classes: music, speech, speech over music, speech over noise, and the other. The evaluation results displayed the difficulty of this segmentation task. In this article, after presenting the database and metric, as well as the feature extraction methods and segmentation techniques used by the submitted systems, the experimental results are analyzed and compared, with the aim of gaining an insight into the proposed solutions, and looking for directions which are promising.

Digital audio watermarking fundamentals, techniques and challenges

CERN Document Server

Xiang, Yong; Yan, Bin

2017-01-01

This book offers comprehensive coverage on the most important aspects of audio watermarking, from classic techniques to the latest advances, from commonly investigated topics to emerging research subdomains, and from the research and development achievements to date, to current limitations, challenges, and future directions. It also addresses key topics such as reversible audio watermarking, audio watermarking with encryption, and imperceptibility control methods. The book sets itself apart from the existing literature in three main ways. Firstly, it not only reviews classical categories of audio watermarking techniques, but also provides detailed descriptions, analysis and experimental results of the latest work in each category. Secondly, it highlights the emerging research topic of reversible audio watermarking, including recent research trends, unique features, and the potentials of this subdomain. Lastly, the joint consideration of audio watermarking and encryption is also reviewed. With the help of this...

Approaches to building single-stage AC/AC conversion switch-mode audio power amplifiers

Energy Technology Data Exchange (ETDEWEB)

Ljusev, P.; Andersen, Michael A.E.

2005-07-01

This paper discusses the possible topologies and promising approaches towards direct single-phase AC-AC conversion of the mains voltage for audio applications. When compared to standard Class-D switching audio power amplifiers with a separate power supply, it is expected that direct conversion will provide better efficiency and higher level of integration, leading to lower component count, volume and cost, but at the expense of a minor performance deterioration. (au)

Carrier Distortion in Hysteretic Self-Oscillating Class-D Audio Power

DEFF Research Database (Denmark)

Høyerby, Mikkel Christian Kofod; Andersen, Michael A. E.

2009-01-01

An important distortion mechanism in hysteretic self-oscillating (SO) class-D (switch mode) power amplifiers-–carrier distortion-–is analyzed and an optimization method is proposed. This mechanism is an issue in any power amplifier application where a high degree of proportionality between input...... and output is required, such as in audio power amplifiers or xDSL drivers. From an average-mode point of view, carrier distortion is shown to be caused by nonlinear variation of the hysteretic comparator input average voltage with the output average voltage. This easily causes total harmonic distortion...... figures in excess of 0.1–0.2%, inadequate for high-quality audio applications. Carrier distortion is shown to be minimized when the feedback system is designed to provide a triangular carrier (sliding) signal at the input of a hysteretic comparator. The proposed optimization method is experimentally...

Semantic Context Detection Using Audio Event Fusion

Directory of Open Access Journals (Sweden)

Cheng Wen-Huang

2006-01-01

Full Text Available Semantic-level content analysis is a crucial issue in achieving efficient content retrieval and management. We propose a hierarchical approach that models audio events over a time series in order to accomplish semantic context detection. Two levels of modeling, audio event and semantic context modeling, are devised to bridge the gap between physical audio features and semantic concepts. In this work, hidden Markov models (HMMs are used to model four representative audio events, that is, gunshot, explosion, engine, and car braking, in action movies. At the semantic context level, generative (ergodic hidden Markov model and discriminative (support vector machine (SVM approaches are investigated to fuse the characteristics and correlations among audio events, which provide cues for detecting gunplay and car-chasing scenes. The experimental results demonstrate the effectiveness of the proposed approaches and provide a preliminary framework for information mining by using audio characteristics.

Distortion Estimation in Compressed Music Using Only Audio Fingerprints

NARCIS (Netherlands)

Doets, P.J.O.; Lagendijk, R.L.

2008-01-01

An audio fingerprint is a compact yet very robust representation of the perceptually relevant parts of an audio signal. It can be used for content-based audio identification, even when the audio is severely distorted. Audio compression changes the fingerprint slightly. We show that these small

Single conversion audio amplifier and DC-AC converters with high performance and low complexity control scheme

DEFF Research Database (Denmark)

Poulsen, Søren; Andersen, Michael Andreas E.

2004-01-01

This paper proposes a novel control topology for a mains isolated single conversion audio amplifier and DC-AC converters. The topology is made for use in audio applications, and differs from prior art in terms of significantly reduced distortion as well as lower system complexity. The topology can...

Detecting double compression of audio signal

Science.gov (United States)

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

2010-01-01

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.

Quality models for audiovisual streaming

Science.gov (United States)

Thang, Truong Cong; Kim, Young Suk; Kim, Cheon Seog; Ro, Yong Man

2006-01-01

Quality is an essential factor in multimedia communication, especially in compression and adaptation. Quality metrics can be divided into three categories: within-modality quality, cross-modality quality, and multi-modality quality. Most research has so far focused on within-modality quality. Moreover, quality is normally just considered from the perceptual perspective. In practice, content may be drastically adapted, even converted to another modality. In this case, we should consider the quality from semantic perspective as well. In this work, we investigate the multi-modality quality from the semantic perspective. To model the semantic quality, we apply the concept of "conceptual graph", which consists of semantic nodes and relations between the nodes. As an typical of multi-modality example, we focus on audiovisual streaming service. Specifically, we evaluate the amount of information conveyed by a audiovisual content where both video and audio channels may be strongly degraded, even audio are converted to text. In the experiments, we also consider the perceptual quality model of audiovisual content, so as to see the difference with semantic quality model.

Quality-aware media scheduling on MPSoC platforms

DEFF Research Database (Denmark)

Gangadharan, Deepak; Chakraborty, Samarjit; Zimmermann, Roger

2013-01-01

and personal computers by providing maximum quality of service to the multiple streams, it is a difficult task in devices with resource constraints. In order to efficiently utilize the resources, it is essential to derive the necessary processor cycles for multiple video streams such that they are displayed...... be scheduled such that a prespecified quality constraint is satisfied with the available service. We present this framework in the context of a PiP application, but it is applicable in general for multiple media streams. The results obtained using the formal framework were further verified using experiments......Applications that stream multiple video/audio or video+audio clips are being implemented in embedded devices. A Picture-in-Picture (PiP) application is one such application scenario, where two videos are played simultaneously. Although the PiP application is very efficiently handled in televisions...

Elicitation of attributes for the evaluation of audio-on audio-interference

DEFF Research Database (Denmark)

Francombe, Jon; Mason, R.; Dewhirst, M.

2014-01-01

procedure was used to reduce these phrases into a comprehensive set of attributes. Groups of experienced and inexperienced listeners determined nine and eight attributes, respectively. These attribute sets were combined by the listeners to produce a final set of 12 attributes: masking, calming, distraction......An experiment to determine the perceptual attributes of the experience of listening to a target audio program in the presence of an audio interferer was performed. The first stage was a free elicitation task in which a total of 572 phrases were produced. In the second stage, a consensus vocabulary...

Audio watermarking robust against D/A and A/D conversions

Directory of Open Access Journals (Sweden)

Xiang Shijun

2011-01-01

Full Text Available Abstract Digital audio watermarking robust against digital-to-analog (D/A and analog-to-digital (A/D conversions is an important issue. In a number of watermark application scenarios, D/A and A/D conversions are involved. In this article, we first investigate the degradation due to DA/AD conversions via sound cards, which can be decomposed into volume change, additional noise, and time-scale modification (TSM. Then, we propose a solution for DA/AD conversions by considering the effect of the volume change, additional noise and TSM. For the volume change, we introduce relation-based watermarking method by modifying groups of the energy relation of three adjacent DWT coefficient sections. For the additional noise, we pick up the lowest-frequency coefficients for watermarking. For the TSM, the synchronization technique (with synchronization codes and an interpolation processing operation is exploited. Simulation tests show the proposed audio watermarking algorithm provides a satisfactory performance to DA/AD conversions and those common audio processing manipulations.

Decentralized Cloud Method For Multicasting Media Stream

Directory of Open Access Journals (Sweden)

D M B N Bandara

2015-08-01

Full Text Available With the advancement of Information technology the concept of idea sharing has advanced. Mostly on presentations personal computer and projector have become essentials. But on most occasions for connecting these equipment cables and physical devices are used. This is inefficient and time consuming. If a problem occurs someone with technical knowledge is necessary to solve the situation. The objective of this research is to use the wireless technology to reduce the manual configuration and build up a platform where one can easily share files a visuals media and feedback. A system has been developed to detect all the devices over a network and upon granted permission will share video audio and access controls. Final outcome of the research was a collaborative software bundle which work together on a network. One part of the system is a Desktop Network Software. And other is a Mobile Application. Desktop application can detect all other devices in the network which provides the same facility and if required can allocate a group and share its screen files and have a message stream to each device using multicasting. Mobile application can act as a mobile remote to the host computer of the group which can detect any input from user and pass it to the system.

APLIKASI MEDIA AUDIO-VISUAL DALAM PEMBELAJARAN SPEAKING SKILL DENGAN PENDEKATAN AUDIOLINGUAL: Studi Kasus di MAN Batang

Directory of Open Access Journals (Sweden)

Slamet Untung

2012-10-01

Full Text Available The research to study the application of audio and visual medium in order to learn speaking skill by audiolingual approach is a good contribution to educational world of senior high school and the Islamic one, particularly, in finding a way to improving the learning component relating directly to the medium and method of learning speaking skill. This research is to find out its significance and relevance. The main variable of this research includes the whole activities of the application of audio and visual medium in learning speaking skill by audio-lingual approach. The data were collected through observation, interview, questionnaire and documentation. This research took place in state Islamic senior high school of Batang in Central Java. The result shows that the application helps the students to speak English correctly and accurately and stresses the message of the speaking skill learning.

CERN automatic audio-conference service

CERN Multimedia

Sierra Moral, R

2009-01-01

Scientists from all over the world need to collaborate with CERN on a daily basis. They must be able to communicate effectively on their joint projects at any time; as a result telephone conferences have become indispensable and widely used. Managed by 6 operators, CERN already has more than 20000 hours and 5700 audio-conferences per year. However, the traditional telephone based audio-conference system needed to be modernized in three ways. Firstly, to provide the participants with more autonomy in the organization of their conferences; secondly, to eliminate the constraints of manual intervention by operators; and thirdly, to integrate the audio-conferences into a collaborative working framework. The large number, and hence cost, of the conferences prohibited externalization and so the CERN telecommunications team drew up a specification to implement a new system. It was decided to use a new commercial collaborative audio-conference solution based on the SIP protocol. The system was tested as the first Euro...

CERN automatic audio-conference service

CERN Document Server

Sierra Moral, R

2010-01-01

Scientists from all over the world need to collaborate with CERN on a daily basis. They must be able to communicate effectively on their joint projects at any time; as a result telephone conferences have become indispensable and widely used. Managed by 6 operators, CERN already has more than 20000 hours and 5700 audio-conferences per year. However, the traditional telephone based audio-conference system needed to be modernized in three ways. Firstly, to provide the participants with more autonomy in the organization of their conferences; secondly, to eliminate the constraints of manual intervention by operators; and thirdly, to integrate the audio-conferences into a collaborative working framework. The large number, and hence cost, of the conferences prohibited externalization and so the CERN telecommunications team drew up a specification to implement a new system. It was decided to use a new commercial collaborative audio-conference solution based on the SIP protocol. The system was tested as the first Euro...

Debugging of Class-D Audio Power Amplifiers

DEFF Research Database (Denmark)

Crone, Lasse; Pedersen, Jeppe Arnsdorf; Mønster, Jakob Døllner

2012-01-01

Determining and optimizing the performance of a Class-D audio power amplier can be very dicult without knowledge of the use of audio performance measuring equipment and of how the various noise and distortion sources in uence the audio performance. This paper gives an introduction on how to measure...

ANALYSIS OF MULTIMODAL FUSION TECHNIQUES FOR AUDIO-VISUAL SPEECH RECOGNITION

Directory of Open Access Journals (Sweden)

D.V. Ivanko

2016-05-01

Full Text Available The paper deals with analytical review, covering the latest achievements in the field of audio-visual (AV fusion (integration of multimodal information. We discuss the main challenges and report on approaches to address them. One of the most important tasks of the AV integration is to understand how the modalities interact and influence each other. The paper addresses this problem in the context of AV speech processing and speech recognition. In the first part of the review we set out the basic principles of AV speech recognition and give the classification of audio and visual features of speech. Special attention is paid to the systematization of the existing techniques and the AV data fusion methods. In the second part we provide a consolidated list of tasks and applications that use the AV fusion based on carried out analysis of research area. We also indicate used methods, techniques, audio and video features. We propose classification of the AV integration, and discuss the advantages and disadvantages of different approaches. We draw conclusions and offer our assessment of the future in the field of AV fusion. In the further research we plan to implement a system of audio-visual Russian continuous speech recognition using advanced methods of multimodal fusion.

HydroCloud: A Web Application for Exploring Stream Gage Data

Directory of Open Access Journals (Sweden)

Martin C. Roberge

2017-08-01

Full Text Available HydroCloud (hydrocloud.org is a mobile-friendly web application for visually browsing hydrology data from multiple sources. Data providers such as the US Geological Survey (USGS and the German 'Wasserstraßen- und Schifffahrtsverwaltung des Bundes' (WSV currently serve stream discharge data from more than 10,000 stream gages around the world. HydroCloud allows users to plot these data while out in the field, while also providing contextual information such as the current NEXRAD weather imagery or descriptive information about the stream gage and its watershed. Additional features include a chat mechanism for contacting developers, and the use of local storage for saving data.   Funding Statement: This project was supported in part by a grant from the Towson University School of Emerging Technology.

Application of piezodetectors for diagnostics of pulsed and quasi-steady-state plasma streams

Energy Technology Data Exchange (ETDEWEB)

Bandura, A.N.; Chebotarev, V.V.; Garkusha, I.E.; Tereshin, V.I.; Ladygina, M.S. [NSC KIPT, Kharkov (Ukraine). Inst. of Plasma Physics

2006-04-15

The paper reports on studies of the plasma streams generated by two experimental devices: the quasi-steady-state plasma accelerator (QSPA) Kh-50 and the pulsed plasma gun PROSVET. The radial distributions of the plasma pressure for different times and varied distances from the accelerator output have been used for investigation of the plasma stream dynamics and study the plasma compression in the focus region for different operational regimes of plasma accelerators. In experiments for the application of pulsed plasma streams for surface modification of different industrial steels, optimal regimes of surface processing have been chosen on the basis of the plasma pressure measurements. Examples of application of the piezodetectors in simulation experiments on plasma surface interaction under high heat loads are presented.

Application of piezodetectors for diagnostics of pulsed and quasi-steady-state plasma streams

International Nuclear Information System (INIS)

Bandura, A.N.; Chebotarev, V.V.; Garkusha, I.E.; Tereshin, V.I.; Ladygina, M.S.

2006-01-01

The paper reports on studies of the plasma streams generated by two experimental devices: the quasi-steady-state plasma accelerator (QSPA) Kh-50 and the pulsed plasma gun PROSVET. The radial distributions of the plasma pressure for different times and varied distances from the accelerator output have been used for investigation of the plasma stream dynamics and study the plasma compression in the focus region for different operational regimes of plasma accelerators. In experiments for the application of pulsed plasma streams for surface modification of different industrial steels, optimal regimes of surface processing have been chosen on the basis of the plasma pressure measurements. Examples of application of the piezodetectors in simulation experiments on plasma surface interaction under high heat loads are presented

Design of an audio advertisement dataset

Science.gov (United States)

Fu, Yutao; Liu, Jihong; Zhang, Qi; Geng, Yuting

2015-12-01

Since more and more advertisements swarm into radios, it is necessary to establish an audio advertising dataset which could be used to analyze and classify the advertisement. A method of how to establish a complete audio advertising dataset is presented in this paper. The dataset is divided into four different kinds of advertisements. Each advertisement's sample is given in *.wav file format, and annotated with a txt file which contains its file name, sampling frequency, channel number, broadcasting time and its class. The classifying rationality of the advertisements in this dataset is proved by clustering the different advertisements based on Principal Component Analysis (PCA). The experimental results show that this audio advertisement dataset offers a reliable set of samples for correlative audio advertisement experimental studies.

Making the Switch to Digital Audio

Directory of Open Access Journals (Sweden)

Shannon Gwin Mitchell

2004-12-01

Full Text Available In this article, the authors describe the process of converting from analog to digital audio data. They address the step-by-step decisions that they made in selecting hardware and software for recording and converting digital audio, issues of system integration, and cost considerations. The authors present a brief description of how digital audio is being used in their current research project and how it has enhanced the “quality” of their qualitative research.

Efficient Audio Power Amplification - Challenges

DEFF Research Database (Denmark)

Andersen, Michael Andreas E.

2005-01-01

For more than a decade efficient audio power amplification has evolved and today switch-mode audio power amplification in various forms are the state-of-the-art. The technical steps that lead to this evolution are described and in addition many of the challenges still to be faced and where...

Layout Study and Application of Mobile App Recommendation Approach Based On Spark Streaming Framework

Science.gov (United States)

Wang, H. T.; Chen, T. T.; Yan, C.; Pan, H.

2018-05-01

For App recommended areas of mobile phone software, made while using conduct App application recommended combined weighted Slope One algorithm collaborative filtering algorithm items based on further improvement of the traditional collaborative filtering algorithm in cold start, data matrix sparseness and other issues, will recommend Spark stasis parallel algorithm platform, the introduction of real-time streaming streaming real-time computing framework to improve real-time software applications recommended.

A High-Voltage Class D Audio Amplifier for Dielectric Elastomer Transducers

DEFF Research Database (Denmark)

Nielsen, Dennis; Knott, Arnold; Andersen, Michael A. E.

2014-01-01

Dielectric Elastomer (DE) transducers have emerged as a very interesting alternative to the traditional electrodynamic transducer. Lightweight, small size and high maneuverability are some of the key features of the DE transducer. An amplifier for the DE transducer suitable for audio applications...... is proposed and analyzed. The amplifier addresses the issue of a high impedance load, ensuring a linear response over the midrange region of the audio bandwidth (100 Hz – 3.5 kHz). THD+N below 0.1% are reported for the ± 300 V prototype amplifier producing a maximum of 125 Var at a peak efficiency of 95 %....

Consequence of audio visual collection in school libraries

OpenAIRE

Kuri, Ramesh

2016-01-01

The collection of Audio-Visual in library plays important role in teaching and learning. The importance of audio visual (AV) technology in education should not be underestimated. If audio-visual collection in library is carefully planned and designed, it can provide a rich learning environment. In this article, an author discussed the consequences of Audio-Visual collection in libraries especially for students of school library

Audio feature extraction using probability distribution function

Science.gov (United States)

Suhaib, A.; Wan, Khairunizam; Aziz, Azri A.; Hazry, D.; Razlan, Zuradzman M.; Shahriman A., B.

2015-05-01

Voice recognition has been one of the popular applications in robotic field. It is also known to be recently used for biometric and multimedia information retrieval system. This technology is attained from successive research on audio feature extraction analysis. Probability Distribution Function (PDF) is a statistical method which is usually used as one of the processes in complex feature extraction methods such as GMM and PCA. In this paper, a new method for audio feature extraction is proposed which is by using only PDF as a feature extraction method itself for speech analysis purpose. Certain pre-processing techniques are performed in prior to the proposed feature extraction method. Subsequently, the PDF result values for each frame of sampled voice signals obtained from certain numbers of individuals are plotted. From the experimental results obtained, it can be seen visually from the plotted data that each individuals' voice has comparable PDF values and shapes.

About subjective evaluation of adaptive video streaming

Science.gov (United States)

Tavakoli, Samira; Brunnström, Kjell; Garcia, Narciso

2015-03-01

The usage of HTTP Adaptive Streaming (HAS) technology by content providers is increasing rapidly. Having available the video content in multiple qualities, using HAS allows to adapt the quality of downloaded video to the current network conditions providing smooth video-playback. However, the time-varying video quality by itself introduces a new type of impairment. The quality adaptation can be done in different ways. In order to find the best adaptation strategy maximizing users perceptual quality it is necessary to investigate about the subjective perception of adaptation-related impairments. However, the novelties of these impairments and their comparably long time duration make most of the standardized assessment methodologies fall less suited for studying HAS degradation. Furthermore, in traditional testing methodologies, the quality of the video in audiovisual services is often evaluated separated and not in the presence of audio. Nevertheless, the requirement of jointly evaluating the audio and the video within a subjective test is a relatively under-explored research field. In this work, we address the research question of determining the appropriate assessment methodology to evaluate the sequences with time-varying quality due to the adaptation. This was done by studying the influence of different adaptation related parameters through two different subjective experiments using a methodology developed to evaluate long test sequences. In order to study the impact of audio presence on quality assessment by the test subjects, one of the experiments was done in the presence of audio stimuli. The experimental results were subsequently compared with another experiment using the standardized single stimulus Absolute Category Rating (ACR) methodology.

A Novel Robust Audio Watermarking Algorithm by Modifying the Average Amplitude in Transform Domain

Directory of Open Access Journals (Sweden)

Qiuling Wu

2018-05-01

Full Text Available In order to improve the robustness and imperceptibility in practical application, a novel audio watermarking algorithm with strong robustness is proposed by exploring the multi-resolution characteristic of discrete wavelet transform (DWT and the energy compaction capability of discrete cosine transform (DCT. The human auditory system is insensitive to the minor changes in the frequency components of the audio signal, so the watermarks can be embedded by slightly modifying the frequency components of the audio signal. The audio fragments segmented from the cover audio signal are decomposed by DWT to obtain several groups of wavelet coefficients with different frequency bands, and then the fourth level detail coefficient is selected to be divided into the former packet and the latter packet, which are executed for DCT to get two sets of transform domain coefficients (TDC respectively. Finally, the average amplitudes of the two sets of TDC are modified to embed the binary image watermark according to the special embedding rule. The watermark extraction is blind without the carrier audio signal. Experimental results confirm that the proposed algorithm has good imperceptibility, large payload capacity and strong robustness when resisting against various attacks such as MP3 compression, low-pass filtering, re-sampling, re-quantization, amplitude scaling, echo addition and noise corruption.

Green computing: efficient energy management of multiprocessor streaming applications via model checking

NARCIS (Netherlands)

Ahmad, W.

2017-01-01

Streaming applications such as virtual reality, video conferencing, and face detection, impose high demands on a system’s performance and battery life. With the advancement in mobile computing, these applications are increasingly implemented on battery-constrained platforms, such as gaming consoles,

Efficient audio power amplification - challenges

Energy Technology Data Exchange (ETDEWEB)

Andersen, Michael A.E.

2005-07-01

For more than a decade efficient audio power amplification has evolved and today switch-mode audio power amplification in various forms are the state-of-the-art. The technical steps that lead to this evolution are described and in addition many of the challenges still to be faced and where extensive research and development are needed is covered. (au)

Detection Of Alterations In Audio Files Using Spectrograph Analysis

Directory of Open Access Journals (Sweden)

Anandha Krishnan G

2015-08-01

Full Text Available The corresponding study was carried out to detect changes in audio file using spectrograph. An audio file format is a file format for storing digital audio data on a computer system. A sound spectrograph is a laboratory instrument that displays a graphical representation of the strengths of the various component frequencies of a sound as time passes. The objectives of the study were to find the changes in spectrograph of audio after altering them to compare altering changes with spectrograph of original files and to check for similarity and difference in mp3 and wav. Five different alterations were carried out on each audio file to analyze the differences between the original and the altered file. For altering the audio file MP3 or WAV by cutcopy the file was opened in Audacity. A different audio was then pasted to the audio file. This new file was analyzed to view the differences. By adjusting the necessary parameters the noise was reduced. The differences between the new file and the original file were analyzed. By adjusting the parameters from the dialog box the necessary changes were made. The edited audio file was opened in the software named spek where after analyzing a graph is obtained of that particular file which is saved for further analysis. The original audio graph received was combined with the edited audio file graph to see the alterations.

Towards a universal representation for audio information retrieval and analysis

DEFF Research Database (Denmark)

Jensen, Bjørn Sand; Troelsgaard, Rasmus; Larsen, Jan

2013-01-01

A fundamental and general representation of audio and music which integrates multi-modal data sources is important for both application and basic research purposes. In this paper we address this challenge by proposing a multi-modal version of the Latent Dirichlet Allocation model which provides a...

Application of two-stream model to solar radiation of rice canopy

International Nuclear Information System (INIS)

Kawakata, T.

2005-01-01

The amount of solar radiation absorbed by a crop canopy is correlated with crop production, and thus it is necessary to estimate both transmission and reflection around the canopy for crop growth models. The 'forward and backward streams' representation of radiation has been refined to account for both transmission and reflection in the crop canopy. However, this model has not been applied to a rice canopy through the growing period. The purpose of this study is to examine whether the two-stream model is applicable to the rice canopy, and to investigate the parameters of the model. The values for both transmittance below the rice canopy and reflectance above it that were derived from the two-stream model represent the observed values throughout the growing period. The inclination factor of leaves (F), which is used in the two-stream model, was almost equivalent to the extinction coefficient of transmittance in the case of the rice canopy

Data streams: algorithms and applications

National Research Council Canada - National Science Library

Muthukrishnan, S

2005-01-01

... massive data sets in general. Researchers in Theoretical Computer Science, Databases, IP Networking and Computer Systems are working on the data stream challenges. This article is an overview and survey of data stream algorithmics and is an updated version of [175]. S. Muthukrishnan Rutgers University, New Brunswick, NJ, USA, muthu@cs...

Measurement of Streaming Potential in Downhole Application: An Insight for Enhanced Oil Recovery Monitoring

Directory of Open Access Journals (Sweden)

Tengku Mohd Tengku Amran

2017-01-01

Full Text Available Downhole monitoring using streaming potential measurement has been developing in order to respond to actual reservoir condition. Most studies have emphasized on monitoring water flooding at various reservoir condition and improving the approaches of measurement. Enhanced Oil Recovery (EOR could significantly improve oil recovery and the efficiency of the process should be well-monitored. Alkaline-surfactant-polymer (ASP flooding is the most promising chemical EOR method due to its synergy of alkaline, surfactant and polymer, which could enhance the extraction of residual oil. However, limited studies have been focused on the application of streaming potential in EOR processes, particularly ASP. Thus, this paper aims to review the streaming potential measurement in downhole monitoring with an insight for EOR application and propose the potential measurement in monitoring ASP flooding. It is important for a preliminary study to investigate the synergy in ASP and the effects on oil recovery. The behaviour of streaming potential should be investigated when the environment of porous media changes with respect to ASP flooding. Numerical model can be generated from the experimental data to forecast the measured streaming potential signal during production associated with ASP flooding. Based on the streaming potential behaviour on foam assisted water alternate gas (FAWAG and water alternate gas (WAG processes, it is expected that the streaming potential could change significantly when ASP flooding alters the environment and surface properties of porous media. The findings could provide new prospect and knowledge in the relationship between streaming potential and ASP mechanisms, which could be a potential approach in monitoring the efficiency of the process.

Predicting the Overall Spatial Quality of Automotive Audio Systems

Science.gov (United States)

Koya, Daisuke

The spatial quality of automotive audio systems is often compromised due to their unideal listening environments. Automotive audio systems need to be developed quickly due to industry demands. A suitable perceptual model could evaluate the spatial quality of automotive audio systems with similar reliability to formal listening tests but take less time. Such a model is developed in this research project by adapting an existing model of spatial quality for automotive audio use. The requirements for the adaptation were investigated in a literature review. A perceptual model called QESTRAL was reviewed, which predicts the overall spatial quality of domestic multichannel audio systems. It was determined that automotive audio systems are likely to be impaired in terms of the spatial attributes that were not considered in developing the QESTRAL model, but metrics are available that might predict these attributes. To establish whether the QESTRAL model in its current form can accurately predict the overall spatial quality of automotive audio systems, MUSHRA listening tests using headphone auralisation with head tracking were conducted to collect results to be compared against predictions by the model. Based on guideline criteria, the model in its current form could not accurately predict the overall spatial quality of automotive audio systems. To improve prediction performance, the QESTRAL model was recalibrated and modified using existing metrics of the model, those that were proposed from the literature review, and newly developed metrics. The most important metrics for predicting the overall spatial quality of automotive audio systems included those that were interaural cross-correlation (IACC) based, relate to localisation of the frontal audio scene, and account for the perceived scene width in front of the listener. Modifying the model for automotive audio systems did not invalidate its use for domestic audio systems. The resulting model predicts the overall spatial

Akamai Streaming

OpenAIRE

ECT Team, Purdue

2007-01-01

Akamai offers world-class streaming media services that enable Internet content providers and enterprises to succeed in today's Web-centric marketplace. They deliver live event Webcasts (complete with video production, encoding, and signal acquisition services), streaming media on demand, 24/7 Webcasts and a variety of streaming application services based upon their EdgeAdvantage.

Weaving together peer assessment, audios and medical vignettes in teaching medical terms

Science.gov (United States)

Khan, Lateef M.

2015-01-01

Objectives The current study aims at exploring the possibility of aligning peer assessment, audiovisuals, and medical case-report extracts (vignettes) in medical terminology teaching. In addition, the study wishes to highlight the effectiveness of audio materials and medical history vignettes in preventing medical students' comprehension, listening, writing, and pronunciation errors. The study also aims at reflecting the medical students' attitudes towards the teaching and learning process. Methods The study involved 161 medical students who received an intensive medical terminology course through audio and medical history extracts. Peer assessment and formative assessment platforms were applied through fake quizzes in a pre- and post-test manner. An 18-item survey was distributed amongst students to investigate their attitudes and feedback towards the teaching and learning process. Quantitative and qualitative data were analysed using the SPSS software. Results The students did better in the posttests than on the pretests for both the quizzes of audios and medical vignettes showing a t-test of -12.09 and -13.60 respectively. Moreover, out of the 133 students, 120 students (90.22%) responded to the survey questions. The students gave positive attitudes towards the application of audios and vignettes in the teaching and learning of medical terminology and towards the learning process. Conclusions The current study revealed that the teaching and learning of medical terminology have more room for the application of advanced technologies, effective assessment platforms, and active learning strategies in higher education. It also highlights that students are capable of carrying more responsibilities of assessment, feedback, and e-learning. PMID:26637986

Incorporation of water-use summaries into the StreamStats web application for Maryland

Science.gov (United States)

Ries, Kernell G.; Horn, Marilee A.; Nardi, Mark R.; Tessler, Steven

2010-01-01

Approximately 25,000 new households and thousands of new jobs will be established in an area that extends from southwest to northeast of Baltimore, Maryland, as a result of the Federal Base Realignment and Closure (BRAC) process, with consequent new demands on the water resources of the area. The U.S. Geological Survey, in cooperation with the Maryland Department of the Environment, has extended the area of implementation and added functionality to an existing map-based Web application named StreamStats to provide an improved tool for planning and managing the water resources in the BRAC-affected areas. StreamStats previously was implemented for only a small area surrounding Baltimore, Maryland, and it was extended to cover all BRAC-affected areas. StreamStats could provide previously published streamflow statistics, such as the 1-percent probability flood and the 7-day, 10-year low flow, for U.S. Geological Survey data-collection stations and estimates of streamflow statistics for any user-selected point on a stream within the implemented area. The application was modified for this study to also provide summaries of water withdrawals and discharges upstream from any user-selected point on a stream. This new functionality was made possible by creating a Web service that accepts a drainage-basin delineation from StreamStats, overlays it on a spatial layer of water withdrawal and discharge points, extracts the water-use data for the identified points, and sends it back to StreamStats, where it is summarized for the user. The underlying water-use data were extracted from the U.S. Geological Survey's Site-Specific Water-Use Database System (SWUDS) and placed into a Microsoft Access database that was created for this study for easy linkage to the Web service and StreamStats. This linkage of StreamStats with water-use information from SWUDS should enable Maryland regulators and planners to make more informed decisions on the use of water resources in the BRAC area, and

Editing Audio with Audacity

Directory of Open Access Journals (Sweden)

Brandon Walsh

2016-08-01

Full Text Available For those interested in audio, basic sound editing skills go a long way. Being able to handle and manipulate the materials can help you take control of your object of study: you can zoom in and extract particular moments to analyze, process the audio, and upload the materials to a server to compliment a blog post on the topic. On a more practical level, these skills could also allow you to record and package recordings of yourself or others for distribution. That guest lecture taking place in your department? Record it and edit it yourself! Doing so is a lightweight way to distribute resources among various institutions, and it also helps make the materials more accessible for readers and listeners with a wide variety of learning needs. In this lesson you will learn how to use Audacity to load, record, edit, mix, and export audio files. Sound editing platforms are often expensive and offer extensive capabilities that can be overwhelming to the first-time user, but Audacity is a free and open source alternative that offers powerful capabilities for sound editing with a low barrier for entry. For this lesson we will work with two audio files: a recording of Bach’s Goldberg Variations available from MusOpen and another recording of your own voice that will be made in the course of the lesson. This tutorial uses Audacity 2.1.2, released January 2016.

Fusion for Audio-Visual Laughter Detection

NARCIS (Netherlands)

Reuderink, B.

2007-01-01

Laughter is a highly variable signal, and can express a spectrum of emotions. This makes the automatic detection of laughter a challenging but interesting task. We perform automatic laughter detection using audio-visual data from the AMI Meeting Corpus. Audio-visual laughter detection is performed

AudioPairBank: Towards A Large-Scale Tag-Pair-Based Audio Content Analysis

OpenAIRE

Sager, Sebastian; Elizalde, Benjamin; Borth, Damian; Schulze, Christian; Raj, Bhiksha; Lane, Ian

2016-01-01

Recently, sound recognition has been used to identify sounds, such as car and river. However, sounds have nuances that may be better described by adjective-noun pairs such as slow car, and verb-noun pairs such as flying insects, which are under explored. Therefore, in this work we investigate the relation between audio content and both adjective-noun pairs and verb-noun pairs. Due to the lack of datasets with these kinds of annotations, we collected and processed the AudioPairBank corpus cons...

Tourism research and audio methods

DEFF Research Database (Denmark)

Jensen, Martin Trandberg

2016-01-01

• Audio methods enriches sensuous tourism ethnographies. • The note suggests five research avenues for future auditory scholarship. • Sensuous tourism research has neglected the role of sounds in embodied tourism experiences.......• Audio methods enriches sensuous tourism ethnographies. • The note suggests five research avenues for future auditory scholarship. • Sensuous tourism research has neglected the role of sounds in embodied tourism experiences....

Newnes audio and Hi-Fi engineer's pocket book

CERN Document Server

Capel, Vivian

2013-01-01

Newnes Audio and Hi-Fi Engineer's Pocket Book, Second Edition provides concise discussion of several audio topics. The book is comprised of 10 chapters that cover different audio equipment. The coverage of the text includes microphones, gramophones, compact discs, and tape recorders. The book also covers high-quality radio, amplifiers, and loudspeakers. The book then reviews the concepts of sound and acoustics, and presents some facts and formulas relevant to audio. The text will be useful to sound engineers and other professionals whose work involves sound systems.

A Perceptually Reweighted Mixed-Norm Method for Sparse Approximation of Audio Signals

DEFF Research Database (Denmark)

Christensen, Mads Græsbøll; Sturm, Bob L.

2011-01-01

using standard software. A prominent feature of the new method is that it solves a problem that is closely related to the objective of coding, namely rate-distortion optimization. In computer simulations, we demonstrate the properties of the algorithm and its application to real audio signals.......In this paper, we consider the problem of finding sparse representations of audio signals for coding purposes. In doing so, it is of utmost importance that when only a subset of the present components of an audio signal are extracted, it is the perceptually most important ones. To this end, we...... propose a new iterative algorithm based on two principles: 1) a reweighted l1-norm based measure of sparsity; and 2) a reweighted l2-norm based measure of perceptual distortion. Using these measures, the considered problem is posed as a constrained convex optimization problem that can be solved optimally...

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

Science.gov (United States)

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

2010-01-01

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.

47 CFR 10.520 - Common audio attention signal.

Science.gov (United States)

2010-10-01

... 47 Telecommunication 1 2010-10-01 2010-10-01 false Common audio attention signal. 10.520 Section... Equipment Requirements § 10.520 Common audio attention signal. A Participating CMS Provider and equipment manufacturers may only market devices for public use under part 10 that include an audio attention signal that...

Audio Journal in an ELT Context

Directory of Open Access Journals (Sweden)

Neşe Aysin Siyli

2012-09-01

Full Text Available It is widely acknowledged that one of the most serious problems students of English as a foreign language face is their deprivation of practicing the language outside the classroom. Generally, the classroom is the sole environment where they can practice English, which by its nature does not provide rich setting to help students develop their competence by putting the language into practice. Motivated by this need, this descriptive study investigated the impact of audio dialog journals on students’ speaking skills. It also aimed to gain insights into students’ and teacher’s opinions on keeping audio dialog journals outside the class. The data of the study developed from student and teacher audio dialog journals, student written feedbacks, interviews held with the students, and teacher observations. The descriptive analysis of the data revealed that audio dialog journals served a number of functions ranging from cognitive to linguistic, from pedagogical to psychological, and social. The findings and pedagogical implications of the study are discussed in detail.

Music Genre Classification Using MIDI and Audio Features

Science.gov (United States)

Cataltepe, Zehra; Yaslan, Yusuf; Sonmez, Abdullah

2007-12-01

We report our findings on using MIDI files and audio features from MIDI, separately and combined together, for MIDI music genre classification. We use McKay and Fujinaga's 3-root and 9-leaf genre data set. In order to compute distances between MIDI pieces, we use normalized compression distance (NCD). NCD uses the compressed length of a string as an approximation to its Kolmogorov complexity and has previously been used for music genre and composer clustering. We convert the MIDI pieces to audio and then use the audio features to train different classifiers. MIDI and audio from MIDI classifiers alone achieve much smaller accuracies than those reported by McKay and Fujinaga who used not NCD but a number of domain-based MIDI features for their classification. Combining MIDI and audio from MIDI classifiers improves accuracy and gets closer to, but still worse, accuracies than McKay and Fujinaga's. The best root genre accuracies achieved using MIDI, audio, and combination of them are 0.75, 0.86, and 0.93, respectively, compared to 0.98 of McKay and Fujinaga. Successful classifier combination requires diversity of the base classifiers. We achieve diversity through using certain number of seconds of the MIDI file, different sample rates and sizes for the audio file, and different classification algorithms.

Realtime Audio with Garbage Collection

OpenAIRE

Matheussen, Kjetil Svalastog

2010-01-01

Two non-moving concurrent garbage collectors tailored for realtime audio processing are described. Both collectors work on copies of the heap to avoid cache misses and audio-disruptive synchronizations. Both collectors are targeted at multiprocessor personal computers. The first garbage collector works in uncooperative environments, and can replace Hans Boehm's conservative garbage collector for C and C++. The collector does not access the virtual memory system. Neither doe...

Audio-Visual Classification of Sports Types

DEFF Research Database (Denmark)

Gade, Rikke; Abou-Zleikha, Mohamed; Christensen, Mads Græsbøll

2015-01-01

In this work we propose a method for classification of sports types from combined audio and visual features ex- tracted from thermal video. From audio Mel Frequency Cepstral Coefficients (MFCC) are extracted, and PCA are applied to reduce the feature space to 10 dimensions. From the visual modali...

Improved Techniques for Automatic Chord Recognition from Music Audio Signals

Science.gov (United States)

Cho, Taemin

2014-01-01

This thesis is concerned with the development of techniques that facilitate the effective implementation of capable automatic chord transcription from music audio signals. Since chord transcriptions can capture many important aspects of music, they are useful for a wide variety of music applications and also useful for people who learn and perform…

Tools for signal compression applications to speech and audio coding

CERN Document Server

Moreau, Nicolas

2013-01-01

This book presents tools and algorithms required to compress/uncompress signals such as speech and music. These algorithms are largely used in mobile phones, DVD players, HDTV sets, etc. In a first rather theoretical part, this book presents the standard tools used in compression systems: scalar and vector quantization, predictive quantization, transform quantization, entropy coding. In particular we show the consistency between these different tools. The second part explains how these tools are used in the latest speech and audio coders. The third part gives Matlab programs simulating t

Temporal analysis of static priority preemptive scheduled cyclic streaming applications using CSDF models

NARCIS (Netherlands)

Kurtin, Philip Sebastian; Bekooij, Marco Jan Gerrit

2016-01-01

Real-time streaming applications with cyclic data dependencies that are executed on multiprocessor systems with processor sharing usually require a temporal analysis to give guarantees on their temporal behavior at design time. Current accurate analysis techniques for cyclic applications that are

CERN automatic audio-conference service

International Nuclear Information System (INIS)

Sierra Moral, Rodrigo

2010-01-01

Scientists from all over the world need to collaborate with CERN on a daily basis. They must be able to communicate effectively on their joint projects at any time; as a result telephone conferences have become indispensable and widely used. Managed by 6 operators, CERN already has more than 20000 hours and 5700 audio-conferences per year. However, the traditional telephone based audio-conference system needed to be modernized in three ways. Firstly, to provide the participants with more autonomy in the organization of their conferences; secondly, to eliminate the constraints of manual intervention by operators; and thirdly, to integrate the audio-conferences into a collaborative working framework. The large number, and hence cost, of the conferences prohibited externalization and so the CERN telecommunications team drew up a specification to implement a new system. It was decided to use a new commercial collaborative audio-conference solution based on the SIP protocol. The system was tested as the first European pilot and several improvements (such as billing, security, redundancy...) were implemented based on CERN's recommendations. The new automatic conference system has been operational since the second half of 2006. It is very popular for the users and has doubled the number of conferences in the past two years.

CERN automatic audio-conference service

Energy Technology Data Exchange (ETDEWEB)

Sierra Moral, Rodrigo, E-mail: Rodrigo.Sierra@cern.c [CERN, IT Department 1211 Geneva-23 (Switzerland)

2010-04-01

Scientists from all over the world need to collaborate with CERN on a daily basis. They must be able to communicate effectively on their joint projects at any time; as a result telephone conferences have become indispensable and widely used. Managed by 6 operators, CERN already has more than 20000 hours and 5700 audio-conferences per year. However, the traditional telephone based audio-conference system needed to be modernized in three ways. Firstly, to provide the participants with more autonomy in the organization of their conferences; secondly, to eliminate the constraints of manual intervention by operators; and thirdly, to integrate the audio-conferences into a collaborative working framework. The large number, and hence cost, of the conferences prohibited externalization and so the CERN telecommunications team drew up a specification to implement a new system. It was decided to use a new commercial collaborative audio-conference solution based on the SIP protocol. The system was tested as the first European pilot and several improvements (such as billing, security, redundancy...) were implemented based on CERN's recommendations. The new automatic conference system has been operational since the second half of 2006. It is very popular for the users and has doubled the number of conferences in the past two years.

CERN automatic audio-conference service

Science.gov (United States)

Sierra Moral, Rodrigo

2010-04-01

Scientists from all over the world need to collaborate with CERN on a daily basis. They must be able to communicate effectively on their joint projects at any time; as a result telephone conferences have become indispensable and widely used. Managed by 6 operators, CERN already has more than 20000 hours and 5700 audio-conferences per year. However, the traditional telephone based audio-conference system needed to be modernized in three ways. Firstly, to provide the participants with more autonomy in the organization of their conferences; secondly, to eliminate the constraints of manual intervention by operators; and thirdly, to integrate the audio-conferences into a collaborative working framework. The large number, and hence cost, of the conferences prohibited externalization and so the CERN telecommunications team drew up a specification to implement a new system. It was decided to use a new commercial collaborative audio-conference solution based on the SIP protocol. The system was tested as the first European pilot and several improvements (such as billing, security, redundancy...) were implemented based on CERN's recommendations. The new automatic conference system has been operational since the second half of 2006. It is very popular for the users and has doubled the number of conferences in the past two years.

Musical Audio Synthesis Using Autoencoding Neural Nets

OpenAIRE

Sarroff, Andy; Casey, Michael A.

2014-01-01

With an optimal network topology and tuning of hyperpa-\\ud rameters, artificial neural networks (ANNs) may be trained\\ud to learn a mapping from low level audio features to one\\ud or more higher-level representations. Such artificial neu-\\ud ral networks are commonly used in classification and re-\\ud gression settings to perform arbitrary tasks. In this work\\ud we suggest repurposing autoencoding neural networks as\\ud musical audio synthesizers. We offer an interactive musi-\\ud cal audio synt...

Geospatial Image Stream Processing: Models, techniques, and applications in remote sensing change detection

Science.gov (United States)

Rueda-Velasquez, Carlos Alberto

Detection of changes in environmental phenomena using remotely sensed data is a major requirement in the Earth sciences, especially in natural disaster related scenarios where real-time detection plays a crucial role in the saving of human lives and the preservation of natural resources. Although various approaches formulated to model multidimensional data can in principle be applied to the inherent complexity of remotely sensed geospatial data, there are still challenging peculiarities that demand a precise characterization in the context of change detection, particularly in scenarios of fast changes. In the same vein, geospatial image streams do not fit appropriately in the standard Data Stream Management System (DSMS) approach because these systems mainly deal with tuple-based streams. Recognizing the necessity for a systematic effort to address the above issues, the work presented in this thesis is a concrete step toward the foundation and construction of an integrated Geospatial Image Stream Processing framework, GISP. First, we present a data and metadata model for remotely sensed image streams. We introduce a precise characterization of images and image streams in the context of remotely sensed geospatial data. On this foundation, we define spatially-aware temporal operators with a consistent semantics for change analysis tasks. We address the change detection problem in settings where multiple image stream sources are available, and thus we introduce an architectural design for the processing of geospatial image streams from multiple sources. With the aim of targeting collaborative scientific environments, we construct a realization of our architecture based on Kepler, a robust and widely used scientific workflow management system, as the underlying computational support; and open data and Web interface standards, as a means to facilitate the interoperability of GISP instances with other processing infrastructures and client applications. We demonstrate our

Robust and Reversible Audio Watermarking by Modifying Statistical Features in Time Domain

Directory of Open Access Journals (Sweden)

Shijun Xiang

2017-01-01

Full Text Available Robust and reversible watermarking is a potential technique in many sensitive applications, such as lossless audio or medical image systems. This paper presents a novel robust reversible audio watermarking method by modifying the statistic features in time domain in the way that the histogram of these statistical values is shifted for data hiding. Firstly, the original audio is divided into nonoverlapped equal-sized frames. In each frame, the use of three samples as a group generates a prediction error and a statistical feature value is calculated as the sum of all the prediction errors in the frame. The watermark bits are embedded into the frames by shifting the histogram of the statistical features. The watermark is reversible and robust to common signal processing operations. Experimental results have shown that the proposed method not only is reversible but also achieves satisfactory robustness to MP3 compression of 64 kbps and additive Gaussian noise of 35 dB.

Music Genre Classification Using MIDI and Audio Features

Directory of Open Access Journals (Sweden)

Abdullah Sonmez

2007-01-01

Full Text Available We report our findings on using MIDI files and audio features from MIDI, separately and combined together, for MIDI music genre classification. We use McKay and Fujinaga's 3-root and 9-leaf genre data set. In order to compute distances between MIDI pieces, we use normalized compression distance (NCD. NCD uses the compressed length of a string as an approximation to its Kolmogorov complexity and has previously been used for music genre and composer clustering. We convert the MIDI pieces to audio and then use the audio features to train different classifiers. MIDI and audio from MIDI classifiers alone achieve much smaller accuracies than those reported by McKay and Fujinaga who used not NCD but a number of domain-based MIDI features for their classification. Combining MIDI and audio from MIDI classifiers improves accuracy and gets closer to, but still worse, accuracies than McKay and Fujinaga's. The best root genre accuracies achieved using MIDI, audio, and combination of them are 0.75, 0.86, and 0.93, respectively, compared to 0.98 of McKay and Fujinaga. Successful classifier combination requires diversity of the base classifiers. We achieve diversity through using certain number of seconds of the MIDI file, different sample rates and sizes for the audio file, and different classification algorithms.

Current-Driven Switch-Mode Audio Power Amplifiers

DEFF Research Database (Denmark)

Knott, Arnold; Buhl, Niels Christian; Andersen, Michael A. E.

2012-01-01

The conversion of electrical energy into sound waves by electromechanical transducers is proportional to the current through the coil of the transducer. However virtually all audio power amplifiers provide a controlled voltage through the interface to the transducer. This paper is presenting...... a switch-mode audio power amplifier not only providing controlled current but also being supplied by current. This results in an output filter size reduction by a factor of 6. The implemented prototype shows decent audio performance with THD + N below 0.1 %....

High-Order Sparse Linear Predictors for Audio Processing

DEFF Research Database (Denmark)

Giacobello, Daniele; van Waterschoot, Toon; Christensen, Mads Græsbøll

2010-01-01

Linear prediction has generally failed to make a breakthrough in audio processing, as it has done in speech processing. This is mostly due to its poor modeling performance, since an audio signal is usually an ensemble of different sources. Nevertheless, linear prediction comes with a whole set...... of interesting features that make the idea of using it in audio processing not far fetched, e.g., the strong ability of modeling the spectral peaks that play a dominant role in perception. In this paper, we provide some preliminary conjectures and experiments on the use of high-order sparse linear predictors...... in audio processing. These predictors, successfully implemented in modeling the short-term and long-term redundancies present in speech signals, will be used to model tonal audio signals, both monophonic and polyphonic. We will show how the sparse predictors are able to model efficiently the different...

Audio Mining with emphasis on Music Genre Classification

DEFF Research Database (Denmark)

Meng, Anders

2004-01-01

Audio is an important part of our daily life, basically it increases our impression of the world around us whether this is communication, music, danger detection etc. Currently the field of Audio Mining, which here includes areas of music genre, music recognition / retrieval, playlist generation...... the world the problem of detecting environments from the input audio is researched as to increase the life quality of hearing-impaired. Basically there is a lot of work within the field of audio mining. The presentation will mainly focus on music genre classification where we have a fixed amount of genres...... to choose from. Basically every audio mining system is more or less consisting of the same stages as for the music genre setting. My research so far has mainly focussed on finding relevant features for music genre classification living at different timescales using early and late information fusion. It has...

Pipeline template for streaming applications on heterogeneous chips

OpenAIRE

Rodríguez, Andrés; Navarro, Ángeles; Asenjo-Plaza, Rafael; Corbera, Francisco; Vilches, Antonio; Garzarán, María

2015-01-01

We address the problem of providing support for executing single streaming applications implemented as a pipeline of stages that run on heterogeneous chips comprised of several cores and one on-chip GPU. In this paper, we mainly focus on the API that allows the user to specify the type of parallelism exploited by each pipeline stage running on the multicore CPU, the mapping of the pipeline stages to the devices (GPU or CPU), and the number of active threads. We use a rea...

Augmenting Environmental Interaction in Audio Feedback Systems

Directory of Open Access Journals (Sweden)

Seunghun Kim

2016-04-01

Full Text Available Audio feedback is defined as a positive feedback of acoustic signals where an audio input and output form a loop, and may be utilized artistically. This article presents new context-based controls over audio feedback, leading to the generation of desired sonic behaviors by enriching the influence of existing acoustic information such as room response and ambient noise. This ecological approach to audio feedback emphasizes mutual sonic interaction between signal processing and the acoustic environment. Mappings from analyses of the received signal to signal-processing parameters are designed to emphasize this specificity as an aesthetic goal. Our feedback system presents four types of mappings: approximate analyses of room reverberation to tempo-scale characteristics, ambient noise to amplitude and two different approximations of resonances to timbre. These mappings are validated computationally and evaluated experimentally in different acoustic conditions.

Investigating the impact of audio instruction and audio-visual biofeedback for lung cancer radiation therapy

Science.gov (United States)

George, Rohini

Lung cancer accounts for 13% of all cancers in the Unites States and is the leading cause of deaths among both men and women. The five-year survival for lung cancer patients is approximately 15%.(ACS facts & figures) Respiratory motion decreases accuracy of thoracic radiotherapy during imaging and delivery. To account for respiration, generally margins are added during radiation treatment planning, which may cause a substantial dose delivery to normal tissues and increase the normal tissue toxicity. To alleviate the above-mentioned effects of respiratory motion, several motion management techniques are available which can reduce the doses to normal tissues, thereby reducing treatment toxicity and allowing dose escalation to the tumor. This may increase the survival probability of patients who have lung cancer and are receiving radiation therapy. However the accuracy of these motion management techniques are inhibited by respiration irregularity. The rationale of this thesis was to study the improvement in regularity of respiratory motion by breathing coaching for lung cancer patients using audio instructions and audio-visual biofeedback. A total of 331 patient respiratory motion traces, each four minutes in length, were collected from 24 lung cancer patients enrolled in an IRB-approved breathing-training protocol. It was determined that audio-visual biofeedback significantly improved the regularity of respiratory motion compared to free breathing and audio instruction, thus improving the accuracy of respiratory gated radiotherapy. It was also observed that duty cycles below 30% showed insignificant reduction in residual motion while above 50% there was a sharp increase in residual motion. The reproducibility of exhale based gating was higher than that of inhale base gating. Modeling the respiratory cycles it was found that cosine and cosine 4 models had the best correlation with individual respiratory cycles. The overall respiratory motion probability distribution

Fusion of audio and visual cues for laughter detection

NARCIS (Netherlands)

Petridis, Stavros; Pantic, Maja

Past research on automatic laughter detection has focused mainly on audio-based detection. Here we present an audio- visual approach to distinguishing laughter from speech and we show that integrating the information from audio and video channels leads to improved performance over single-modal

BDVC (Bimodal Database of Violent Content): A database of violent audio and video

Science.gov (United States)

Rivera Martínez, Jose Luis; Mijes Cruz, Mario Humberto; Rodríguez Vázqu, Manuel Antonio; Rodríguez Espejo, Luis; Montoya Obeso, Abraham; García Vázquez, Mireya Saraí; Ramírez Acosta, Alejandro Álvaro

2017-09-01

Nowadays there is a trend towards the use of unimodal databases for multimedia content description, organization and retrieval applications of a single type of content like text, voice and images, instead bimodal databases allow to associate semantically two different types of content like audio-video, image-text, among others. The generation of a bimodal database of audio-video implies the creation of a connection between the multimedia content through the semantic relation that associates the actions of both types of information. This paper describes in detail the used characteristics and methodology for the creation of the bimodal database of violent content; the semantic relationship is stablished by the proposed concepts that describe the audiovisual information. The use of bimodal databases in applications related to the audiovisual content processing allows an increase in the semantic performance only and only if these applications process both type of content. This bimodal database counts with 580 audiovisual annotated segments, with a duration of 28 minutes, divided in 41 classes. Bimodal databases are a tool in the generation of applications for the semantic web.

Perceptual Audio Hashing Functions

Directory of Open Access Journals (Sweden)

Emin Anarım

2005-07-01

Full Text Available Perceptual hash functions provide a tool for fast and reliable identification of content. We present new audio hash functions based on summarization of the time-frequency spectral characteristics of an audio document. The proposed hash functions are based on the periodicity series of the fundamental frequency and on singular-value description of the cepstral frequencies. They are found, on one hand, to perform very satisfactorily in identification and verification tests, and on the other hand, to be very resilient to a large variety of attacks. Moreover, we address the issue of security of hashes and propose a keying technique, and thereby a key-dependent hash function.

Stream Classification Tool User Manual: For Use in Applications in Hydropower-Related Evironmental Mitigation

Energy Technology Data Exchange (ETDEWEB)

McManamay, Ryan A. [Oak Ridge National Lab. (ORNL), Oak Ridge, TN (United States); Troia, Matthew J. [Oak Ridge National Lab. (ORNL), Oak Ridge, TN (United States); DeRolph, Christopher R. [Oak Ridge National Lab. (ORNL), Oak Ridge, TN (United States); Samu, Nicole M. [Oak Ridge National Lab. (ORNL), Oak Ridge, TN (United States)

2016-01-01

Stream classifications are an inventory of different types of streams. Classifications help us explore similarities and differences among different types of streams, make inferences regarding stream ecosystem behavior, and communicate the complexities of ecosystems. We developed a nested, layered, and spatially contiguous stream classification to characterize the biophysical settings of stream reaches within the Eastern United States (~ 900,000 reaches). The classification is composed of five natural characteristics (hydrology, temperature, size, confinement, and substrate) along with several disturbance regime layers, and each was selected because of their relevance to hydropower mitigation. We developed the classification at the stream reach level using the National Hydrography Dataset Plus Version 1 (1:100k scale). The stream classification is useful to environmental mitigation for hydropower dams in multiple ways. First, it creates efficiency in the regulatory process by creating an objective and data-rich means to address meaningful mitigation actions. Secondly, the SCT addresses data gaps as it quickly provides an inventory of hydrology, temperature, morphology, and ecological communities for the immediate project area, but also surrounding streams. This includes identifying potential reference streams as those that are proximate to the hydropower facility and fall within the same class. These streams can potentially be used to identify ideal environmental conditions or identify desired ecological communities. In doing so, the stream provides some context for how streams may function, respond to dam regulation, and an overview of specific mitigation needs. Herein, we describe the methodology in developing each stream classification layer and provide a tutorial to guide applications of the classification (and associated data) in regulatory settings, such as hydropower (re)licensing.

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

Science.gov (United States)

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

2010-01-01

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…

Audio localization for mobile robots

OpenAIRE

de Guillebon, Thibaut; Grau Saldes, Antoni; Bolea Monte, Yolanda

2009-01-01

The department of the University for which I worked is developing a project based on the interaction with robots in the environment. My work was to define an audio system for the robot. This audio system that I have to realize consists on a mobile head which is able to follow the sound in its environment. This subject was treated as a research problem, with the liberty to find and develop different solutions and make them evolve in the chosen way.

Removable Watermarking Sebagai Pengendalian Terhadap Cyber Crime Pada Audio Digital

Directory of Open Access Journals (Sweden)

Reyhani Lian Putri

2017-08-01

Full Text Available Perkembangan teknologi informasi yang pesat menuntut penggunanya untuk lebih berhati-hati seiring semakin meningkatnya cyber crime.Banyak pihak telah mengembangkan berbagai teknik perlindungan data digital, salah satunya adalah watermarking. Teknologi watermarking berfungsi untuk memberikan identitas, melindungi, atau menandai data digital, baik audio, citra, ataupun video, yang mereka miliki. Akan tetapi, teknik tersebut masih dapat diretas oleh oknum-oknum yang tidak bertanggung jawab.Pada penelitian ini, proses watermarking diterapkan pada audio digital dengan menyisipkan watermark yang terdengar jelas oleh indera pendengaran manusia (perceptible pada audio host.Hal ini bertujuan agar data audio dapat terlindungi dan apabila ada pihak lain yang ingin mendapatkan data audio tersebut harus memiliki “kunci” untuk menghilangkan watermark. Proses removable watermarking ini dilakukan pada data watermark yang sudah diketahui metode penyisipannya, agar watermark dapat dihilangkan sehingga kualitas audio menjadi lebih baik. Dengan menggunakan metode ini diperoleh kinerja audio watermarking pada nilai distorsi tertinggi dengan rata-rata nilai SNR sebesar7,834 dB dan rata-rata nilai ODG sebesar -3,77.Kualitas audio meningkat setelah watermark dihilangkan, di mana rata-rata SNR menjadi sebesar 24,986 dB dan rata-rata ODG menjadi sebesar -1,064 serta nilai MOS sebesar 4,40.

A listening test system for automotive audio

DEFF Research Database (Denmark)

Christensen, Flemming; Geoff, Martin; Minnaar, Pauli

2005-01-01

This paper describes a system for simulating automotive audio through headphones for the purposes of conducting listening experiments in the laboratory. The system is based on binaural technology and consists of a component for reproducing the sound of the audio system itself and a component...

Voice activity detection using audio-visual information

DEFF Research Database (Denmark)

Petsatodis, Theodore; Pnevmatikakis, Aristodemos; Boukis, Christos

2009-01-01

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...

Audio power amplifier design handbook

CERN Document Server

Self, Douglas

2013-01-01

This book is essential for audio power amplifier designers and engineers for one simple reason...it enables you as a professional to develop reliable, high-performance circuits. The Author Douglas Self covers the major issues of distortion and linearity, power supplies, overload, DC-protection and reactive loading. He also tackles unusual forms of compensation and distortion produced by capacitors and fuses. This completely updated fifth edition includes four NEW chapters including one on The XD Principle, invented by the author, and used by Cambridge Audio. Cro

MP3 audio-editing software for the department of radiology

International Nuclear Information System (INIS)

Hong Qingfen; Sun Canhui; Li Ziping; Meng Quanfei; Jiang Li

2006-01-01

Objective: To evaluate the MP3 audio-editing software in the daily work in the department of radiology. Methods: The audio content of daily consultation seminar, held in the department of radiology every morning, was recorded and converted into MP3 audio format by a computer integrated recording device. The audio data were edited, archived, and eventually saved in the computer memory storage media, which was experimentally replayed and applied in the research or teaching. Results: MP3 audio-editing was a simple process and convenient for saving and searching the data. The record could be easily replayed. Conclusion: MP3 audio-editing perfectly records and saves the contents of consultation seminar, and has replaced the conventional hand writing notes. It is a valuable tool in both research and teaching in the department. (authors)

An Interactive Mobile Application for the Visually Impaired to Have Access to Listening Audio Books with Handy Books Portal

Directory of Open Access Journals (Sweden)

Avanthika Meenakshi

2015-01-01

Full Text Available Mobile phones are used in almost all aspects of life by people. But in the case of visually impaired, they are still a step behind in using smart phones for various purposes. Having interactive android OS, navigation and travel aiding apps using sensors and voice user interfaces (VUI or the voice response systems, we are still a step lagging in giving them an application for educational purposes. This paper proposes a complete new idea of having a portal where they can store audio books aided with interactive system so that they can use them whenever needed.

Use of Effective Audio in E-learning Courseware

OpenAIRE

Ray, Kisor

2015-01-01

E-Learning uses electronic media, information & communication technologies to provide education to the masses. E-learning deliver hypertext, text, audio, images, animation and videos using desktop standalone computer, local area network based intranet and internet based contents. While producing an e-learning content or course-ware, a major decision making factor is whether to use audio for the benefit of the end users. Generally, three types of audio can be used in e-learning: narration, mus...

A Perceptual Model for Sinusoidal Audio Coding Based on Spectral Integration

Directory of Open Access Journals (Sweden)

Jensen Søren Holdt

2005-01-01

Full Text Available Psychoacoustical models have been used extensively within audio coding applications over the past decades. Recently, parametric coding techniques have been applied to general audio and this has created the need for a psychoacoustical model that is specifically suited for sinusoidal modelling of audio signals. In this paper, we present a new perceptual model that predicts masked thresholds for sinusoidal distortions. The model relies on signal detection theory and incorporates more recent insights about spectral and temporal integration in auditory masking. As a consequence, the model is able to predict the distortion detectability. In fact, the distortion detectability defines a (perceptually relevant norm on the underlying signal space which is beneficial for optimisation algorithms such as rate-distortion optimisation or linear predictive coding. We evaluate the merits of the model by combining it with a sinusoidal extraction method and compare the results with those obtained with the ISO MPEG-1 Layer I-II recommended model. Listening tests show a clear preference for the new model. More specifically, the model presented here leads to a reduction of more than 20% in terms of number of sinusoids needed to represent signals at a given quality level.

TC9447F, single-chip DSP (digital signal processor) for audio; 1 chip audio yo DSP LSI TC9447F

Energy Technology Data Exchange (ETDEWEB)

NONE

1999-03-01

TC9447F is a single-chip DSP for audio which builds in 2-channel AD converter/4-channel DA converter. It can build various application programs such as the sound field control like hall simulation, digital filter like equalizer, and dynamic range control, in the program memory (ROM). Further, it builds in {+-}10dB trim use electronic volume for two channels. It also builds data delay use RAM (64K-bit) in, so no RAM to be separately attached is necessary. (translated by NEDO)

New Jersey StreamStats: A web application for streamflow statistics and basin characteristics

Science.gov (United States)

Watson, Kara M.; Janowicz, Jon A.

2017-08-02

StreamStats is an interactive, map-based web application from the U.S. Geological Survey (USGS) that allows users to easily obtain streamflow statistics and watershed characteristics for both gaged and ungaged sites on streams throughout New Jersey. Users can determine flood magnitude and frequency, monthly flow-duration, monthly low-flow frequency statistics, and watershed characteristics for ungaged sites by selecting a point along a stream, or they can obtain this information for streamgages by selecting a streamgage location on the map. StreamStats provides several additional tools useful for water-resources planning and management, as well as for engineering purposes. StreamStats is available for most states and some river basins through a single web portal.Streamflow statistics for water resources professionals include the 1-percent annual chance flood flow (100-year peak flow) used to define flood plain areas and the monthly 7-day, 10-year low flow (M7D10Y) used in water supply management and studies of recreation, wildlife conservation, and wastewater dilution. Additionally, watershed or basin characteristics, including drainage area, percent area forested, and average percent of impervious areas, are commonly used in land-use planning and environmental assessments. These characteristics are easily derived through StreamStats.

Behavioral System Level Power Consumption Modeling of Mobile Video Streaming applications

OpenAIRE

Benmoussa , Yahia; Boukhobza , Jalil; Hadjadj-Aoul , Yassine; Lagadec , Loïc; Benazzouz , Djamel

2012-01-01

National audience; Nowadays, the use of mobile applications and terminals faces fundamental challenges related to energy constraint. This is due to the limited battery lifetime as compared to the increasing hardware evolution. Video streaming is one of the most energy consuming applications in a mobile system because of its intensive use of bandwidth, memory and processing power. In this work, we aim to propose a methodology for building and validating a high level global power consumption mo...

Car audio using DSP for active sound control. DSP ni yoru active seigyo wo mochiita audio

Energy Technology Data Exchange (ETDEWEB)

Yamada, K.; Asano, S.; Furukawa, N. (Mitsubishi Motor Corp., Tokyo (Japan))

1993-06-01

In the automobile cabin, there are some unique problems which spoil the quality of sound reproduction from audio equipment, such as the narrow space and/or the background noise. The audio signal processing by using DSP (digital signal processor) makes enable a solution to these problems. A car audio with a high amenity has been successfully made by the active sound control using DSP. The DSP consists of an adder, coefficient multiplier, delay unit, and connections. For the actual processing by DSP, are used functions, such as sound field correction, response and processing of noises during driving, surround reproduction, graphic equalizer processing, etc. High effectiveness of the method was confirmed through the actual driving evaluation test. The present paper describes the actual method of sound control technology using DSP. Especially, the dynamic processing of the noise during driving is discussed in detail. 1 ref., 12 figs., 1 tab.

Hierarchical programming language for modal multi-rate real-time stream processing applications

NARCIS (Netherlands)

Geuns, S.J.; Hausmans, J.P.H.M.; Bekooij, Marco Jan Gerrit

2014-01-01

Modal multi-rate stream processing applications with real-time constraints which are executed on multi-core embedded systems often cannot be conveniently specified using current programming languages. An important issue is that sequential programming languages do not allow for convenient programming

Perceived Audio Quality Analysis in Digital Audio Broadcasting Plus System Based on PEAQ

Directory of Open Access Journals (Sweden)

K. Ulovec

2018-04-01

Full Text Available Broadcasters need to decide on bitrates of the services in the multiplex transmitted via Digital Audio Broadcasting Plus system. The bitrate should be set as low as possible for maximal number of services, but with high quality, not lower than in conventional analog systems. In this paper, the objective method Perceptual Evaluation of Audio Quality is used to analyze the perceived audio quality for appropriate codecs --- MP2 and AAC offering three profiles. The main aim is to determine dependencies on the type of signal --- music and speech, the number of channels --- stereo and mono, and the bitrate. Results indicate that only MP2 codec and AAC Low Complexity profile reach imperceptible quality loss. The MP2 codec needs higher bitrate than AAC Low Complexity profile for the same quality. For the both versions of AAC High-Efficiency profiles, the limit bitrates are determined above which less complex profiles outperform the more complex ones and higher bitrates above these limits are not worth using. It is shown that stereo music has worse quality than stereo speech generally, whereas for mono, the dependencies vary upon the codec/profile. Furthermore, numbers of services satisfying various quality criteria are presented.

Parameter and state estimation using audio and video signals

OpenAIRE

Evestedt, Magnus

2005-01-01

The complexity of industrial systems and the mathematical models to describe them increases. In many cases point sensors are no longer sufficient to provide controllers and monitoring instruments with the information necessary for operation. The need for other types of information, such as audio and video, has grown. Suitable applications range in a broad spectrum from microelectromechanical systems and bio-medical engineering to papermaking and steel production. This thesis is divided into f...

A feasibility study of stateful automaton packet inspection for streaming application detection systems

Science.gov (United States)

Tseng, Kuo-Kun; Lo, Jiao; Liu, Yiming; Chang, Shih-Hao; Merabti, Madjid; Ng, Felix, C. K.; Wu, C. H.

2017-10-01

The rapid development of the internet has brought huge benefits and social impacts; however, internet security has also become a great problem for users, since traditional approaches to packet classification cannot achieve satisfactory detection performance due to their low accuracy and efficiency. In this paper, a new stateful packet inspection method is introduced, which can be embedded in the network gateway and used by a streaming application detection system. This new detection method leverages the inexact automaton approach, using part of the header field and part of the application layer data of a packet. Based on this approach, an advanced detection system is proposed for streaming applications. The workflow of the system involves two stages: the training stage and the detection stage. In the training stage, the system initially captures characteristic patterns from a set of application packet flows. After this training is completed, the detection stage allows the user to detect the target application by capturing new application flows. This new detection approach is also evaluated using experimental analysis; the results of this analysis show that this new approach not only simplifies the management of the state detection system, but also improves the accuracy of data flow detection, making it feasible for real-world network applications.

InSTREAM: the individual-based stream trout research and environmental assessment model

Science.gov (United States)

Steven F. Railsback; Bret C. Harvey; Stephen K. Jackson; Roland H. Lamberson

2009-01-01

This report documents Version 4.2 of InSTREAM, including its formulation, software, and application to research and management problems. InSTREAM is a simulation model designed to understand how stream and river salmonid populations respond to habitat alteration, including altered flow, temperature, and turbidity regimes and changes in channel morphology. The model...

The Chameleon Architecture for Streaming DSP Applications

Directory of Open Access Journals (Sweden)

André B. J. Kokkeler

2007-02-01

Full Text Available We focus on architectures for streaming DSP applications such as wireless baseband processing and image processing. We aim at a single generic architecture that is capable of dealing with different DSP applications. This architecture has to be energy efficient and fault tolerant. We introduce a heterogeneous tiled architecture and present the details of a domain-specific reconfigurable tile processor called Montium. This reconfigurable processor has a small footprint (1.8 mm2 in a 130 nm process, is power efficient and exploits the locality of reference principle. Reconfiguring the device is very fast, for example, loading the coefficients for a 200 tap FIR filter is done within 80 clock cycles. The tiles on the tiled architecture are connected to a Network-on-Chip (NoC via a network interface (NI. Two NoCs have been developed: a packet-switched and a circuit-switched version. Both provide two types of services: guaranteed throughput (GT and best effort (BE. For both NoCs estimates of power consumption are presented. The NI synchronizes data transfers, configures and starts/stops the tile processor. For dynamically mapping applications onto the tiled architecture, we introduce a run-time mapping tool.

The Chameleon Architecture for Streaming DSP Applications

Directory of Open Access Journals (Sweden)

Heysters PaulM

2007-01-01

Full Text Available We focus on architectures for streaming DSP applications such as wireless baseband processing and image processing. We aim at a single generic architecture that is capable of dealing with different DSP applications. This architecture has to be energy efficient and fault tolerant. We introduce a heterogeneous tiled architecture and present the details of a domain-specific reconfigurable tile processor called Montium. This reconfigurable processor has a small footprint (1.8 mm2 in a 130 nm process, is power efficient and exploits the locality of reference principle. Reconfiguring the device is very fast, for example, loading the coefficients for a 200 tap FIR filter is done within 80 clock cycles. The tiles on the tiled architecture are connected to a Network-on-Chip (NoC via a network interface (NI. Two NoCs have been developed: a packet-switched and a circuit-switched version. Both provide two types of services: guaranteed throughput (GT and best effort (BE. For both NoCs estimates of power consumption are presented. The NI synchronizes data transfers, configures and starts/stops the tile processor. For dynamically mapping applications onto the tiled architecture, we introduce a run-time mapping tool.

The Fungible Audio-Visual Mapping and its Experience

Directory of Open Access Journals (Sweden)

Adriana Sa

2014-12-01

Full Text Available This article draws a perceptual approach to audio-visual mapping. Clearly perceivable cause and effect relationships can be problematic if one desires the audience to experience the music. Indeed perception would bias those sonic qualities that fit previous concepts of causation, subordinating other sonic qualities, which may form the relations between the sounds themselves. The question is, how can an audio-visual mapping produce a sense of causation, and simultaneously confound the actual cause-effect relationships. We call this a fungible audio-visual mapping. Our aim here is to glean its constitution and aspect. We will report a study, which draws upon methods from experimental psychology to inform audio-visual instrument design and composition. The participants are shown several audio-visual mapping prototypes, after which we pose quantitative and qualitative questions regarding their sense of causation, and their sense of understanding the cause-effect relationships. The study shows that a fungible mapping requires both synchronized and seemingly non-related components – sufficient complexity to be confusing. As the specific cause-effect concepts remain inconclusive, the sense of causation embraces the whole. 

Efficiency Optimization in Class-D Audio Amplifiers

DEFF Research Database (Denmark)

Yamauchi, Akira; Knott, Arnold; Jørgensen, Ivan Harald Holger

2015-01-01

This paper presents a new power efficiency optimization routine for designing Class-D audio amplifiers. The proposed optimization procedure finds design parameters for the power stage and the output filter, and the optimum switching frequency such that the weighted power losses are minimized under...... the given constraints. The optimization routine is applied to minimize the power losses in a 130 W class-D audio amplifier based on consumer behavior investigations, where the amplifier operates at idle and low power levels most of the time. Experimental results demonstrate that the optimization method can...... lead to around 30 % of efficiency improvement at 1.3 W output power without significant effects on both audio performance and the efficiency at high power levels....

Predistortion of a Bidirectional Cuk Audio Amplifier

DEFF Research Database (Denmark)

Birch, Thomas Hagen; Nielsen, Dennis; Knott, Arnold

2014-01-01

Some non-linear amplifier topologies are capable of providing a larger voltage gain than one from a DC source, which could make them suitable for various applications. However, the non-linearities introduce a significant amount of harmonic distortion (THD). Some of this distortion could be reduced...... using predistortion. This paper suggests linearizing a nonlinear bidirectional Cuk audio amplifier using an analog predistortion approach. A prototype power stage was built and results show that a voltage gain of up to 9 dB and reduction in THD from 6% down to 3% was obtainable using this approach....

DOA Estimation of Audio Sources in Reverberant Environments

DEFF Research Database (Denmark)

Jensen, Jesper Rindom; Nielsen, Jesper Kjær; Heusdens, Richard

2016-01-01

Reverberation is well-known to have a detrimental impact on many localization methods for audio sources. We address this problem by imposing a model for the early reflections as well as a model for the audio source itself. Using these models, we propose two iterative localization methods...... that estimate the direction-of-arrival (DOA) of both the direct path of the audio source and the early reflections. In these methods, the contribution of the early reflections is essentially subtracted from the signal observations before localization of the direct path component, which may reduce the estimation...

Portable Audio Design

DEFF Research Database (Denmark)

Groth, Sanne Krogh

2014-01-01

attention to the specific genre; a grasping of the complex relationship between site and time, the actual and the virtual; and getting aquatint with the specific site’s soundscape by approaching it both intuitively and systematically. These steps will finally lead to an audio production that not only...

Adaptive DCTNet for Audio Signal Classification

OpenAIRE

Xian, Yin; Pu, Yunchen; Gan, Zhe; Lu, Liang; Thompson, Andrew

2016-01-01

In this paper, we investigate DCTNet for audio signal classification. Its output feature is related to Cohen's class of time-frequency distributions. We introduce the use of adaptive DCTNet (A-DCTNet) for audio signals feature extraction. The A-DCTNet applies the idea of constant-Q transform, with its center frequencies of filterbanks geometrically spaced. The A-DCTNet is adaptive to different acoustic scales, and it can better capture low frequency acoustic information that is sensitive to h...

RStorm: Developing and Testing Streaming Algorithms in R

NARCIS (Netherlands)

Kaptein, M.C.

2014-01-01

Streaming data, consisting of indefinitely evolving sequences, are becoming ubiquitous in many branches of science and in various applications. Computer scientists have developed streaming applications such as Storm and the S4 distributed stream computing platform1 to deal with data streams.

RStorm : Developing and testing streaming algorithms in R

NARCIS (Netherlands)

Kaptein, M.C.

2014-01-01

Streaming data, consisting of indefinitely evolving sequences, are becoming ubiquitous in many branches of science and in various applications. Computer scientists have developed streaming applications such as Storm and the S4 distributed stream computing platform1 to deal with data streams.

Audio-Tutorial Instruction: A Strategy For Teaching Introductory College Geology.

Science.gov (United States)

Fenner, Peter; Andrews, Ted F.

The rationale of audio-tutorial instruction is discussed, and the history and development of the audio-tutorial botany program at Purdue University is described. Audio-tutorial programs in geology at eleven colleges and one school are described, illustrating several ways in which programs have been developed and integrated into courses. Programs…

Implementation and Analysis Audio Steganography Used Parity Coding for Symmetric Cryptography Key Delivery

Directory of Open Access Journals (Sweden)

Afany Zeinata Firdaus

2013-12-01

Full Text Available In today's era of communication, online data transactions is increasing. Various information even more accessible, both upload and download. Because it takes a capable security system. Blowfish cryptographic equipped with Audio Steganography is one way to secure the data so that the data can not be accessed by unauthorized parties. In this study Audio Steganography technique is implemented using parity coding method that is used to send the key cryptography blowfish in e-commerce applications based on Android. The results obtained for the average computation time on stage insertion (embedding the secret message is shorter than the average computation time making phase (extracting the secret message. From the test results can also be seen that the more the number of characters pasted the greater the noise received, where the highest SNR is obtained when a character is inserted as many as 506 characters is equal to 11.9905 dB, while the lowest SNR obtained when a character is inserted as many as 2006 characters at 5,6897 dB . Keywords: audio steganograph, parity coding, embedding, extractin, cryptography blowfih.

Teacher’s Voice on Metacognitive Strategy Based Instruction Using Audio Visual Aids for Listening

Directory of Open Access Journals (Sweden)

Salasiah Salasiah

2018-02-01

Full Text Available The paper primarily stresses on exploring the teacher’s voice toward the application of metacognitive strategy with audio-visual aid in improving listening comprehension. The metacognitive strategy model applied in the study was inspired from Vandergrift and Tafaghodtari (2010 instructional model. Thus it is modified in the procedure and applied with audio-visual aids for improving listening comprehension. The study’s setting was at SMA Negeri 2 Parepare, South Sulawesi Province, Indonesia. The population of the research was the teacher of English at tenth grade at SMAN 2. The sample was taken by using random sampling technique. The data was collected by using in depth interview during the research, recorded, and analyzed using qualitative analysis. This study explored the teacher’s response toward the modified model of metacognitive strategy with audio visual aids in class of listening which covers positive and negative response toward the strategy applied during the teaching of listening. The result of data showed that this strategy helped the teacher a lot in teaching listening comprehension as the procedure has systematic steps toward students’ listening comprehension. Also, it eases the teacher to teach listening by empowering audio visual aids such as video taken from youtube.

StreamStats in Oklahoma - Drainage-Basin Characteristics and Peak-Flow Frequency Statistics for Ungaged Streams

Science.gov (United States)

Smith, S. Jerrod; Esralew, Rachel A.

2010-01-01

The USGS Streamflow Statistics (StreamStats) Program was created to make geographic information systems-based estimation of streamflow statistics easier, faster, and more consistent than previously used manual techniques. The StreamStats user interface is a map-based internet application that allows users to easily obtain streamflow statistics, basin characteristics, and other information for user-selected U.S. Geological Survey data-collection stations and ungaged sites of interest. The application relies on the data collected at U.S. Geological Survey streamflow-gaging stations, computer aided computations of drainage-basin characteristics, and published regression equations for several geographic regions comprising the United States. The StreamStats application interface allows the user to (1) obtain information on features in selected map layers, (2) delineate drainage basins for ungaged sites, (3) download drainage-basin polygons to a shapefile, (4) compute selected basin characteristics for delineated drainage basins, (5) estimate selected streamflow statistics for ungaged points on a stream, (6) print map views, (7) retrieve information for U.S. Geological Survey streamflow-gaging stations, and (8) get help on using StreamStats. StreamStats was designed for national application, with each state, territory, or group of states responsible for creating unique geospatial datasets and regression equations to compute selected streamflow statistics. With the cooperation of the Oklahoma Department of Transportation, StreamStats has been implemented for Oklahoma and is available at http://water.usgs.gov/osw/streamstats/. The Oklahoma StreamStats application covers 69 processed hydrologic units and most of the state of Oklahoma. Basin characteristics available for computation include contributing drainage area, contributing drainage area that is unregulated by Natural Resources Conservation Service floodwater retarding structures, mean-annual precipitation at the

Audio Description as a Pedagogical Tool

Directory of Open Access Journals (Sweden)

Georgina Kleege

2015-05-01

Full Text Available Audio description is the process of translating visual information into words for people who are blind or have low vision. Typically such description has focused on films, museum exhibitions, images and video on the internet, and live theater. Because it allows people with visual impairments to experience a variety of cultural and educational texts that would otherwise be inaccessible, audio description is a mandated aspect of disability inclusion, although it remains markedly underdeveloped and underutilized in our classrooms and in society in general. Along with increasing awareness of disability, audio description pushes students to practice close reading of visual material, deepen their analysis, and engage in critical discussions around the methodology, standards and values, language, and role of interpretation in a variety of academic disciplines. We outline a few pedagogical interventions that can be customized to different contexts to develop students' writing and critical thinking skills through guided description of visual material.

Extracting meaning from audio signals - a machine learning approach

DEFF Research Database (Denmark)

Larsen, Jan

2007-01-01

* Machine learning framework for sound search * Genre classification * Music and audio separation * Wind noise suppression......* Machine learning framework for sound search * Genre classification * Music and audio separation * Wind noise suppression...

Efficiency in audio processing : filter banks and transcoding

NARCIS (Netherlands)

Lee, Jun Wei

2007-01-01

Audio transcoding is the conversion of digital audio from one compressed form A to another compressed form B, where A and B have different compression properties, such as a different bit-rate, sampling frequency or compression method. This is typically achieved by decoding A to an intermediate

Decision-level fusion for audio-visual laughter detection

NARCIS (Netherlands)

Reuderink, B.; Poel, M.; Truong, K.; Poppe, R.; Pantic, M.

2008-01-01

Laughter is a highly variable signal, which can be caused by a spectrum of emotions. This makes the automatic detection of laughter a challenging, but interesting task. We perform automatic laughter detection using audio-visual data from the AMI Meeting Corpus. Audio-visual laughter detection is

Decision-Level Fusion for Audio-Visual Laughter Detection

NARCIS (Netherlands)

Reuderink, B.; Poel, Mannes; Truong, Khiet Phuong; Poppe, Ronald Walter; Pantic, Maja; Popescu-Belis, Andrei; Stiefelhagen, Rainer

Laughter is a highly variable signal, which can be caused by a spectrum of emotions. This makes the automatic detection of laugh- ter a challenging, but interesting task. We perform automatic laughter detection using audio-visual data from the AMI Meeting Corpus. Audio- visual laughter detection is

Content Discovery from Composite Audio : An unsupervised approach

NARCIS (Netherlands)

Lu, L.

2009-01-01

In this thesis, we developed and assessed a novel robust and unsupervised framework for semantic inference from composite audio signals. We focused on the problem of detecting audio scenes and grouping them into meaningful clusters. Our approach addressed all major steps in a general process of

Paper-Based Textbooks with Audio Support for Print-Disabled Students.

Science.gov (United States)

Fujiyoshi, Akio; Ohsawa, Akiko; Takaira, Takuya; Tani, Yoshiaki; Fujiyoshi, Mamoru; Ota, Yuko

2015-01-01

Utilizing invisible 2-dimensional codes and digital audio players with a 2-dimensional code scanner, we developed paper-based textbooks with audio support for students with print disabilities, called "multimodal textbooks." Multimodal textbooks can be read with the combination of the two modes: "reading printed text" and "listening to the speech of the text from a digital audio player with a 2-dimensional code scanner." Since multimodal textbooks look the same as regular textbooks and the price of a digital audio player is reasonable (about 30 euro), we think multimodal textbooks are suitable for students with print disabilities in ordinary classrooms.

Stream.cz a jeho originální seriálová tvorba

OpenAIRE

Vašíčková, Dorota

2017-01-01

In general, the rise of the internet has brought many changes of production and distribution into the audio-visual industry. These changes triggered the development of specific internet portals that offer video content. The thesis focuses on the current development of internet televisions and especially on a particular platform and enriches the current theoretical list of forms with a specific example of a Czech internet television. The case study of the Czech internet television Stream.cz de...

Sounding ruins: reflections on the production of an ‘audio drift’

Science.gov (United States)

Gallagher, Michael

2014-01-01

This article is about the use of audio media in researching places, which I term ‘audio geography’. The article narrates some episodes from the production of an ‘audio drift’, an experimental environmental sound work designed to be listened to on a portable MP3 player whilst walking in a ruinous landscape. Reflecting on how this work functions, I argue that, as well as representing places, audio geography can shape listeners’ attention and bodily movements, thereby reworking places, albeit temporarily. I suggest that audio geography is particularly apt for amplifying the haunted and uncanny qualities of places. I discuss some of the issues raised for research ethics, epistemology and spectral geographies. PMID:29708107

Interactive 3D audio: Enhancing awareness of details in immersive soundscapes?

DEFF Research Database (Denmark)

Schmidt, Mikkel Nørgaard; Schwartz, Stephen; Larsen, Jan

2012-01-01

Spatial audio and the possibility of interacting with the audio environment is thought to increase listeners' attention to details in a soundscape. This work examines if interactive 3D audio enhances listeners' ability to recall details in a soundscape. Nine different soundscapes were constructed...

Personalized Audio Systems - a Bayesian Approach

DEFF Research Database (Denmark)

Nielsen, Jens Brehm; Jensen, Bjørn Sand; Hansen, Toke Jansen

2013-01-01

Modern audio systems are typically equipped with several user-adjustable parameters unfamiliar to most users listening to the system. To obtain the best possible setting, the user is forced into multi-parameter optimization with respect to the users's own objective and preference. To address this......, the present paper presents a general inter-active framework for personalization of such audio systems. The framework builds on Bayesian Gaussian process regression in which a model of the users's objective function is updated sequentially. The parameter setting to be evaluated in a given trial is selected...

IELTS speaking instruction through audio/voice conferencing

Directory of Open Access Journals (Sweden)

Hamed Ghaemi

2012-02-01

Full Text Available The currentstudyaimsatinvestigatingtheimpactofAudio/Voiceconferencing,asanewapproachtoteaching speaking, on the speakingperformanceand/orspeakingband score ofIELTScandidates.Experimentalgroupsubjectsparticipated in an audio conferencing classwhile those of the control group enjoyed attending in a traditional IELTS Speakingclass. At the endofthestudy,allsubjectsparticipatedinanIELTSExaminationheldonNovemberfourthin Tehran,Iran.To compare thegroupmeansforthestudy,anindependentt-testanalysiswasemployed.Thedifferencebetween experimental and control groupwasconsideredtobestatisticallysignificant(P<0.01.Thatisthecandidates in experimental group have outperformed the ones in control group in IELTS Speaking test scores.

A high efficiency PWM CMOS class-D audio power amplifier

Energy Technology Data Exchange (ETDEWEB)

Zhu Zhangming; Liu Lianxi; Yang Yintang [Institute of Microelectronics, Xidian University, Xi' an 710071 (China); Lei Han, E-mail: zmyh@263.ne [Xi' an Power-Rail Micro Co., Ltd, Xi' an 710075 (China)

2009-02-15

Based on the difference close-loop feedback technique and the difference pre-amp, a high efficiency PWM CMOS class-D audio power amplifier is proposed. A rail-to-rail PWM comparator with window function has been embedded in the class-D audio power amplifier. Design results based on the CSMC 0.5 mum CMOS process show that the max efficiency is 90%, the PSRR is -75 dB, the power supply voltage range is 2.5-5.5 V, the THD+N in 1 kHz input frequency is less than 0.20%, the quiescent current in no load is 2.8 mA, and the shutdown current is 0.5 muA. The active area of the class-D audio power amplifier is about 1.47 x 1.52 mm{sup 2}. With the good performance, the class-D audio power amplifier can be applied to several audio power systems.

A high efficiency PWM CMOS class-D audio power amplifier

International Nuclear Information System (INIS)

Zhu Zhangming; Liu Lianxi; Yang Yintang; Lei Han

2009-01-01

Based on the difference close-loop feedback technique and the difference pre-amp, a high efficiency PWM CMOS class-D audio power amplifier is proposed. A rail-to-rail PWM comparator with window function has been embedded in the class-D audio power amplifier. Design results based on the CSMC 0.5 μm CMOS process show that the max efficiency is 90%, the PSRR is -75 dB, the power supply voltage range is 2.5-5.5 V, the THD+N in 1 kHz input frequency is less than 0.20%, the quiescent current in no load is 2.8 mA, and the shutdown current is 0.5 μA. The active area of the class-D audio power amplifier is about 1.47 x 1.52 mm 2 . With the good performance, the class-D audio power amplifier can be applied to several audio power systems.

A high efficiency PWM CMOS class-D audio power amplifier

Science.gov (United States)

Zhangming, Zhu; Lianxi, Liu; Yintang, Yang; Han, Lei

2009-02-01

Based on the difference close-loop feedback technique and the difference pre-amp, a high efficiency PWM CMOS class-D audio power amplifier is proposed. A rail-to-rail PWM comparator with window function has been embedded in the class-D audio power amplifier. Design results based on the CSMC 0.5 μm CMOS process show that the max efficiency is 90%, the PSRR is -75 dB, the power supply voltage range is 2.5-5.5 V, the THD+N in 1 kHz input frequency is less than 0.20%, the quiescent current in no load is 2.8 mA, and the shutdown current is 0.5 μA. The active area of the class-D audio power amplifier is about 1.47 × 1.52 mm2. With the good performance, the class-D audio power amplifier can be applied to several audio power systems.

Audio-Visual Fusion for Sound Source Localization and Improved Attention

Energy Technology Data Exchange (ETDEWEB)

Lee, Byoung Gi; Choi, Jong Suk; Yoon, Sang Suk; Choi, Mun Taek; Kim, Mun Sang [Korea Institute of Science and Technology, Daejeon (Korea, Republic of); Kim, Dai Jin [Pohang University of Science and Technology, Pohang (Korea, Republic of)

2011-07-15

Service robots are equipped with various sensors such as vision camera, sonar sensor, laser scanner, and microphones. Although these sensors have their own functions, some of them can be made to work together and perform more complicated functions. AudioFvisual fusion is a typical and powerful combination of audio and video sensors, because audio information is complementary to visual information and vice versa. Human beings also mainly depend on visual and auditory information in their daily life. In this paper, we conduct two studies using audioFvision fusion: one is on enhancing the performance of sound localization, and the other is on improving robot attention through sound localization and face detection.

Audio-Visual Fusion for Sound Source Localization and Improved Attention

International Nuclear Information System (INIS)

Lee, Byoung Gi; Choi, Jong Suk; Yoon, Sang Suk; Choi, Mun Taek; Kim, Mun Sang; Kim, Dai Jin

2011-01-01

Service robots are equipped with various sensors such as vision camera, sonar sensor, laser scanner, and microphones. Although these sensors have their own functions, some of them can be made to work together and perform more complicated functions. AudioFvisual fusion is a typical and powerful combination of audio and video sensors, because audio information is complementary to visual information and vice versa. Human beings also mainly depend on visual and auditory information in their daily life. In this paper, we conduct two studies using audioFvision fusion: one is on enhancing the performance of sound localization, and the other is on improving robot attention through sound localization and face detection

A review of lossless audio compression standards and algorithms

Science.gov (United States)

Muin, Fathiah Abdul; Gunawan, Teddy Surya; Kartiwi, Mira; Elsheikh, Elsheikh M. A.

2017-09-01

Over the years, lossless audio compression has gained popularity as researchers and businesses has become more aware of the need for better quality and higher storage demand. This paper will analyse various lossless audio coding algorithm and standards that are used and available in the market focusing on Linear Predictive Coding (LPC) specifically due to its popularity and robustness in audio compression, nevertheless other prediction methods are compared to verify this. Advanced representation of LPC such as LSP decomposition techniques are also discussed within this paper.

WebGL and web audio software lightweight components for multimedia education

Science.gov (United States)

Chang, Xin; Yuksel, Kivanc; Skarbek, Władysław

2017-08-01

The paper presents the results of our recent work on development of contemporary computing platform DC2 for multimedia education usingWebGL andWeb Audio { the W3C standards. Using literate programming paradigm the WEBSA educational tools were developed. It offers for a user (student), the access to expandable collection of WEBGL Shaders and web Audio scripts. The unique feature of DC2 is the option of literate programming, offered for both, the author and the reader in order to improve interactivity to lightweightWebGL andWeb Audio components. For instance users can define: source audio nodes including synthetic sources, destination audio nodes, and nodes for audio processing such as: sound wave shaping, spectral band filtering, convolution based modification, etc. In case of WebGL beside of classic graphics effects based on mesh and fractal definitions, the novel image processing analysis by shaders is offered like nonlinear filtering, histogram of gradients, and Bayesian classifiers.

On Modeling Affect in Audio with Non-Linear Symbolic Dynamics

Directory of Open Access Journals (Sweden)

Pauline Mouawad

2017-09-01

Full Text Available The discovery of semantic information from complex signals is a task concerned with connecting humans’ perceptions and/or intentions with the signals content. In the case of audio signals, complex perceptions are appraised in a listener’s mind, that trigger affective responses that may be relevant for well-being and survival. In this paper we are interested in the broader question of relations between uncertainty in data as measured using various information criteria and emotions, and we propose a novel method that combines nonlinear dynamics analysis with a method of adaptive time series symbolization that finds the meaningful audio structure in terms of symbolized recurrence properties. In a first phase we obtain symbolic recurrence quantification measures from symbolic recurrence plots, without the need to reconstruct the phase space with embedding. Then we estimate symbolic dynamical invariants from symbolized time series, after embedding. The invariants are: correlation dimension, correlation entropy and Lyapunov exponent. Through their application for the logistic map, we show that our measures are in agreement with known methods from literature. We further show that one symbolic recurrence measure, namely the symbolic Shannon entropy, correlates positively with the positive Lyapunov exponents. Finally we evaluate the performance of our measures in emotion recognition through the implementation of classification tasks for different types of audio signals, and show that in some cases, they perform better than state-of-the-art methods that rely on low-level acoustic features.

Enhanced audio-visual interactions in the auditory cortex of elderly cochlear-implant users.

Science.gov (United States)

Schierholz, Irina; Finke, Mareike; Schulte, Svenja; Hauthal, Nadine; Kantzke, Christoph; Rach, Stefan; Büchner, Andreas; Dengler, Reinhard; Sandmann, Pascale

2015-10-01

Auditory deprivation and the restoration of hearing via a cochlear implant (CI) can induce functional plasticity in auditory cortical areas. How these plastic changes affect the ability to integrate combined auditory (A) and visual (V) information is not yet well understood. In the present study, we used electroencephalography (EEG) to examine whether age, temporary deafness and altered sensory experience with a CI can affect audio-visual (AV) interactions in post-lingually deafened CI users. Young and elderly CI users and age-matched NH listeners performed a speeded response task on basic auditory, visual and audio-visual stimuli. Regarding the behavioral results, a redundant signals effect, that is, faster response times to cross-modal (AV) than to both of the two modality-specific stimuli (A, V), was revealed for all groups of participants. Moreover, in all four groups, we found evidence for audio-visual integration. Regarding event-related responses (ERPs), we observed a more pronounced visual modulation of the cortical auditory response at N1 latency (approximately 100 ms after stimulus onset) in the elderly CI users when compared with young CI users and elderly NH listeners. Thus, elderly CI users showed enhanced audio-visual binding which may be a consequence of compensatory strategies developed due to temporary deafness and/or degraded sensory input after implantation. These results indicate that the combination of aging, sensory deprivation and CI facilitates the coupling between the auditory and the visual modality. We suggest that this enhancement in multisensory interactions could be used to optimize auditory rehabilitation, especially in elderly CI users, by the application of strong audio-visually based rehabilitation strategies after implant switch-on. Copyright © 2015 Elsevier B.V. All rights reserved.

Class D audio amplifiers for high voltage capacitive transducers

DEFF Research Database (Denmark)

Nielsen, Dennis

of high volume, weight, and cost. High efficient class D amplifiers are now widely available offering power densities, that their linear counterparts can not match. Unlike the technology of audio amplifiers, the loudspeaker is still based on the traditional electrodynamic transducer invented by C.W. Rice......Audio reproduction systems contains two key components, the amplifier and the loudspeaker. In the last 20 – 30 years the technology of audio amplifiers have performed a fundamental shift of paradigm. Class D audio amplifiers have replaced the linear amplifiers, suffering from the well-known issues...... with the low level of acoustical output power and complex amplifier requirements, have limited the commercial success of the technology. Horn or compression drivers are typically favoured, when high acoustic output power is required, this is however at the expense of significant distortion combined...

Can audio recording improve patients' recall of outpatient consultations?

DEFF Research Database (Denmark)

Wolderslund, Maiken; Kofoed, Poul-Erik; Axboe, Mette

Introduction In order to give patients possibility to listen to their consultation again, we have designed a system which gives the patients access to digital audio recordings of their consultations. An Interactive Voice Response platform enables the audio recording and gives the patients access...... and those who have not (control).The audio recordings and the interviews are coded according to six themes: Test results, Treatment, Risks, Future tests, Advice and Plan. Afterwards the extent of patients recall is assessed by comparing the accuracy of the patient’s statements (interview...

Accuracy improvement of dataflow analysis for cyclic stream processing applications scheduled by static priority preemptive schedulers

NARCIS (Netherlands)

Kurtin, Philip Sebastian; Hausmans, J.P.H.M.; Geuns, S.J.; Bekooij, Marco Jan Gerrit

2014-01-01

Stream processing applications executed on embedded multiprocessor systems regularly contain cyclic data dependencies due to the presence of feedback loops and bounded FIFO buffers. Dataflow modeling is suitable for the temporal analysis of such applications. However, the accuracy can be

Performance Evaluation of UML2-Modeled Embedded Streaming Applications with System-Level Simulation

Directory of Open Access Journals (Sweden)

Arpinen Tero

2009-01-01

Full Text Available This article presents an efficient method to capture abstract performance model of streaming data real-time embedded systems (RTESs. Unified Modeling Language version 2 (UML2 is used for the performance modeling and as a front-end for a tool framework that enables simulation-based performance evaluation and design-space exploration. The adopted application meta-model in UML resembles the Kahn Process Network (KPN model and it is targeted at simulation-based performance evaluation. The application workload modeling is done using UML2 activity diagrams, and platform is described with structural UML2 diagrams and model elements. These concepts are defined using a subset of the profile for Modeling and Analysis of Realtime and Embedded (MARTE systems from OMG and custom stereotype extensions. The goal of the performance modeling and simulation is to achieve early estimates on task response times, processing element, memory, and on-chip network utilizations, among other information that is used for design-space exploration. As a case study, a video codec application on multiple processors is modeled, evaluated, and explored. In comparison to related work, this is the first proposal that defines transformation between UML activity diagrams and streaming data application workload meta models and successfully adopts it for RTES performance evaluation.

Audio Networking in the Music Industry

Directory of Open Access Journals (Sweden)

Glebs Kuzmics

2018-01-01

Full Text Available This paper surveys the rôle of computer networking technologies in the music industry. A comparison of their relevant technologies, their defining advantages and disadvantages; analyses and discussion of the situation in the market of network enabled audio products followed by a discussion of different devices are presented. The idea of replacing a proprietary solution with open-source and freeware software programs has been chosen as the fundamental concept of this research. The technologies covered include: native IEEE AVnu Alliance Audio Video Bridging (AVB, CobraNet®, Audinate Dante™ and Harman BLU Link.

Frequency Hopping Method for Audio Watermarking

Directory of Open Access Journals (Sweden)

A. Anastasijević

2012-11-01

Full Text Available This paper evaluates the degradation of audio content for a perceptible removable watermark. Two different approaches to embedding the watermark in the spectral domain were investigated. The frequencies for watermark embedding are chosen according to a pseudorandom sequence making the methods robust. Consequentially, the lower quality audio can be used for promotional purposes. For a fee, the watermark can be removed with a secret watermarking key. Objective and subjective testing was conducted in order to measure degradation level for the watermarked music samples and to examine residual distortion for different parameters of the watermarking algorithm and different music genres.

Four-quadrant flyback converter for direct audio power amplification

DEFF Research Database (Denmark)

Ljusev, Petar; Andersen, Michael Andreas E.

2005-01-01

This paper presents a bidirectional, four-quadrant flyback converter for use in direct audio power amplification. When compared to the standard Class-D switching audio power amplifier with a separate power supply, the proposed four-quadrant flyback converter provides simple solution with better...

Audio Technology and Mobile Human Computer Interaction

DEFF Research Database (Denmark)

Chamberlain, Alan; Bødker, Mads; Hazzard, Adrian

2017-01-01

Audio-based mobile technology is opening up a range of new interactive possibilities. This paper brings some of those possibilities to light by offering a range of perspectives based in this area. It is not only the technical systems that are developing, but novel approaches to the design...... and understanding of audio-based mobile systems are evolving to offer new perspectives on interaction and design and support such systems to be applied in areas, such as the humanities....

Unsupervised topic modelling on South African parliament audio data

CSIR Research Space (South Africa)

Kleynhans, N

2014-11-01

Full Text Available Using a speech recognition system to convert spoken audio to text can enable the structuring of large collections of spoken audio data. A convenient means to summarise or cluster spoken data is to identify the topic under discussion. There are many...

Classifying laughter and speech using audio-visual feature prediction

NARCIS (Netherlands)

Petridis, Stavros; Asghar, Ali; Pantic, Maja

2010-01-01

In this study, a system that discriminates laughter from speech by modelling the relationship between audio and visual features is presented. The underlying assumption is that this relationship is different between speech and laughter. Neural networks are trained which learn the audio-to-visual and

Streaming video - præsentation af trailere over internettet

DEFF Research Database (Denmark)

Jensen, Ole Riis; Forchhammer, Søren

1998-01-01

interaktiv tjeneste baseret på realtidsfremvisning af videotrailere, såkaldt streaming video og audio, og hvor brugeren kan få information om en film der tænkes udbudt på en pay-per-view kanal. Der i arbejdets forløb blevet opbygget en demo, der implementerer en sådan tjeneste.......Som et led i Tele Danmark Kabel TV's markedsføring ønsker man at kunne præsentere filmtrailere og andet materiale i form af levende billeder til kunderne via Wold Wide Web (WWW). Dette projekts hovedformål er at undersøge eksisterende metoder og udvikle redskaber til at præsentere en tilsvarende...

Sistema de adquisición y procesamiento de audio

OpenAIRE

Pérez Segurado, Rubén

2015-01-01

El objetivo de este proyecto es el diseño y la implementación de una plataforma para un sistema de procesamiento de audio. El sistema recibirá una señal de audio analógica desde una fuente de audio, permitirá realizar un tratamiento digital de dicha señal y generará una señal procesada que se enviará a unos altavoces externos. Para la realización del sistema de procesamiento se empleará: - Un dispositivo FPGA de Lattice, modelo MachX02-7000-HE, en la cual estarán todas la...

Analytical Features: A Knowledge-Based Approach to Audio Feature Generation

Directory of Open Access Journals (Sweden)

Pachet François

2009-01-01

Full Text Available We present a feature generation system designed to create audio features for supervised classification tasks. The main contribution to feature generation studies is the notion of analytical features (AFs, a construct designed to support the representation of knowledge about audio signal processing. We describe the most important aspects of AFs, in particular their dimensional type system, on which are based pattern-based random generators, heuristics, and rewriting rules. We show how AFs generalize or improve previous approaches used in feature generation. We report on several projects using AFs for difficult audio classification tasks, demonstrating their advantage over standard audio features. More generally, we propose analytical features as a paradigm to bring raw signals into the world of symbolic computation.

Audible Aliasing Distortion in Digital Audio Synthesis

Directory of Open Access Journals (Sweden)

J. Schimmel

2012-04-01

Full Text Available This paper deals with aliasing distortion in digital audio signal synthesis of classic periodic waveforms with infinite Fourier series, for electronic musical instruments. When these waveforms are generated in the digital domain then the aliasing appears due to its unlimited bandwidth. There are several techniques for the synthesis of these signals that have been designed to avoid or reduce the aliasing distortion. However, these techniques have high computing demands. One can say that today's computers have enough computing power to use these methods. However, we have to realize that today’s computer-aided music production requires tens of multi-timbre voices generated simultaneously by software synthesizers and the most of the computing power must be reserved for hard-disc recording subsystem and real-time audio processing of many audio channels with a lot of audio effects. Trivially generated classic analog synthesizer waveforms are therefore still effective for sound synthesis. We cannot avoid the aliasing distortion but spectral components produced by the aliasing can be masked with harmonic components and thus made inaudible if sufficient oversampling ratio is used. This paper deals with the assessment of audible aliasing distortion with the help of a psychoacoustic model of simultaneous masking and compares the computing demands of trivial generation using oversampling with those of other methods.

ENERGY STAR Certified Audio Video

Data.gov (United States)

U.S. Environmental Protection Agency — Certified models meet all ENERGY STAR requirements as listed in the Version 3.0 ENERGY STAR Program Requirements for Audio Video Equipment that are effective as of...

Applications of spatial statistical network models to stream data

Science.gov (United States)

Daniel J. Isaak; Erin E. Peterson; Jay M. Ver Hoef; Seth J. Wenger; Jeffrey A. Falke; Christian E. Torgersen; Colin Sowder; E. Ashley Steel; Marie-Josee Fortin; Chris E. Jordan; Aaron S. Ruesch; Nicholas Som; Pascal. Monestiez

2014-01-01

Streams and rivers host a significant portion of Earth's biodiversity and provide important ecosystem services for human populations. Accurate information regarding the status and trends of stream resources is vital for their effective conservation and management. Most statistical techniques applied to data measured on stream networks were developed for...

Text Stream Trend Analysis using Multiscale Visual Analytics with Applications to Social Media Systems

Energy Technology Data Exchange (ETDEWEB)

Steed, Chad A [ORNL; Beaver, Justin M [ORNL; BogenII, Paul L. [Google Inc.; Drouhard, Margaret MEG G [ORNL; Pyle, Joshua M [ORNL

2015-01-01

In this paper, we introduce a new visual analytics system, called Matisse, that allows exploration of global trends in textual information streams with specific application to social media platforms. Despite the potential for real-time situational awareness using these services, interactive analysis of such semi-structured textual information is a challenge due to the high-throughput and high-velocity properties. Matisse addresses these challenges through the following contributions: (1) robust stream data management, (2) automated sen- timent/emotion analytics, (3) inferential temporal, geospatial, and term-frequency visualizations, and (4) a flexible drill-down interaction scheme that progresses from macroscale to microscale views. In addition to describing these contributions, our work-in-progress paper concludes with a practical case study focused on the analysis of Twitter 1% sample stream information captured during the week of the Boston Marathon bombings.

Audio-Visual Feedback for Self-monitoring Posture in Ballet Training

DEFF Research Database (Denmark)

Knudsen, Esben Winther; Hølledig, Malte Lindholm; Bach-Nielsen, Sebastian Siem

2017-01-01

An application for ballet training is presented that monitors the posture position (straightness of the spine and rotation of the pelvis) deviation from the ideal position in real-time. The human skeletal data is acquired through a Microsoft Kinect v2. The movement of the student is mirrored......-coded. In an experiment with 9-12 year-old dance students from a ballet school, comparing the audio-visual feedback modality with no feedback leads to an increase in posture accuracy (p

Class-D audio amplifiers with negative feedback

OpenAIRE

Cox, Stephen M.; Candy, B. H.

2006-01-01

There are many different designs for audio amplifiers. Class-D, or switching, amplifiers generate their output signal in the form of a high-frequency square wave of variable duty cycle (ratio of on time to off time). The square-wave nature of the output allows a particularly efficient output stage, with minimal losses. The output is ultimately filtered to remove components of the spectrum above the audio range. Mathematical models are derived here for a variety of related class-D amplifier de...

A second-order class-D audio amplifier

OpenAIRE

Cox, Stephen M.; Tan, M.T.; Yu, J.

2011-01-01

Class-D audio amplifiers are particularly efficient, and this efficiency has led to their ubiquity in a wide range of modern electronic appliances. Their output takes the form of a high-frequency square wave whose duty cycle (ratio of on-time to off-time) is modulated at low frequency according to the audio signal. A mathematical model is developed here for a second-order class-D amplifier design (i.e., containing one second-order integrator) with negative feedback. We derive exact expression...

Documentary management of the sport audio-visual information in the generalist televisions

OpenAIRE

Jorge Caldera Serrano; Felipe Alonso

2007-01-01

The management of the sport audio-visual documentation of the Information Systems of the state, zonal and local chains is analyzed within the framework. For it it is made makes a route by the documentary chain that makes the sport audio-visual information with the purpose of being analyzing each one of the parameters, showing therefore a series of recommendations and norms for the preparation of the sport audio-visual registry. Evidently the audio-visual sport documentation difference i...

Multi Carrier Modulation Audio Power Amplifier with Programmable Logic

DEFF Research Database (Denmark)

Christiansen, Theis; Andersen, Toke Meyer; Knott, Arnold

2009-01-01

While switch-mode audio power amplifiers allow compact implementations and high output power levels due to their high power efficiency, they are very well known for creating electromagnetic interference (EMI) with other electronic equipment. To lower the EMI of switch-mode (class D) audio power a...

A Psychoacoustic-Based Multiple Audio Object Coding Approach via Intra-Object Sparsity

Directory of Open Access Journals (Sweden)

Maoshen Jia

2017-12-01

Full Text Available Rendering spatial sound scenes via audio objects has become popular in recent years, since it can provide more flexibility for different auditory scenarios, such as 3D movies, spatial audio communication and virtual classrooms. To facilitate high-quality bitrate-efficient distribution for spatial audio objects, an encoding scheme based on intra-object sparsity (approximate k-sparsity of the audio object itself is proposed in this paper. The statistical analysis is presented to validate the notion that the audio object has a stronger sparseness in the Modified Discrete Cosine Transform (MDCT domain than in the Short Time Fourier Transform (STFT domain. By exploiting intra-object sparsity in the MDCT domain, multiple simultaneously occurring audio objects are compressed into a mono downmix signal with side information. To ensure a balanced perception quality of audio objects, a Psychoacoustic-based time-frequency instants sorting algorithm and an energy equalized Number of Preserved Time-Frequency Bins (NPTF allocation strategy are proposed, which are employed in the underlying compression framework. The downmix signal can be further encoded via Scalar Quantized Vector Huffman Coding (SQVH technique at a desirable bitrate, and the side information is transmitted in a lossless manner. Both objective and subjective evaluations show that the proposed encoding scheme outperforms the Sparsity Analysis (SPA approach and Spatial Audio Object Coding (SAOC in cases where eight objects were jointly encoded.

Implementation and Analysis of Real-Time Streaming Protocols.

Science.gov (United States)

Santos-González, Iván; Rivero-García, Alexandra; Molina-Gil, Jezabel; Caballero-Gil, Pino

2017-04-12

Communication media have become the primary way of interaction thanks to the discovery and innovation of many new technologies. One of the most widely used communication systems today is video streaming, which is constantly evolving. Such communications are a good alternative to face-to-face meetings, and are therefore very useful for coping with many problems caused by distance. However, they suffer from different issues such as bandwidth limitation, network congestion, energy efficiency, cost, reliability and connectivity. Hence, the quality of service and the quality of experience are considered the two most important issues for this type of communication. This work presents a complete comparative study of two of the most used protocols of video streaming, Real Time Streaming Protocol (RTSP) and the Web Real-Time Communication (WebRTC). In addition, this paper proposes two new mobile applications that implement those protocols in Android whose objective is to know how they are influenced by the aspects that most affect the streaming quality of service, which are the connection establishment time and the stream reception time. The new video streaming applications are also compared with the most popular video streaming applications for Android, and the experimental results of the analysis show that the developed WebRTC implementation improves the performance of the most popular video streaming applications with respect to the stream packet delay.

BAT: An open-source, web-based audio events annotation tool

OpenAIRE

Blai Meléndez-Catalan, Emilio Molina, Emilia Gómez

2017-01-01

In this paper we present BAT (BMAT Annotation Tool), an open-source, web-based tool for the manual annotation of events in audio recordings developed at BMAT (Barcelona Music and Audio Technologies). The main feature of the tool is that it provides an easy way to annotate the salience of simultaneous sound sources. Additionally, it allows to define multiple ontologies to adapt to multiple tasks and offers the possibility to cross-annotate audio data. Moreover, it is easy to install and deploy...

On the Use of Memory Models in Audio Features

DEFF Research Database (Denmark)

Jensen, Karl Kristoffer

2011-01-01

Audio feature estimation is potentially improved by including higher- level models. One such model is the Short Term Memory (STM) model. A new paradigm of audio feature estimation is obtained by adding the influence of notes in the STM. These notes are identified when the perceptual spectral flux...

Parallel field line and stream line tracing algorithms for space physics applications

Science.gov (United States)

Toth, G.; de Zeeuw, D.; Monostori, G.

2004-05-01

Field line and stream line tracing is required in various space physics applications, such as the coupling of the global magnetosphere and inner magnetosphere models, the coupling of the solar energetic particle and heliosphere models, or the modeling of comets, where the multispecies chemical equations are solved along stream lines of a steady state solution obtained with single fluid MHD model. Tracing a vector field is an inherently serial process, which is difficult to parallelize. This is especially true when the data corresponding to the vector field is distributed over a large number of processors. We designed algorithms for the various applications, which scale well to a large number of processors. In the first algorithm the computational domain is divided into blocks. Each block is on a single processor. The algorithm folows the vector field inside the blocks, and calculates a mapping of the block surfaces. The blocks communicate the values at the coinciding surfaces, and the results are interpolated. Finally all block surfaces are defined and values inside the blocks are obtained. In the second algorithm all processors start integrating along the vector field inside the accessible volume. When the field line leaves the local subdomain, the position and other information is stored in a buffer. Periodically the processors exchange the buffers, and continue integration of the field lines until they reach a boundary. At that point the results are sent back to the originating processor. Efficiency is achieved by a careful phasing of computation and communication. In the third algorithm the results of a steady state simulation are stored on a hard drive. The vector field is contained in blocks. All processors read in all the grid and vector field data and the stream lines are integrated in parallel. If a stream line enters a block, which has already been integrated, the results can be interpolated. By a clever ordering of the blocks the execution speed can be

Automatic Organisation and Quality Analysis of User-Generated Content with Audio Fingerprinting

OpenAIRE

Cavaco, Sofia; Magalhaes, Joao; Mordido, Gonçalo

2018-01-01

The increase of the quantity of user-generated content experienced in social media has boosted the importance of analysing and organising the content by its quality. Here, we propose a method that uses audio fingerprinting to organise and infer the quality of user-generated audio content. The proposed method detects the overlapping segments between different audio clips to organise and cluster the data according to events, and to infer the audio quality of the samples. A test setup with conce...

StreamExplorer: A Multi-Stage System for Visually Exploring Events in Social Streams.

Science.gov (United States)

Wu, Yingcai; Chen, Zhutian; Sun, Guodao; Xie, Xiao; Cao, Nan; Liu, Shixia; Cui, Weiwei

2017-10-18

Analyzing social streams is important for many applications, such as crisis management. However, the considerable diversity, increasing volume, and high dynamics of social streams of large events continue to be significant challenges that must be overcome to ensure effective exploration. We propose a novel framework by which to handle complex social streams on a budget PC. This framework features two components: 1) an online method to detect important time periods (i.e., subevents), and 2) a tailored GPU-assisted Self-Organizing Map (SOM) method, which clusters the tweets of subevents stably and efficiently. Based on the framework, we present StreamExplorer to facilitate the visual analysis, tracking, and comparison of a social stream at three levels. At a macroscopic level, StreamExplorer uses a new glyph-based timeline visualization, which presents a quick multi-faceted overview of the ebb and flow of a social stream. At a mesoscopic level, a map visualization is employed to visually summarize the social stream from either a topical or geographical aspect. At a microscopic level, users can employ interactive lenses to visually examine and explore the social stream from different perspectives. Two case studies and a task-based evaluation are used to demonstrate the effectiveness and usefulness of StreamExplorer.Analyzing social streams is important for many applications, such as crisis management. However, the considerable diversity, increasing volume, and high dynamics of social streams of large events continue to be significant challenges that must be overcome to ensure effective exploration. We propose a novel framework by which to handle complex social streams on a budget PC. This framework features two components: 1) an online method to detect important time periods (i.e., subevents), and 2) a tailored GPU-assisted Self-Organizing Map (SOM) method, which clusters the tweets of subevents stably and efficiently. Based on the framework, we present Stream

Task-FIFO co-scheduling of streaming applications on MPSoCs with predictable memory hierarchy

NARCIS (Netherlands)

Tang, Q.; Basten, A.A.; Geilen, M.C.W.; Stuijk, S.; Wei, Ji-Bo

This article studies the scheduling of real-time streaming applications on multiprocessor systems-on-chips with predictable memory hierarchy. An iteration-based task-FIFO co-scheduling framework is proposed for this problem. We obtain FIFO size distributions using Pareto space searching, based on

Task-FIFO co-scheduling of streaming applications on MPSoCs with predictable memory hierarchy

NARCIS (Netherlands)

Tang, Q.; Basten, T.; Geilen, M.; Stuijk, S.; Wei, J.B.

2017-01-01

This article studies the scheduling of real-time streaming applications on multiprocessor systems-on-chips with predictable memory hierarchy. An iteration-based task-FIFO co-scheduling framework is proposed for this problem. We obtain FIFO size distributions using Pareto space searching, based on

Haptic and Audio-visual Stimuli: Enhancing Experiences and Interaction

NARCIS (Netherlands)

Nijholt, Antinus; Dijk, Esko O.; Lemmens, Paul M.C.; Luitjens, S.B.

2010-01-01

The intention of the symposium on Haptic and Audio-visual stimuli at the EuroHaptics 2010 conference is to deepen the understanding of the effect of combined Haptic and Audio-visual stimuli. The knowledge gained will be used to enhance experiences and interactions in daily life. To this end, a

The Effect of Audio and Animation in Multimedia Instruction

Science.gov (United States)

Koroghlanian, Carol; Klein, James D.

2004-01-01

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.…

Selected Audio-Visual Materials for Consumer Education. [New Version.

Science.gov (United States)

Johnston, William L.

Ninety-two films, filmstrips, multi-media kits, slides, and audio cassettes, produced between 1964 and 1974, are listed in this selective annotated bibliography on consumer education. The major portion of the bibliography is devoted to films and filmstrips. The main topics of the audio-visual materials include purchasing, advertising, money…

Audio-visual temporal recalibration can be constrained by content cues regardless of spatial overlap

Directory of Open Access Journals (Sweden)

Warrick eRoseboom

2013-04-01

Full Text Available It has now been well established that the point of subjective synchrony for audio and visual events can be shifted following exposure to asynchronous audio-visual presentations, an effect often referred to as temporal recalibration. Recently it was further demonstrated that it is possible to concurrently maintain two such recalibrated, and opposing, estimates of audio-visual temporal synchrony. However, it remains unclear precisely what defines a given audio-visual pair such that it is possible to maintain a temporal relationship distinct from other pairs. It has been suggested that spatial separation of the different audio-visual pairs is necessary to achieve multiple distinct audio-visual synchrony estimates. Here we investigated if this was necessarily true. Specifically, we examined whether it is possible to obtain two distinct temporal recalibrations for stimuli that differed only in featural content. Using both complex (audio visual speech; Experiment 1 and simple stimuli (high and low pitch audio matched with either vertically or horizontally oriented Gabors; Experiment 2 we found concurrent, and opposite, recalibrations despite there being no spatial difference in presentation location at any point throughout the experiment. This result supports the notion that the content of an audio-visual pair can be used to constrain distinct audio-visual synchrony estimates regardless of spatial overlap.

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

International Nuclear Information System (INIS)

Nakamura, Mitsuhiro; Narita, Yuichiro; Matsuo, Yukinori; Narabayashi, Masaru; Nakata, Manabu; Sawada, Akira; Mizowaki, Takashi; Nagata, Yasushi; Hiraoka, Masahiro

2009-01-01

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.

Four-quadrant flyback converter for direct audio power amplification

OpenAIRE

Ljusev, Petar; Andersen, Michael Andreas E.

2005-01-01

This paper presents a bidirectional, four-quadrant flyback converter for use in direct audio power amplification. When compared to the standard Class-D switching audio power amplifier with a separate power supply, the proposed four-quadrant flyback converter provides simple solution with better efficiency, higher level of integration and lower component count.

Data Centric Sensor Stream Reduction for Real-Time Applications in Wireless Sensor Networks

Science.gov (United States)

Aquino, Andre Luiz Lins; Nakamura, Eduardo Freire

2009-01-01

This work presents a data-centric strategy to meet deadlines in soft real-time applications in wireless sensor networks. This strategy considers three main aspects: (i) The design of real-time application to obtain the minimum deadlines; (ii) An analytic model to estimate the ideal sample size used by data-reduction algorithms; and (iii) Two data-centric stream-based sampling algorithms to perform data reduction whenever necessary. Simulation results show that our data-centric strategies meet deadlines without loosing data representativeness. PMID:22303145

Selective attention modulates the direction of audio-visual temporal recalibration.

Science.gov (United States)

Ikumi, Nara; Soto-Faraco, Salvador

2014-01-01

Temporal recalibration of cross-modal synchrony has been proposed as a mechanism to compensate for timing differences between sensory modalities. However, far from the rich complexity of everyday life sensory environments, most studies to date have examined recalibration on isolated cross-modal pairings. Here, we hypothesize that selective attention might provide an effective filter to help resolve which stimuli are selected when multiple events compete for recalibration. We addressed this question by testing audio-visual recalibration following an adaptation phase where two opposing audio-visual asynchronies were present. The direction of voluntary visual attention, and therefore to one of the two possible asynchronies (flash leading or flash lagging), was manipulated using colour as a selection criterion. We found a shift in the point of subjective audio-visual simultaneity as a function of whether the observer had focused attention to audio-then-flash or to flash-then-audio groupings during the adaptation phase. A baseline adaptation condition revealed that this effect of endogenous attention was only effective toward the lagging flash. This hints at the role of exogenous capture and/or additional endogenous effects producing an asymmetry toward the leading flash. We conclude that selective attention helps promote selected audio-visual pairings to be combined and subsequently adjusted in time but, stimulus organization exerts a strong impact on recalibration. We tentatively hypothesize that the resolution of recalibration in complex scenarios involves the orchestration of top-down selection mechanisms and stimulus-driven processes.

Selective attention modulates the direction of audio-visual temporal recalibration.

Directory of Open Access Journals (Sweden)

Nara Ikumi

Full Text Available Temporal recalibration of cross-modal synchrony has been proposed as a mechanism to compensate for timing differences between sensory modalities. However, far from the rich complexity of everyday life sensory environments, most studies to date have examined recalibration on isolated cross-modal pairings. Here, we hypothesize that selective attention might provide an effective filter to help resolve which stimuli are selected when multiple events compete for recalibration. We addressed this question by testing audio-visual recalibration following an adaptation phase where two opposing audio-visual asynchronies were present. The direction of voluntary visual attention, and therefore to one of the two possible asynchronies (flash leading or flash lagging, was manipulated using colour as a selection criterion. We found a shift in the point of subjective audio-visual simultaneity as a function of whether the observer had focused attention to audio-then-flash or to flash-then-audio groupings during the adaptation phase. A baseline adaptation condition revealed that this effect of endogenous attention was only effective toward the lagging flash. This hints at the role of exogenous capture and/or additional endogenous effects producing an asymmetry toward the leading flash. We conclude that selective attention helps promote selected audio-visual pairings to be combined and subsequently adjusted in time but, stimulus organization exerts a strong impact on recalibration. We tentatively hypothesize that the resolution of recalibration in complex scenarios involves the orchestration of top-down selection mechanisms and stimulus-driven processes.

Semantic Analysis of Multimedial Information Usign Both Audio and Visual Clues

Directory of Open Access Journals (Sweden)

Andrej Lukac

2008-01-01

Full Text Available Nowadays, there is a lot of information in databases (text, audio/video form, etc.. It is important to be able to describe this data for better orientation in them. It is necessary to apply audio/video properties, which are used for metadata management, segmenting the document into semantically meaningful units, classifying each unit into a predefined scene type, indexing, summarizing the document for efficient retrieval and browsing. Data can be used for system that automatically searches for a specific person in a sequence also for special video sequences. Audio/video properties are presented by descriptors and description schemes. There are many features that can be used to characterize multimedial signals. We can analyze audio and video sequences jointly or considered them completely separately. Our aim is oriented to possibilities of combining multimedial features. Focus is direct into discussion programs, because there are more decisions how to combine audio features with video sequences.

Efficient Buffer Capacity and Scheduler Setting Computation for Soft Real-Time Stream Processing Applications

NARCIS (Netherlands)

Bekooij, Marco; Bekooij, Marco Jan Gerrit; Wiggers, M.H.; van Meerbergen, Jef

2007-01-01

Soft real-time applications that process data streams can often be intuitively described as dataflow process networks. In this paper we present a novel analysis technique to compute conservative estimates of the required buffer capacities in such process networks. With the same analysis technique

Self-oscillating modulators for direct energy conversion audio power amplifiers

Energy Technology Data Exchange (ETDEWEB)

Ljusev, P.; Andersen, Michael A.E.

2005-07-01

Direct energy conversion audio power amplifier represents total integration of switching-mode power supply and Class D audio power amplifier into one compact stage, achieving high efficiency, high level of integration, low component count and eventually low cost. This paper presents how self-oscillating modulators can be used with the direct switching-mode audio power amplifier to improve its performance by providing fast hysteretic control with high power supply rejection ratio, open-loop stability and high bandwidth. Its operation is thoroughly analyzed and simulated waveforms of a prototype amplifier are presented. (au)

Rehabilitation of balance-impaired stroke patients through audio-visual biofeedback

DEFF Research Database (Denmark)

Gheorghe, Cristina; Nissen, Thomas; Juul Rosengreen Christensen, Daniel

2015-01-01

This study explored how audio-visual biofeedback influences physical balance of seven balance-impaired stroke patients, between 33–70 years-of-age. The setup included a bespoke balance board and a music rhythm game. The procedure was designed as follows: (1) a control group who performed a balance...... training exercise without any technological input, (2) a visual biofeedback group, performing via visual input, and (3) an audio-visual biofeedback group, performing via audio and visual input. Results retrieved from comparisons between the data sets (2) and (3) suggested superior postural stability...

Industrial-Strength Streaming Video.

Science.gov (United States)

Avgerakis, George; Waring, Becky

1997-01-01

Corporate training, financial services, entertainment, and education are among the top applications for streaming video servers, which send video to the desktop without downloading the whole file to the hard disk, saving time and eliminating copyrights questions. Examines streaming video technology, lists ten tips for better net video, and ranks…

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

Science.gov (United States)

2010-10-01

... 47 Telecommunication 2 2010-10-01 2010-10-01 false Licensing provisions for the 2.3 GHz satellite digital audio radio service. 25.144 Section 25.144 Telecommunication FEDERAL COMMUNICATIONS COMMISSION (CONTINUED) COMMON CARRIER SERVICES SATELLITE COMMUNICATIONS Applications and Licenses Space Stations § 25...

Balancing Audio

DEFF Research Database (Denmark)

Walther-Hansen, Mads

2016-01-01

is not thoroughly understood. In this paper I treat balance as a metaphor that we use to reason about several different actions in music production, such as adjusting levels, editing the frequency spectrum or the spatiality of the recording. This study is based on an exploration of a linguistic corpus of sound......This paper explores the concept of balance in music production and examines the role of conceptual metaphors in reasoning about audio editing. Balance may be the most central concept in record production, however, the way we cognitively understand and respond meaningfully to a mix requiring balance...

Audio computer-assisted self interview compared to traditional interview in an HIV-related behavioral survey in Vietnam.

Science.gov (United States)

Le, Linh Cu; Vu, Lan T H

2012-10-01

Globally, population surveys on HIV/AIDS and other sensitive topics have been using audio computer-assisted self interview for many years. This interview technique, however, is still new to Vietnam and little is known about its application and impact in general population surveys. One plausible hypothesis is that residents of Vietnam interviewed using this technique may provide a higher response rate and be more willing to reveal their true behaviors than if interviewed with traditional methods. This study aims to compare audio computer-assisted self interview with traditional face-to-face personal interview and self-administered interview with regard to rates of refusal and affirmative responses to questions on sensitive topics related to HIV/AIDS. In June 2010, a randomized study was conducted in three cities (Ha Noi, Da Nan and Can Tho), using a sample of 4049 residents aged 15 to 49 years. Respondents were randomly assigned to one of three interviewing methods: audio computer-assisted self interview, personal face-to-face interview, and self-administered paper interview. Instead of providing answers directly to interviewer questions as with traditional methods, audio computer-assisted self-interview respondents read the questions displayed on a laptop screen, while listening to the questions through audio headphones, then entered responses using a laptop keyboard. A MySQL database was used for data management and SPSS statistical package version 18 used for data analysis with bivariate and multivariate statistical techniques. Rates of high risk behaviors and mean values of continuous variables were compared for the three data collection methods. Audio computer-assisted self interview showed advantages over comparison techniques, achieving lower refusal rates and reporting higher prevalence of some sensitive and risk behaviors (perhaps indication of more truthful answers). Premarital sex was reported by 20.4% in the audio computer-assisted self-interview survey

Application of quasi-steady-state plasma streams for simulation of ITER transient heat loads

International Nuclear Information System (INIS)

Bandura, A.N.; Chebotarev, V.V.; Garkusha, I.E.; Makhlaj, V.A.; Marchenko, A.K.; Solyakov, D.G.; Tereshin, V.I.; Trubchaninov, S.A.; Tsarenko, A.V.; Landman, I.

2004-01-01

The paper presents experimental investigations of energy characteristics of the plasma streams generated with quasi-steady-state plasma accelerator QSPA Kh-50 and adjustment of plasma parameters from the point of view its applicability for simulation of transient plasma heat loads expected for ITER disruptions and type I ELMs. Possibility of generation of high-power magnetized plasma streams with ion impact energy up to 0.6 keV, pulse length of 0.25 ms and heat loads varied in wide range from 0.5 to 30 MJ/m 2 has been demonstrated and some features of plasma interaction with tungsten targets in dependence on plasma heat loads are discussed. (author)

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

Science.gov (United States)

Koroghlanian, Carol May

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

Automated Speech and Audio Analysis for Semantic Access to Multimedia

NARCIS (Netherlands)

Jong, F.M.G. de; Ordelman, R.; Huijbregts, M.

2006-01-01

The deployment and integration of audio processing tools can enhance the semantic annotation of multimedia content, and as a consequence, improve the effectiveness of conceptual access tools. This paper overviews the various ways in which automatic speech and audio analysis can contribute to

Automated speech and audio analysis for semantic access to multimedia

NARCIS (Netherlands)

de Jong, Franciska M.G.; Ordelman, Roeland J.F.; Huijbregts, M.A.H.; Avrithis, Y.; Kompatsiaris, Y.; Staab, S.; O' Connor, N.E.

2006-01-01

The deployment and integration of audio processing tools can enhance the semantic annotation of multimedia content, and as a consequence, improve the effectiveness of conceptual access tools. This paper overviews the various ways in which automatic speech and audio analysis can contribute to

Audio Feedback -- Better Feedback?

Science.gov (United States)

Voelkel, Susanne; Mello, Luciane V.

2014-01-01

National Student Survey (NSS) results show that many students are dissatisfied with the amount and quality of feedback they get for their work. This study reports on two case studies in which we tried to address these issues by introducing audio feedback to one undergraduate (UG) and one postgraduate (PG) class, respectively. In case study one…

Emotion-based Music Rretrieval on a Well-reduced Audio Feature Space

DEFF Research Database (Denmark)

Ruxanda, Maria Magdalena; Chua, Bee Yong; Nanopoulos, Alexandros

2009-01-01

-emotion. However, the real-time systems that retrieve music over large music databases, can achieve order of magnitude performance increase, if applying multidimensional indexing over a dimensionally reduced audio feature space. To meet this performance achievement, in this paper, extensive studies are conducted......Music expresses emotion. A number of audio extracted features have influence on the perceived emotional expression of music. These audio features generate a high-dimensional space, on which music similarity retrieval can be performed effectively, with respect to human perception of the music...... on a number of dimensionality reduction algorithms, including both classic and novel approaches. The paper clearly envisages which dimensionality reduction techniques on the considered audio feature space, can preserve in average the accuracy of the emotion-based music retrieval....

News video story segmentation method using fusion of audio-visual features

Science.gov (United States)

Wen, Jun; Wu, Ling-da; Zeng, Pu; Luan, Xi-dao; Xie, Yu-xiang

2007-11-01

News story segmentation is an important aspect for news video analysis. This paper presents a method for news video story segmentation. Different form prior works, which base on visual features transform, the proposed technique uses audio features as baseline and fuses visual features with it to refine the results. At first, it selects silence clips as audio features candidate points, and selects shot boundaries and anchor shots as two kinds of visual features candidate points. Then this paper selects audio feature candidates as cues and develops different fusion method, which effectively using diverse type visual candidates to refine audio candidates, to get story boundaries. Experiment results show that this method has high efficiency and adaptability to different kinds of news video.

Improving audio chord transcription by exploiting harmonic and metric knowledge

NARCIS (Netherlands)

de Haas, W.B.; Rodrigues Magalhães, J.P.; Wiering, F.

2012-01-01

We present a new system for chord transcription from polyphonic musical audio that uses domain-specific knowledge about tonal harmony and metrical position to improve chord transcription performance. Low-level pulse and spectral features are extracted from an audio source using the Vamp plugin

Self-oscillating modulators for direct energy conversion audio power amplifiers

DEFF Research Database (Denmark)

Ljusev, Petar; Andersen, Michael Andreas E.

2005-01-01

Direct energy conversion audio power amplifier represents total integration of switching-mode power supply and Class D audio power amplifier into one compact stage, achieving high efficiency, high level of integration, low component count and eventually low cost. This paper presents how self-oscillating...

Extraction, Mapping, and Evaluation of Expressive Acoustic Features for Adaptive Digital Audio Effects

DEFF Research Database (Denmark)

Holfelt, Jonas; Csapo, Gergely; Andersson, Nikolaj Schwab

2017-01-01

This paper describes the design and implementation of a real-time adaptive digital audio effect with an emphasis on using expressive audio features that control effect param- eters. Research in adaptive digital audio effects is cov- ered along with studies about expressivity and important...

Let Their Voices Be Heard! Building a Multicultural Audio Collection.

Science.gov (United States)

Tucker, Judith Cook

1992-01-01

Discusses building a multicultural audio collection for a library. Gives some guidelines about selecting materials that really represent different cultures. Audio materials that are considered fall roughly into the categories of children's stories, didactic materials, oral histories, poetry and folktales, and music. The goal is an authentic…

Sampling the stream landscape: Improving the applicability of an ecoregion-level capture probability model for stream fishes

Science.gov (United States)

Mollenhauer, Robert; Mouser, Joshua B.; Brewer, Shannon K.

2018-01-01

Temporal and spatial variability in streams result in heterogeneous gear capture probability (i.e., the proportion of available individuals identified) that confounds interpretation of data used to monitor fish abundance. We modeled tow-barge electrofishing capture probability at multiple spatial scales for nine Ozark Highland stream fishes. In addition to fish size, we identified seven reach-scale environmental characteristics associated with variable capture probability: stream discharge, water depth, conductivity, water clarity, emergent vegetation, wetted width–depth ratio, and proportion of riffle habitat. The magnitude of the relationship between capture probability and both discharge and depth varied among stream fishes. We also identified lithological characteristics among stream segments as a coarse-scale source of variable capture probability. The resulting capture probability model can be used to adjust catch data and derive reach-scale absolute abundance estimates across a wide range of sampling conditions with similar effort as used in more traditional fisheries surveys (i.e., catch per unit effort). Adjusting catch data based on variable capture probability improves the comparability of data sets, thus promoting both well-informed conservation and management decisions and advances in stream-fish ecology.

Exploiting the Power of Relational Databases for Efficient Stream Processing

NARCIS (Netherlands)

E. Liarou (Erietta); R.A. Goncalves (Romulo); S. Idreos (Stratos)

2009-01-01

textabstractStream applications gained significant popularity over the last years that lead to the development of specialized stream engines. These systems are designed from scratch with a different philosophy than nowadays database engines in order to cope with the stream applications

Round-Robin Streaming with Generations

DEFF Research Database (Denmark)

Li, Yao; Vingelmann, Peter; Pedersen, Morten Videbæk

2012-01-01

We consider three types of application layer coding for streaming over lossy links: random linear coding, systematic random linear coding, and structured coding. The file being streamed is divided into sub-blocks (generations). Code symbols are formed by combining data belonging to the same...

The Use of Audio and Animation in Computer Based Instruction.

Science.gov (United States)

Koroghlanian, Carol; Klein, James D.

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…

Audio-Visual Speaker Diarization Based on Spatiotemporal Bayesian Fusion.

Science.gov (United States)

Gebru, Israel D; Ba, Sileye; Li, Xiaofei; Horaud, Radu

2018-05-01

Speaker diarization consists of assigning speech signals to people engaged in a dialogue. An audio-visual spatiotemporal diarization model is proposed. The model is well suited for challenging scenarios that consist of several participants engaged in multi-party interaction while they move around and turn their heads towards the other participants rather than facing the cameras and the microphones. Multiple-person visual tracking is combined with multiple speech-source localization in order to tackle the speech-to-person association problem. The latter is solved within a novel audio-visual fusion method on the following grounds: binaural spectral features are first extracted from a microphone pair, then a supervised audio-visual alignment technique maps these features onto an image, and finally a semi-supervised clustering method assigns binaural spectral features to visible persons. The main advantage of this method over previous work is that it processes in a principled way speech signals uttered simultaneously by multiple persons. The diarization itself is cast into a latent-variable temporal graphical model that infers speaker identities and speech turns, based on the output of an audio-visual association process, executed at each time slice, and on the dynamics of the diarization variable itself. The proposed formulation yields an efficient exact inference procedure. A novel dataset, that contains audio-visual training data as well as a number of scenarios involving several participants engaged in formal and informal dialogue, is introduced. The proposed method is thoroughly tested and benchmarked with respect to several state-of-the art diarization algorithms.

El Digital Audio Tape Recorder. Contra autores y creadores

Directory of Open Access Journals (Sweden)

Jun Ono

2015-01-01

Full Text Available La llamada "DAT" (abreviatura por "digital audio tape recorder" / grabadora digital de audio ha recibido cobertura durante mucho tiempo en los medios masivos de Japón y otros países, como un producto acústico electrónico nuevo y controversial de la industria japonesa de artefactos electrónicos. ¿Qué ha pasado con el objeto de esta controversia?

Haptic and Audio Interaction Design

DEFF Research Database (Denmark)

This book constitutes the refereed proceedings of the 5th International Workshop on Haptic and Audio Interaction Design, HAID 2010 held in Copenhagen, Denmark, in September 2010. The 21 revised full papers presented were carefully reviewed and selected for inclusion in the book. The papers are or...

Periodic-drop-take calculus for stream transformers

NARCIS (Netherlands)

Mak, R.H.

2005-01-01

Stream transformers are a formalism to specify and reason about stream processing systems. Many application specific circuits, e.g. in the area of signal processing, classify as such systems. This paper presents a two- operator calculus to reason about a specific class of stream operators, viz. the

Application of regression model on stream water quality parameters

International Nuclear Information System (INIS)

Suleman, M.; Maqbool, F.; Malik, A.H.; Bhatti, Z.A.

2012-01-01

Statistical analysis was conducted to evaluate the effect of solid waste leachate from the open solid waste dumping site of Salhad on the stream water quality. Five sites were selected along the stream. Two sites were selected prior to mixing of leachate with the surface water. One was of leachate and other two sites were affected with leachate. Samples were analyzed for pH, water temperature, electrical conductivity (EC), total dissolved solids (TDS), Biological oxygen demand (BOD), chemical oxygen demand (COD), dissolved oxygen (DO) and total bacterial load (TBL). In this study correlation coefficient r among different water quality parameters of various sites were calculated by using Pearson model and then average of each correlation between two parameters were also calculated, which shows TDS and EC and pH and BOD have significantly increasing r value, while temperature and TDS, temp and EC, DO and BL, DO and COD have decreasing r value. Single factor ANOVA at 5% level of significance was used which shows EC, TDS, TCL and COD were significantly differ among various sites. By the application of these two statistical approaches TDS and EC shows strongly positive correlation because the ions from the dissolved solids in water influence the ability of that water to conduct an electrical current. These two parameters significantly vary among 5 sites which are further confirmed by using linear regression. (author)

State Waste Discharge Permit application for industrial discharge to land: 200 East Area W-252 streams

International Nuclear Information System (INIS)

1993-12-01

This document constitutes the WAC 173-216 State Waste Discharge Permit application for six W-252 liquid effluent streams at the Hanford Site. Appendices B through H correspond to Section B through H in the permit application form. Within each appendix, sections correspond directly to the respective questions on the application form. The appendices include: Product or service information; Plant operational characteristics; Water consumption and waterloss; Wastewater information; Stormwater; Other information; and Site assessment

Responding Effectively to Composition Students: Comparing Student Perceptions of Written and Audio Feedback

Science.gov (United States)

Bilbro, J.; Iluzada, C.; Clark, D. E.

2013-01-01

The authors compared student perceptions of audio and written feedback in order to assess what types of students may benefit from receiving audio feedback on their essays rather than written feedback. Many instructors previously have reported the advantages they see in audio feedback, but little quantitative research has been done on how the…

Towards Unification of Methods for Speech, Audio, Picture and Multimedia Quality Assessment

DEFF Research Database (Denmark)

Zielinski, S.; Rumsey, F.; Bech, Søren

2015-01-01

attempting to “bridge the gap” between the quality assessment methods used in various disciplines are indicated. Prospective challenges faced by researchers in the unification process are outlined. They include development of unified scales, defining unified anchors, integration of objective models......The paper addresses the need to develop unified methods for subjective and objective quality assessment across speech, audio, picture, and multimedia applications. Commonalities and differences between the currently used standards are overviewed. Examples of the already undertaken research...

A conceptual framework for audio-visual museum media

DEFF Research Database (Denmark)

Kirkedahl Lysholm Nielsen, Mikkel

2017-01-01

In today's history museums, the past is communicated through many other means than original artefacts. This interdisciplinary and theoretical article suggests a new approach to studying the use of audio-visual media, such as film, video and related media types, in a museum context. The centre...... and museum studies, existing case studies, and real life observations, the suggested framework instead stress particular characteristics of contextual use of audio-visual media in history museums, such as authenticity, virtuality, interativity, social context and spatial attributes of the communication...

Nonlinear dynamic macromodeling techniques for audio systems

Science.gov (United States)

Ogrodzki, Jan; Bieńkowski, Piotr

2015-09-01

This paper develops a modelling method and a models identification technique for the nonlinear dynamic audio systems. Identification is performed by means of a behavioral approach based on a polynomial approximation. This approach makes use of Discrete Fourier Transform and Harmonic Balance Method. A model of an audio system is first created and identified and then it is simulated in real time using an algorithm of low computational complexity. The algorithm consists in real time emulation of the system response rather than in simulation of the system itself. The proposed software is written in Python language using object oriented programming techniques. The code is optimized for a multithreads environment.

Engaging Students with Audio Feedback

Science.gov (United States)

Cann, Alan

2014-01-01

Students express widespread dissatisfaction with academic feedback. Teaching staff perceive a frequent lack of student engagement with written feedback, much of which goes uncollected or unread. Published evidence shows that audio feedback is highly acceptable to students but is underused. This paper explores methods to produce and deliver audio…

Audio-Visual Temporal Recalibration Can be Constrained by Content Cues Regardless of Spatial Overlap

OpenAIRE

Roseboom, Warrick; Kawabe, Takahiro; Nishida, Shin?Ya

2013-01-01

It has now been well established that the point of subjective synchrony for audio and visual events can be shifted following exposure to asynchronous audio-visual presentations, an effect often referred to as temporal recalibration. Recently it was further demonstrated that it is possible to concurrently maintain two such recalibrated, and opposing, estimates of audio-visual temporal synchrony. However, it remains unclear precisely what defines a given audio-visual pair such that it is possib...

Conflicting audio-haptic feedback in physically based simulation of walking sounds

DEFF Research Database (Denmark)

Turchet, Luca; Serafin, Stefania; Dimitrov, Smilen

2010-01-01

We describe an audio-haptic experiment conducted using a system which simulates in real-time the auditory and haptic sensation of walking on different surfaces. The system is based on physical models, that drive both the haptic and audio synthesizers, and a pair of shoes enhanced with sensors...... and actuators. Such experiment was run to examine the ability of subjects to recognize the different surfaces with both coherent and incoherent audio-haptic stimuli. Results show that in this kind of tasks the auditory modality is dominant on the haptic one....

Investigation of Energy Consumption and Sound Quality for Class-D Audio Amplifiers using Tracking Power Supplies

DEFF Research Database (Denmark)

Yamauchi, Akira; Schneider, Henrik; Knott, Arnold

2015-01-01

power supply tracking and its influence on power losses, audio performance and environmental impact for a 130 W class-D amplifier prototype as well as a commercialized class-D amplifier. Both modeled and experimental results verify that a large improvement of efficiency can be achieved. The total...... harmonic is found to be unaffected by stepless power supply tracking due the high supply rejection ratio of the used amplifiers under test.......The main advantage of Class-D audio amplifiers is high efficiency which is often stated to be more than 90 % but at idle or low power levels the efficiency is much lower. The waste energy is an environmental concern, a concern in mobile applications where long battery operation is required...

Audio-vocal interaction in single neurons of the monkey ventrolateral prefrontal cortex.

Science.gov (United States)

Hage, Steffen R; Nieder, Andreas

2015-05-06

Complex audio-vocal integration systems depend on a strong interconnection between the auditory and the vocal motor system. To gain cognitive control over audio-vocal interaction during vocal motor control, the PFC needs to be involved. Neurons in the ventrolateral PFC (VLPFC) have been shown to separately encode the sensory perceptions and motor production of vocalizations. It is unknown, however, whether single neurons in the PFC reflect audio-vocal interactions. We therefore recorded single-unit activity in the VLPFC of rhesus monkeys (Macaca mulatta) while they produced vocalizations on command or passively listened to monkey calls. We found that 12% of randomly selected neurons in VLPFC modulated their discharge rate in response to acoustic stimulation with species-specific calls. Almost three-fourths of these auditory neurons showed an additional modulation of their discharge rates either before and/or during the monkeys' motor production of vocalization. Based on these audio-vocal interactions, the VLPFC might be well positioned to combine higher order auditory processing with cognitive control of the vocal motor output. Such audio-vocal integration processes in the VLPFC might constitute a precursor for the evolution of complex learned audio-vocal integration systems, ultimately giving rise to human speech. Copyright © 2015 the authors 0270-6474/15/357030-11$15.00/0.

Effects of Hearing Protection Device Attenuation on Unmanned Aerial Vehicle (UAV) Audio Signatures

Science.gov (United States)

2016-03-01

UAV ) Audio Signatures by Melissa Bezandry, Adrienne Raglin, and John Noble Approved for public release; distribution...Research Laboratory Effects of Hearing Protection Device Attenuation on Unmanned Aerial Vehicle ( UAV ) Audio Signatures by Melissa Bezandry...Aerial Vehicle ( UAV ) Audio Signatures 5a. CONTRACT NUMBER 5b. GRANT NUMBER 5c. PROGRAM ELEMENT NUMBER 6. AUTHOR(S) Melissa Bezandry

Effect of road salt application on seasonal chloride concentrations and toxicity in south-central Indiana streams.

Science.gov (United States)

Gardner, Kristin M; Royer, Todd V

2010-01-01

Contemporary information on road salt runoff is needed for management of water resources in regions experiencing urbanization and increased road density. We investigated seasonal Cl(-) concentrations among five streams in south-central Indiana that drained watersheds varying in degree of urbanization and ranging in size from 9.3 to 27 km(2). We also conducted acute toxicity tests with Daphnia pulex to assess the potential effects of the observed Cl(-) concentrations on aquatic life. Periods of elevated Cl(-) concentrations were observed during the winters of 2007-08 and 2008-09 at all sites except the reference site. The highest Cl(-) concentration observed during the study was 2100 mg L(-1) and occurred at the most urbanized site. The Cl(-) concentration at the reference site never exceeded 22 mg L(-1). The application of road salt caused large increases in stream Cl(-) concentrations, but the elevated Cl(-) levels did not appear to be a significant threat to aquatic life based on our toxicity testing. Only the most urbanized site showed evidence of salt retention within the watershed, whereas the other sites exported the road salt relatively quickly after its application, suggesting storm drains and impervious surfaces minimized interaction between soils and salt-laden runoff. During winter at these sites, the response in stream Cl(-) concentrations appeared to be controlled by the timing and intensity of road salt application, the magnitude of precipitation, and the occurrence of air temperatures that caused snowmelt and generated runoff.

Task-oriented quality assessment and adaptation in real-time mission critical video streaming applications

Science.gov (United States)

Nightingale, James; Wang, Qi; Grecos, Christos

2015-02-01

In recent years video traffic has become the dominant application on the Internet with global year-on-year increases in video-oriented consumer services. Driven by improved bandwidth in both mobile and fixed networks, steadily reducing hardware costs and the development of new technologies, many existing and new classes of commercial and industrial video applications are now being upgraded or emerging. Some of the use cases for these applications include areas such as public and private security monitoring for loss prevention or intruder detection, industrial process monitoring and critical infrastructure monitoring. The use of video is becoming commonplace in defence, security, commercial, industrial, educational and health contexts. Towards optimal performances, the design or optimisation in each of these applications should be context aware and task oriented with the characteristics of the video stream (frame rate, spatial resolution, bandwidth etc.) chosen to match the use case requirements. For example, in the security domain, a task-oriented consideration may be that higher resolution video would be required to identify an intruder than to simply detect his presence. Whilst in the same case, contextual factors such as the requirement to transmit over a resource-limited wireless link, may impose constraints on the selection of optimum task-oriented parameters. This paper presents a novel, conceptually simple and easily implemented method of assessing video quality relative to its suitability for a particular task and dynamically adapting videos streams during transmission to ensure that the task can be successfully completed. Firstly we defined two principle classes of tasks: recognition tasks and event detection tasks. These task classes are further subdivided into a set of task-related profiles, each of which is associated with a set of taskoriented attributes (minimum spatial resolution, minimum frame rate etc.). For example, in the detection class

Audio-Visual Perception System for a Humanoid Robotic Head

Directory of Open Access Journals (Sweden)

Raquel Viciana-Abad

2014-05-01

Full Text Available 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.

Radioactive Decay: Audio Data Collection

Science.gov (United States)

Struthers, Allan

2009-01-01

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…

Music Identification System Using MPEG-7 Audio Signature Descriptors

Science.gov (United States)

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

2013-01-01

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. PMID:23533359

Music Identification System Using MPEG-7 Audio Signature Descriptors

Directory of Open Access Journals (Sweden)

Shingchern D. You

2013-01-01

Full Text Available 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.

Technical Evaluation Report 31: Internet Audio Products (3/ 3

Directory of Open Access Journals (Sweden)

Jim Rudolph

2004-08-01

Full Text Available Two contrasting additions to the online audio market are reviewed: iVocalize, a browser-based audio-conferencing software, and Skype, a PC-to-PC Internet telephone tool. These products are selected for review on the basis of their success in gaining rapid popular attention and usage during 2003-04. The iVocalize review emphasizes the product’s role in the development of a series of successful online audio communities – notably several serving visually impaired users. The Skype review stresses the ease with which the product may be used for simultaneous PC-to-PC communication among up to five users. Editor’s Note: This paper serves as an introduction to reports about online community building, and reviews of online products for disabled persons, in the next ten reports in this series. JPB, Series Ed.

Ferrite bead effect on Class-D amplifier audio quality

OpenAIRE

Haddad , Kevin El; Mrad , Roberto; Morel , Florent; Pillonnet , Gael; Vollaire , Christian; Nagari , Angelo

2014-01-01

International audience; This paper studies the effect of ferrite beads on the audio quality of Class-D audio amplifiers. This latter is a switch-ing circuit which creates high frequency harmonics. Generally, a filter is used at the amplifier output for the sake of electro-magnetic compatibility (EMC). So often, in integrated solutions, this filter contains ferrite beads which are magnetic components and present nonlinear behavior. Time domain measurements and their equivalence in frequency do...

Maximizing Resource Utilization in Video Streaming Systems

Science.gov (United States)

Alsmirat, Mohammad Abdullah

2013-01-01

Video streaming has recently grown dramatically in popularity over the Internet, Cable TV, and wire-less networks. Because of the resource demanding nature of video streaming applications, maximizing resource utilization in any video streaming system is a key factor to increase the scalability and decrease the cost of the system. Resources to…

Estimation of violin bowing features from Audio recordings with Convolutional Networks

DEFF Research Database (Denmark)

Perez-Carillo, Alfonso; Purwins, Hendrik

The acquisition of musical gestures and particularly of instrument controls from a musical performance is a field of increasing interest with applications in many research areas. In the last years, the development of novel sensing technologies has allowed the fine measurement of such controls...... and low-cost of the acquisition and its nonintrusive nature. The main challenge is designing robust detection algorithms to be as accurate as the direct approaches. In this paper, we present an indirect acquisition method to estimate violin bowing controls from audio signal analysis based on training...

Assessing the importance of audio/video synchronization for simultaneous translation of video sequences

OpenAIRE

Staelens, Nicolas; De Meulenaere, Jonas; Bleumers, Lizzy; Van Wallendael, Glenn; De Cock, Jan; Geeraert, Koen; Vercammen, Nick; Van den Broeck, Wendy; Vermeulen, Brecht; Van de Walle, Rik; Demeester, Piet

2012-01-01

Lip synchronization is considered a key parameter during interactive communication. In the case of video conferencing and television broadcasting, the differential delay between audio and video should remain below certain thresholds, as recommended by several standardization bodies. However, further research has also shown that these thresholds can be relaxed, depending on the targeted application and use case. In this article, we investigate the influence of lip sync on the ability to perfor...

Applications of on-stream analysis systems

International Nuclear Information System (INIS)

Howarth, W.J.

1982-01-01

Benefits of on-stream analysis include increased metal recovery, reduced consumption of reagents, increases in concentrate grade and savings in labour. A poorly designed analysis zone can give rise to incorrect assays as a result of segregation or excessive or variable aeration

Capabilities of Raspberry Pi 2 for Big Data and Video Streaming Applications in Data Centres

NARCIS (Netherlands)

Schot, Nick J.; Velthuis, Paul J.E.; Postema, Björn Frits; Remke, Anne Katharina Ingrid; Remke, A.K.I.; Haverkort, Boudewijn R.H.M.; Haverkort, B.R.H.M.

Many new data centres have been built in recent years in order to keep up with the rising demand for server capacity. These data centres require a lot of electrical energy and cooling. Big data and video streaming are two heavily used applications in data centres. This paper experimentally

A compact electroencephalogram recording device with integrated audio stimulation system

Science.gov (United States)

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

2010-06-01

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 μVrms 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.

Audio segmentation using Flattened Local Trimmed Range for ecological acoustic space analysis

Directory of Open Access Journals (Sweden)

Giovany Vega

2016-06-01

Full Text Available The acoustic space in a given environment is filled with footprints arising from three processes: biophony, geophony and anthrophony. Bioacoustic research using passive acoustic sensors can result in thousands of recordings. An important component of processing these recordings is to automate signal detection. In this paper, we describe a new spectrogram-based approach for extracting individual audio events. Spectrogram-based audio event detection (AED relies on separating the spectrogram into background (i.e., noise and foreground (i.e., signal classes using a threshold such as a global threshold, a per-band threshold, or one given by a classifier. These methods are either too sensitive to noise, designed for an individual species, or require prior training data. Our goal is to develop an algorithm that is not sensitive to noise, does not need any prior training data and works with any type of audio event. To do this, we propose: (1 a spectrogram filtering method, the Flattened Local Trimmed Range (FLTR method, which models the spectrogram as a mixture of stationary and non-stationary energy processes and mitigates the effect of the stationary processes, and (2 an unsupervised algorithm that uses the filter to detect audio events. We measured the performance of the algorithm using a set of six thoroughly validated audio recordings and obtained a sensitivity of 94% and a positive predictive value of 89%. These sensitivity and positive predictive values are very high, given that the validated recordings are diverse and obtained from field conditions. The algorithm was then used to extract audio events in three datasets. Features of these audio events were plotted and showed the unique aspects of the three acoustic communities.

Training of audio descriptors: the cinematographic aesthetics as basis for the learning of the audio description aesthetics – materials, methods and products

Directory of Open Access Journals (Sweden)

Soraya Ferreira Alves

2016-12-01

Full Text Available Audio description (AD, a resource used to make theater, cinema, TV, and visual works of art accessible to people with visual impairments, is slowly being implemented in Brazil and demanding qualified professionals. Based on this statement, this article reports the results of a research developed during post-doctoral studies. The study is dedicated to the confrontation of film aesthetics with audio description techniques to check how the knowledge of the former can contribute to audiodescritor training. Through action research, a short film adapted from a Mario de Andrade’s, a Brazilian writer, short story called O Peru de Natal (Christmas Turkey was produced. The film as well as its audio description were carried out involving students and teachers from the discipline Intersemiotic Translation at the State University of Ceará. Thus, we intended to suggest pedagogical procedures generated by the students experiences by evaluating their choices and their implications.

Multinomial N-mixture models improve the applicability of electrofishing for developing population estimates of stream-dwelling Smallmouth Bass

Science.gov (United States)

Mollenhauer, Robert; Brewer, Shannon K.

2017-01-01

Failure to account for variable detection across survey conditions constrains progressive stream ecology and can lead to erroneous stream fish management and conservation decisions. In addition to variable detection’s confounding long-term stream fish population trends, reliable abundance estimates across a wide range of survey conditions are fundamental to establishing species–environment relationships. Despite major advancements in accounting for variable detection when surveying animal populations, these approaches remain largely ignored by stream fish scientists, and CPUE remains the most common metric used by researchers and managers. One notable advancement for addressing the challenges of variable detection is the multinomial N-mixture model. Multinomial N-mixture models use a flexible hierarchical framework to model the detection process across sites as a function of covariates; they also accommodate common fisheries survey methods, such as removal and capture–recapture. Effective monitoring of stream-dwelling Smallmouth Bass Micropterus dolomieu populations has long been challenging; therefore, our objective was to examine the use of multinomial N-mixture models to improve the applicability of electrofishing for estimating absolute abundance. We sampled Smallmouth Bass populations by using tow-barge electrofishing across a range of environmental conditions in streams of the Ozark Highlands ecoregion. Using an information-theoretic approach, we identified effort, water clarity, wetted channel width, and water depth as covariates that were related to variable Smallmouth Bass electrofishing detection. Smallmouth Bass abundance estimates derived from our top model consistently agreed with baseline estimates obtained via snorkel surveys. Additionally, confidence intervals from the multinomial N-mixture models were consistently more precise than those of unbiased Petersen capture–recapture estimates due to the dependency among data sets in the

Direct-conversion switching-mode audio power amplifier with active capacitive voltage clamp

Energy Technology Data Exchange (ETDEWEB)

Ljusev, P.; Andersen, Michael A.E.

2005-07-01

This paper discusses the advantages and problems when implementing direct energy conversion switching-mode audio power amplifiers. It is shown that the total integration of the power supply and Class D audio power amplifier into one compact direct converter can simplify design, increase efficiency and integration level, reduce product volume and lower its cost. As an example, the principle of operation and the measurements made on a direct-conversion switching-mode audio power amplifier with active capacitive voltage clamp are presented. (au)

Direct-conversion switching-mode audio power amplifier with active capacitive voltage clamp

DEFF Research Database (Denmark)

Ljusev, Petar; Andersen, Michael Andreas E.

2005-01-01

This paper discusses the advantages and problems when implementing direct energy conversion switching-mode audio power amplifiers. It is shown that the total integration of the power supply and Class D audio power amplifier into one compact direct converter can simplify the design, increase...... efficiency, reduce the product volume and lower its cost. As an example, the principle of operation and the measurements made on a direct-conversion switching-mode audio power amplifier with active capacitive voltage clamp are presented....

An Exploratory Evaluation of User Interfaces for 3D Audio Mixing

DEFF Research Database (Denmark)

Gelineck, Steven; Korsgaard, Dannie Michael

2015-01-01

The paper presents an exploratory evaluation comparing different versions of a mid-air gesture based interface for mixing 3D audio exploring: (1) how such an interface generally compares to a more traditional physical interface, (2) methods for grabbing/releasing audio channels in mid-air and (3...

A 240W Monolithic Class-D Audio Amplifier Output Stage

DEFF Research Database (Denmark)

Nyboe, Flemming; Kaya, Cetin; Risbo, Lars

2006-01-01

A single-channel class-D audio amplifier output stage outputs 240W undipped into 4Omega 0.1% open-loop THD+N allows using the device in a fully-digital audio signal path with no feedback. The output current capability is plusmn18A and the part is fabricated in a 0.4mum/1.8mum high-voltage Bi...

A survey of systems for massive stream analytics

OpenAIRE

Singh, Maninder Pal; Hoque, Mohammad A.; Tarkoma, Sasu

2016-01-01

The immense growth of data demands switching from traditional data processing solutions to systems, which can process a continuous stream of real time data. Various applications employ stream processing systems to provide solutions to emerging Big Data problems. Open-source solutions such as Storm, Spark Streaming, and S4 are the attempts to answer key stream processing questions. The recent introduction of real time stream processing commercial solutions such as Amazon Kinesis, IBM Infospher...

Design guidelines for audio presentation of graphs and tables

OpenAIRE

Brown, L.M.; Brewster, S.A.; Ramloll, S.A.; Burton, R.; Riedel, B.

2003-01-01

Audio can be used to make visualisations accessible to blind and visually impaired people. The MultiVis Project has carried out research into suitable methods for presenting graphs and tables to blind people through the use of both speech and non-speech audio. This paper presents guidelines extracted from this research. These guidelines will enable designers to implement visualisation systems for blind and visually impaired users, and will provide a framework for researchers wishing to invest...

Can We Afford These Affordances? GarageBand and the Double-Edged Sword of the Digital Audio Workstation

Science.gov (United States)

Bell, Adam Patrick

2015-01-01

The proliferation of computers, tablets, and smartphones has resulted in digital audio workstations (DAWs) such as GarageBand in being some of the most widely distributed musical instruments. Positing that software designers are dictating the music education of DAW-dependent music-makers, I examine the fallacy that music-making applications such…

The implementation of Project-Based Learning in courses Audio Video to Improve Employability Skills

Science.gov (United States)

Sulistiyo, Edy; Kustono, Djoko; Purnomo; Sutaji, Eddy

2018-04-01

This paper presents a project-based learning (PjBL) in subjects with Audio Video the Study Programme Electro Engineering Universitas Negeri Surabaya which consists of two ways namely the design of the prototype audio-video and assessment activities project-based learning tailored to the skills of the 21st century in the form of employability skills. The purpose of learning innovation is applying the lab work obtained in the theory classes. The PjBL aims to motivate students, centering on the problems of teaching in accordance with the world of work. Measures of learning include; determine the fundamental questions, designs, develop a schedule, monitor the learners and progress, test the results, evaluate the experience, project assessment, and product assessment. The results of research conducted showed the level of mastery of the ability to design tasks (of 78.6%), technical planning (39,3%), creativity (42,9%), innovative (46,4%), problem solving skills (the 57.1%), skill to communicate (75%), oral expression (75%), searching and understanding information (to 64.3%), collaborative work skills (71,4%), and classroom conduct (of 78.6%). In conclusion, instructors have to do the reflection and make improvements in some of the aspects that have a level of mastery of the skills less than 60% both on the application of project-based learning courses, audio video.

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

Science.gov (United States)

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

2010-01-01

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…

Revealing the ecological content of long-duration audio-recordings of the environment through clustering and visualisation.

Science.gov (United States)

Phillips, Yvonne F; Towsey, Michael; Roe, Paul

2018-01-01

Audio recordings of the environment are an increasingly important technique to monitor biodiversity and ecosystem function. While the acquisition of long-duration recordings is becoming easier and cheaper, the analysis and interpretation of that audio remains a significant research area. The issue addressed in this paper is the automated reduction of environmental audio data to facilitate ecological investigations. We describe a method that first reduces environmental audio to vectors of acoustic indices, which are then clustered. This can reduce the audio data by six to eight orders of magnitude yet retain useful ecological information. We describe techniques to visualise sequences of cluster occurrence (using for example, diel plots, rose plots) that assist interpretation of environmental audio. Colour coding acoustic clusters allows months and years of audio data to be visualised in a single image. These techniques are useful in identifying and indexing the contents of long-duration audio recordings. They could also play an important role in monitoring long-term changes in species abundance brought about by habitat degradation and/or restoration.

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

Science.gov (United States)

Dickinson Public Schools, ND. Instructional Media 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)…

Side Stream Filtration for Cooling Towers

Energy Technology Data Exchange (ETDEWEB)

None

2012-10-20

This technology evaluation assesses side stream filtration options for cooling towers, with an objective to assess key attributes that optimize energy and water savings along with providing information on specific technology and implementation options. This information can be used to assist Federal sites to determine which options may be most appropriate for their applications. This evaluation provides an overview of the characterization of side stream filtration technology, describes typical applications, and details specific types of filtration technology.

Design And Construction Of 300W Audio Power Amplifier For Classroom

Directory of Open Access Journals (Sweden)

Shune Lei Aung

2015-07-01

Full Text Available Abstract This paper describes the design and construction of 300W audio power amplifier for classroom. In the construction of this amplifier microphone preamplifier tone preamplifier equalizer line amplifier output power amplifier and sound level indicator are included. The output power amplifier is designed as O.C.L system and constructed by using Class B among many types of amplifier classes. There are two types in O.C.L system quasi system and complementary system. Between them the complementary system is used in the construction of 300W audio power amplifier. The Multisim software is utilized for the construction of audio power amplifier.

Créer des ressources audio pour le cours de FLE

Directory of Open Access Journals (Sweden)

Florence Gérard Lojacono

2010-01-01

Full Text Available These last ten years, web applicationshave gained ascendency over the consumersociety as shown by the success of iTunesand the increase of podcasting. The academicworld, particularly in the field oflanguage teaching, could take advantage ofthis massive use of audio files. The creationand the diffusion of customized ad hocaudio files and the broadcast of these resourcesthrough educational podcasts addressthe upcoming challenges of a knowledgebased society. Teaching and learningwith audio files also meet the recommendationsof the European Higher EducationArea (EHEA. This paper will provide languageteachers, especially French teachers,with the tools to create, edit, upload andplay their own audio files. No specific computerskills are required.

Perceptual Coding of Audio Signals Using Adaptive Time-Frequency Transform

Directory of Open Access Journals (Sweden)

Umapathy Karthikeyan

2007-01-01

Full Text Available Wide band digital audio signals have a very high data-rate associated with them due to their complex nature and demand for high-quality reproduction. Although recent technological advancements have significantly reduced the cost of bandwidth and miniaturized storage facilities, the rapid increase in the volume of digital audio content constantly compels the need for better compression algorithms. Over the years various perceptually lossless compression techniques have been introduced, and transform-based compression techniques have made a significant impact in recent years. In this paper, we propose one such transform-based compression technique, where the joint time-frequency (TF properties of the nonstationary nature of the audio signals were exploited in creating a compact energy representation of the signal in fewer coefficients. The decomposition coefficients were processed and perceptually filtered to retain only the relevant coefficients. Perceptual filtering (psychoacoustics was applied in a novel way by analyzing and performing TF specific psychoacoustics experiments. An added advantage of the proposed technique is that, due to its signal adaptive nature, it does not need predetermined segmentation of audio signals for processing. Eight stereo audio signal samples of different varieties were used in the study. Subjective (mean opinion score—MOS listening tests were performed and the subjective difference grades (SDG were used to compare the performance of the proposed coder with MP3, AAC, and HE-AAC encoders. Compression ratios in the range of 8 to 40 were achieved by the proposed technique with subjective difference grades (SDG ranging from –0.53 to –2.27.

Knowledge discovery from data streams

CERN Document Server

Gama, Joao

2010-01-01

Since the beginning of the Internet age and the increased use of ubiquitous computing devices, the large volume and continuous flow of distributed data have imposed new constraints on the design of learning algorithms. Exploring how to extract knowledge structures from evolving and time-changing data, Knowledge Discovery from Data Streams presents a coherent overview of state-of-the-art research in learning from data streams.The book covers the fundamentals that are imperative to understanding data streams and describes important applications, such as TCP/IP traffic, GPS data, sensor networks,

Audio pacemaker : Walking, talking indigenous knowledge

CSIR Research Space (South Africa)

Bidwell, NJ

2012-10-01

Full Text Available stream_source_info Bidwell1_2012_ABSTRACT ONLY.pdf.txt stream_content_type text/plain stream_size 1422 Content-Encoding ISO-8859-1 stream_name Bidwell1_2012_ABSTRACT ONLY.pdf.txt Content-Type text/plain; charset=ISO-8859-1...

Pengaruh layanan informasi bimbingan konseling berbantuan media audio visual terhadap empati siswa

Directory of Open Access Journals (Sweden)

Rita Kumalasari

2017-05-01

The results of research effective of audio-visual media counseling techniques effective and practical to increase the empathy of students are rational design, key concepts, understanding, purpose, content models, the role and qualifications tutor (counselor is expected, procedures or steps in the implementation of the audio-visual, evaluation, follow-up, support system. This research is proven effective in improving student behavior. Empathy behavior of students increases 28.9% from the previous 45.08% increase to 73.98%. This increase occurred in all aspects of empathy Keywords: Effective, Audio visual, Empathy

Stream-processing pipelines: processing of streams on multiprocessor architecture

NARCIS (Netherlands)

Kavaldjiev, N.K.; Smit, Gerardus Johannes Maria; Jansen, P.G.

In this paper we study the timing aspects of the operation of stream-processing applications that run on a multiprocessor architecture. Dependencies are derived for the processing and communication times of the processors in such a system. Three cases of real-time constrained operation and four

Quick Response (QR) Codes for Audio Support in Foreign Language Learning

Science.gov (United States)

Vigil, Kathleen Murray

2017-01-01

This study explored the potential benefits and barriers of using quick response (QR) codes as a means by which to provide audio materials to middle-school students learning Spanish as a foreign language. Eleven teachers of Spanish to middle-school students created transmedia materials containing QR codes linking to audio resources. Students…

About the theory of congested transport streams

OpenAIRE

Valeriy GUK

2009-01-01

Talked about a theory, based on integrity of continuous motion of a transport stream. Placing of car and its speed is in a stream - second. Principle of application of the generalized methods of design and new descriptions of the states of transport streams opens up. Travelling and transport potentials are set, and also external capacity of the system a «transport stream» is an exergy, that allows to make differential equation and decide the applied tasks of organization of travelling motion....

Spatial audio quality perception (part 2)

DEFF Research Database (Denmark)

Conetta, R.; Brookes, T.; Rumsey, F.

2015-01-01

location, envelopment, coverage angle, ensemble width, and spaciousness. They can also impact timbre, and changes to timbre can then influence spatial perception. Previously obtained data was used to build a regression model of perceived spatial audio quality in terms of spatial and timbral metrics...

Nutrient spiraling in streams and river networks

Science.gov (United States)

Ensign, Scott H.; Doyle, Martin W.

2006-12-01

Over the past 3 decades, nutrient spiraling has become a unifying paradigm for stream biogeochemical research. This paper presents (1) a quantitative synthesis of the nutrient spiraling literature and (2) application of these data to elucidate trends in nutrient spiraling within stream networks. Results are based on 404 individual experiments on ammonium (NH4), nitrate (NO3), and phosphate (PO4) from 52 published studies. Sixty-nine percent of the experiments were performed in first- and second-order streams, and 31% were performed in third- to fifth-order streams. Uptake lengths, Sw, of NH4 (median = 86 m) and PO4 (median = 96 m) were significantly different (α = 0.05) than NO3 (median = 236 m). Areal uptake rates of NH4 (median = 28 μg m-2 min-1) were significantly different than NO3 and PO4 (median = 15 and 14 μg m-2 min-1, respectively). There were significant differences among NH4, NO3, and PO4 uptake velocity (median = 5, 1, and 2 mm min-1, respectively). Correlation analysis results were equivocal on the effect of transient storage on nutrient spiraling. Application of these data to a stream network model showed that recycling (defined here as stream length ÷ Sw) of NH4 and NO3 generally increased with stream order, while PO4 recycling remained constant along a first- to fifth-order stream gradient. Within this hypothetical stream network, cumulative NH4 uptake decreased slightly with stream order, while cumulative NO3 and PO4 uptake increased with stream order. These data suggest the importance of larger rivers to nutrient spiraling and the need to consider how stream networks affect nutrient flux between terrestrial and marine ecosystems.

Overview of the 2015 Workshop on Speech, Language and Audio in Multimedia

NARCIS (Netherlands)

Gravier, Guillaume; Jones, Gareth J.F.; Larson, Martha; Ordelman, Roeland J.F.

2015-01-01

The Workshop on Speech, Language and Audio in Multimedia (SLAM) positions itself at at the crossroad of multiple scientific fields - music and audio processing, speech processing, natural language processing and multimedia - to discuss and stimulate research results, projects, datasets and

Audio-visual Classification and Fusion of Spontaneous Affect Data in Likelihood Space

NARCIS (Netherlands)

Nicolaou, Mihalis A.; Gunes, Hatice; Pantic, Maja

2010-01-01

This paper focuses on audio-visual (using facial expression, shoulder and audio cues) classification of spontaneous affect, utilising generative models for classification (i) in terms of Maximum Likelihood Classification with the assumption that the generative model structure in the classifier is

Can audio recording of outpatient consultations improve patient outcome?

DEFF Research Database (Denmark)

Wolderslund, Maiken; Kofoed, Poul-Erik; Axboe, Mette

different departments: Orthopedics, Urology, Internal Medicine and Pediatrics. A total of 5,460 patients will be included from the outpatient clinics. All patients randomized to an intervention group are offered audio recording of their consultation. An Interactive Voice Response platform enables an audio....... The intervention will be evaluated using a questionnaire measuring different aspect of patients recall and understanding of the information given, patients need for additional information subsequent to the consultation and their overall satisfaction with the consultation. Results The study will be conducted from...

A 240W Monolithic Class-D Audio Amplifier Output Stage

OpenAIRE

Nyboe, Flemming; Kaya, Cetin; Risbo, Lars; Andreani, Pietro

2006-01-01

A single-channel class-D audio amplifier output stage outputs 240W undipped into 4Omega 0.1% open-loop THD+N allows using the device in a fully-digital audio signal path with no feedback. The output current capability is plusmn18A and the part is fabricated in a 0.4mum/1.8mum high-voltage BiCMOS process. Over-current sensing protects the output from short circuits.

Audio Control Handbook For Radio and Television Broadcasting. Third Revised Edition.

Science.gov (United States)

Oringel, Robert S.

Audio control is the operation of all the types of sound equipment found in the studios and control rooms of a radio or television station. Written in a nontechnical style for beginners, the book explains thoroughly the operation of all types of audio equipment. Diagrams and photographs of commercial consoles, microphones, turntables, and tape…

Fault Diagnosis using Audio and Vibration Signals in a Circulating Pump

International Nuclear Information System (INIS)

Henríquez, P; Alonso, J B; Ferrer, M A; Travieso, C M; Gómez, G

2012-01-01

This paper presents the use of audio and vibration signals in fault diagnosis of a circulating pump. The novelty of this paper is the use of audio signals acquired by microphones. The objective of this paper is to determine if audio signals are capable to distinguish between normal and different abnormal conditions in a circulating pump. In order to compare results, vibration signals are also acquired and analysed. Wavelet package is used to obtain the energies in different frequency bands from the audio and vibration signals. Neural networks are used to evaluate the discrimination ability of the extracted features between normal and fault conditions. The results show that information from sound signals can distinguish between normal and different faulty conditions with a success rate of 83.33%, 98% and 91.33% for each microphone respectively. These success rates are similar and even higher that those obtained from accelerometers (68%, 90.67% and 71.33% for each accelerometer respectively). Success rates also show that the position of microphones and accelerometers affects on the final results.

Application of acoustic agglomeration for removing sulfuric acid mist from air stream

Directory of Open Access Journals (Sweden)

Asghar Sadighzadeh

2018-01-01

Full Text Available The application of acoustic fields at high sound pressure levels (SPLs for removing sulfuric acid mists from the air stream was studied. An acoustic agglomeration chamber was used to conduct the experiments. The studied SPLs ranged from 115 to 165 decibel (dB, with three inlet concentrations of acid mist at 5–10, 15–20, and 25–30 ppm. The air flow rates for conducting experiments were 20, 30, and 40 L min−1. The concentration of sulfuric acid mist was measured using US Environmental Protection Agency Method 8 at inlet and outlet of the chamber. The resonance frequencies for experiments were found to be 852, 1410, and 3530 Hz. The maximum acoustic agglomeration efficiency of 86% was obtained at optimum frequency of 852 Hz. The analysis of variance test revealed significant differences between agglomeration efficiency at three resonance frequencies (p-value < 0.001. The maximum acoustic agglomeration efficiency was obtained at SPL level of 165 dB. High initial concentrations of acid mists and lower air flow rates enhance the acoustic agglomeration of mists. High removal efficiency of acid mists from air stream could be achieved by the application of acoustic agglomeration method with appropriate range of frequencies and SPLs. Keywords: Sulfuric acid, Mist, Acoustic agglomeration, SPL

Audio-Visual Speech Recognition Using Lip Information Extracted from Side-Face Images

Directory of Open Access Journals (Sweden)

Koji Iwano

2007-03-01

Full Text Available This paper proposes an audio-visual speech recognition method using lip information extracted from side-face images as an attempt to increase noise robustness in mobile environments. Our proposed method assumes that lip images can be captured using a small camera installed in a handset. Two different kinds of lip features, lip-contour geometric features and lip-motion velocity features, are used individually or jointly, in combination with audio features. Phoneme HMMs modeling the audio and visual features are built based on the multistream HMM technique. Experiments conducted using Japanese connected digit speech contaminated with white noise in various SNR conditions show effectiveness of the proposed method. Recognition accuracy is improved by using the visual information in all SNR conditions. These visual features were confirmed to be effective even when the audio HMM was adapted to noise by the MLLR method.

Turkish Music Genre Classification using Audio and Lyrics Features

Directory of Open Access Journals (Sweden)

Önder ÇOBAN

2017-05-01

Full Text Available Music Information Retrieval (MIR has become a popular research area in recent years. In this context, researchers have developed music information systems to find solutions for such major problems as automatic playlist creation, hit song detection, and music genre or mood classification. Meta-data information, lyrics, or melodic content of music are used as feature resource in previous works. However, lyrics do not often used in MIR systems and the number of works in this field is not enough especially for Turkish. In this paper, firstly, we have extended our previously created Turkish MIR (TMIR dataset, which comprises of Turkish lyrics, by including the audio file of each song. Secondly, we have investigated the effect of using audio and textual features together or separately on automatic Music Genre Classification (MGC. We have extracted textual features from lyrics using different feature extraction models such as word2vec and traditional Bag of Words. We have conducted our experiments on Support Vector Machine (SVM algorithm and analysed the impact of feature selection and different feature groups on MGC. We have considered lyrics based MGC as a text classification task and also investigated the effect of term weighting method. Experimental results show that textual features can also be effective as well as audio features for Turkish MGC, especially when a supervised term weighting method is employed. We have achieved the highest success rate as 99,12\\% by using both audio and textual features together.

Perceptual Coding of Audio Signals Using Adaptive Time-Frequency Transform

Directory of Open Access Journals (Sweden)

Karthikeyan Umapathy

2007-08-01

Full Text Available Wide band digital audio signals have a very high data-rate associated with them due to their complex nature and demand for high-quality reproduction. Although recent technological advancements have significantly reduced the cost of bandwidth and miniaturized storage facilities, the rapid increase in the volume of digital audio content constantly compels the need for better compression algorithms. Over the years various perceptually lossless compression techniques have been introduced, and transform-based compression techniques have made a significant impact in recent years. In this paper, we propose one such transform-based compression technique, where the joint time-frequency (TF properties of the nonstationary nature of the audio signals were exploited in creating a compact energy representation of the signal in fewer coefficients. The decomposition coefficients were processed and perceptually filtered to retain only the relevant coefficients. Perceptual filtering (psychoacoustics was applied in a novel way by analyzing and performing TF specific psychoacoustics experiments. An added advantage of the proposed technique is that, due to its signal adaptive nature, it does not need predetermined segmentation of audio signals for processing. Eight stereo audio signal samples of different varieties were used in the study. Subjective (mean opinion score—MOS listening tests were performed and the subjective difference grades (SDG were used to compare the performance of the proposed coder with MP3, AAC, and HE-AAC encoders. Compression ratios in the range of 8 to 40 were achieved by the proposed technique with subjective difference grades (SDG ranging from –0.53 to –2.27.

The relationship between basic audio quality and overall listening experience.

Science.gov (United States)

Schoeffler, Michael; Herre, Jürgen

2016-09-01

Basic audio quality (BAQ) is a well-known perceptual attribute, which is rated in various listening test methods to measure the performance of audio systems. Unfortunately, when it comes to purchasing audio systems, BAQ might not have a significant influence on the customers' buying decisions since other factors, like brand loyalty, might be more important. In contrast to BAQ, overall listening experience (OLE) is an affective attribute which incorporates all aspects that are important to an individual assessor, including his or her preference for music genre and audio quality. In this work, the relationship between BAQ and OLE is investigated in more detail. To this end, an experiment was carried out, in which participants rated the BAQ and the OLE of music excerpts with different timbral and spatial degradations. In a between-group-design procedure, participants were assigned into two groups, in each of which a different set of stimuli was rated. The results indicate that rating of both attributes, BAQ and OLE, leads to similar rankings, even if a different set of stimuli is rated. In contrast to the BAQ ratings, which were more influenced by timbral than spatial degradations, the OLE ratings were almost equally influenced by timbral and spatial degradations.

Design of an integrated analog controller for a Class-D Audio Amplifier

OpenAIRE

Verbrugghe, Jochen; De Bock, Maarten; Rombouts, Pieter

2009-01-01

An integrated analog controller for a self-oscillating class-D audio power amplifier is designed in a 0.35 μm CMOS technology for a 3.3 Volt power supply. It is intended to be used with an external output stage and passive filter, for medium power applications of upto a few 100 Watts. The controller was optimized with regard to its loop gain to suppress the distortion of the output stage. In typical commercially available output stages, the distortion is dominated by dead time effects and th...

StreamQRE: Modular Specification and Efficient Evaluation of Quantitative Queries over Streaming Data.

Science.gov (United States)

Mamouras, Konstantinos; Raghothaman, Mukund; Alur, Rajeev; Ives, Zachary G; Khanna, Sanjeev

2017-06-01

Real-time decision making in emerging IoT applications typically relies on computing quantitative summaries of large data streams in an efficient and incremental manner. To simplify the task of programming the desired logic, we propose StreamQRE, which provides natural and high-level constructs for processing streaming data. Our language has a novel integration of linguistic constructs from two distinct programming paradigms: streaming extensions of relational query languages and quantitative extensions of regular expressions. The former allows the programmer to employ relational constructs to partition the input data by keys and to integrate data streams from different sources, while the latter can be used to exploit the logical hierarchy in the input stream for modular specifications. We first present the core language with a small set of combinators, formal semantics, and a decidable type system. We then show how to express a number of common patterns with illustrative examples. Our compilation algorithm translates the high-level query into a streaming algorithm with precise complexity bounds on per-item processing time and total memory footprint. We also show how to integrate approximation algorithms into our framework. We report on an implementation in Java, and evaluate it with respect to existing high-performance engines for processing streaming data. Our experimental evaluation shows that (1) StreamQRE allows more natural and succinct specification of queries compared to existing frameworks, (2) the throughput of our implementation is higher than comparable systems (for example, two-to-four times greater than RxJava), and (3) the approximation algorithms supported by our implementation can lead to substantial memory savings.

16 CFR 307.8 - Requirements for disclosure in audiovisual and audio advertising.

Science.gov (United States)

2010-01-01

... 16 Commercial Practices 1 2010-01-01 2010-01-01 false Requirements for disclosure in audiovisual and audio advertising. 307.8 Section 307.8 Commercial Practices FEDERAL TRADE COMMISSION REGULATIONS... ACT OF 1986 Advertising Disclosures § 307.8 Requirements for disclosure in audiovisual and audio...

Introduction of audio gating to further reduce organ motion in breathing synchronized radiotherapy

International Nuclear Information System (INIS)

Kubo, H. Dale; Wang Lili

2002-01-01

With breathing synchronized radiotherapy (BSRT), a voltage signal derived from an organ displacement detector is usually displayed on the vertical axis whereas the elapsed time is shown on the horizontal axis. The voltage gate window is set on the breathing voltage signal. Whenever the breathing signal falls between the two gate levels, a gate pulse is produced to enable the treatment machine. In this paper a new gating mechanism, audio (or time-sequence) gating, is introduced and is integrated into the existing voltage gating system. The audio gating takes advantage of the repetitive nature of the breathing signal when repetitive audio instruction is given to the patient. The audio gating is aimed at removing the regions of sharp rises and falls in the breathing signal that cannot be removed by the voltage gating. When the breathing signal falls between voltage gate levels as well as between audio-gate levels, the voltage- and audio-gated radiotherapy (ART) system will generate an AND gate pulse. When this gate pulse is received by a linear accelerator, the linear accelerator becomes 'enabled' for beam delivery and will deliver the beam when all other interlocks are removed. This paper describes a new gating mechanism and a method of recording beam-on signal, both of which are, configured into a laptop computer. The paper also presents evidence of some clinical advantages achieved with the ART system

The MIT Lincoln Laboratory RT-04F Diarization Systems: Applications to Broadcast Audio and Telephone Conversations

Science.gov (United States)

2004-11-01

this paper we describe the systems developed by MITLL and used in DARPA EARS Rich Transcription Fall 2004 (RT-04F) speaker diarization evaluation...many types of audio sources, the focus if the DARPA EARS project and the NIST Rich Transcription evaluations is primarily speaker diarization ...present or samples of any of the speakers . An overview of the general diarization problem and approaches can be found in [1]. In this paper, we

Objective Audio Quality Assessment Based on Spectro-Temporal Modulation Analysis

OpenAIRE

Guo, Ziyuan

2011-01-01

Objective audio quality assessment is an interdisciplinary research area that incorporates audiology and machine learning. Although much work has been made on the machine learning aspect, the audiology aspect also deserves investigation. This thesis proposes a non-intrusive audio quality assessment algorithm, which is based on an auditory model that simulates human auditory system. The auditory model is based on spectro-temporal modulation analysis of spectrogram, which has been proven to be ...

Feature Fusion Based Audio-Visual Speaker Identification Using Hidden Markov Model under Different Lighting Variations

Directory of Open Access Journals (Sweden)

Md. Rabiul Islam

2014-01-01

Full Text Available The aim of the paper is to propose a feature fusion based Audio-Visual Speaker Identification (AVSI system with varied conditions of illumination environments. Among the different fusion strategies, feature level fusion has been used for the proposed AVSI system where Hidden Markov Model (HMM is used for learning and classification. Since the feature set contains richer information about the raw biometric data than any other levels, integration at feature level is expected to provide better authentication results. In this paper, both Mel Frequency Cepstral Coefficients (MFCCs and Linear Prediction Cepstral Coefficients (LPCCs are combined to get the audio feature vectors and Active Shape Model (ASM based appearance and shape facial features are concatenated to take the visual feature vectors. These combined audio and visual features are used for the feature-fusion. To reduce the dimension of the audio and visual feature vectors, Principal Component Analysis (PCA method is used. The VALID audio-visual database is used to measure the performance of the proposed system where four different illumination levels of lighting conditions are considered. Experimental results focus on the significance of the proposed audio-visual speaker identification system with various combinations of audio and visual features.

SmartCell: An Energy Efficient Coarse-Grained Reconfigurable Architecture for Stream-Based Applications

Directory of Open Access Journals (Sweden)

Liang Cao

2009-01-01

Full Text Available This paper presents SmartCell, a novel coarse-grained reconfigurable architecture, which tiles a large number of processor elements with reconfigurable interconnection fabrics on a single chip. SmartCell is able to provide high performance and energy efficient processing for stream-based applications. It can be configured to operate in various modes, such as SIMD, MIMD, and systolic array. This paper describes the SmartCell architecture design, including processing element, reconfigurable interconnection fabrics, instruction and control process, and configuration scheme. The SmartCell prototype with 64 PEs is implemented using 0.13  m CMOS standard cell technology. The core area is about 8.5  , and the power consumption is about 1.6 mW/MHz. The performance is evaluated through a set of benchmark applications, and then compared with FPGA, ASIC, and two well-known reconfigurable architectures including RaPiD and Montium. The results show that the SmartCell can bridge the performance and flexibility gap between ASIC and FPGA. It is also about 8% and 69% more energy efficient than Montium and RaPiD systems for evaluated benchmarks. Meanwhile, SmartCell can achieve 4 and 2 times more throughput gains when comparing with Montium and RaPiD, respectively. It is concluded that SmartCell system is a promising reconfigurable and energy efficient architecture for stream processing.

Advanced content delivery, streaming, and cloud services

CERN Document Server

Sitaraman, Ramesh Kumar; Robinson, Dom

2014-01-01

While other books on the market provide limited coverage of advanced CDNs and streaming technologies, concentrating solely on the fundamentals, this book provides an up-to-date comprehensive coverage of the state-of-the-art advancements in CDNs, with a special focus on Cloud-based CDNs. The book includes CDN and media streaming basics, performance models, practical applications, and business analysis. It features industry case studies, CDN applications, and open research issues to aid practitioners and researchers, and a market analysis to provide a reference point for commercial entities. The book covers Adaptive Bitrate Streaming (ABR), Content Delivery Cloud (CDC), Web Acceleration, Front End Optimization (FEO), Transparent Caching, Next Generation CDNs, CDN Business Intelligence and more.

Estimation of inhalation flow profile using audio-based methods to assess inhaler medication adherence

Science.gov (United States)

Lacalle Muls, Helena; Costello, Richard W.; Reilly, Richard B.

2018-01-01

Asthma and chronic obstructive pulmonary disease (COPD) patients are required to inhale forcefully and deeply to receive medication when using a dry powder inhaler (DPI). There is a clinical need to objectively monitor the inhalation flow profile of DPIs in order to remotely monitor patient inhalation technique. Audio-based methods have been previously employed to accurately estimate flow parameters such as the peak inspiratory flow rate of inhalations, however, these methods required multiple calibration inhalation audio recordings. In this study, an audio-based method is presented that accurately estimates inhalation flow profile using only one calibration inhalation audio recording. Twenty healthy participants were asked to perform 15 inhalations through a placebo Ellipta™ DPI at a range of inspiratory flow rates. Inhalation flow signals were recorded using a pneumotachograph spirometer while inhalation audio signals were recorded simultaneously using the Inhaler Compliance Assessment device attached to the inhaler. The acoustic (amplitude) envelope was estimated from each inhalation audio signal. Using only one recording, linear and power law regression models were employed to determine which model best described the relationship between the inhalation acoustic envelope and flow signal. Each model was then employed to estimate the flow signals of the remaining 14 inhalation audio recordings. This process repeated until each of the 15 recordings were employed to calibrate single models while testing on the remaining 14 recordings. It was observed that power law models generated the highest average flow estimation accuracy across all participants (90.89±0.9% for power law models and 76.63±2.38% for linear models). The method also generated sufficient accuracy in estimating inhalation parameters such as peak inspiratory flow rate and inspiratory capacity within the presence of noise. Estimating inhaler inhalation flow profiles using audio based methods may be

An ESL Audio-Script Writing Workshop

Science.gov (United States)

Miller, Carla

2012-01-01

The roles of dialogue, collaborative writing, and authentic communication have been explored as effective strategies in second language writing classrooms. In this article, the stages of an innovative, multi-skill writing method, which embeds students' personal voices into the writing process, are explored. A 10-step ESL Audio Script Writing Model…

Digital signal processing methods and algorithms for audio conferencing systems

OpenAIRE

Lindström, Fredric

2007-01-01

Today, we are interconnected almost all over the planet. Large multinational companies operate worldwide, but also an increasing number of small and medium sized companies do business overseas. As people travel to meet and do businesses, the already exposed earth is subject to even more strain. Audio conferencing is an attractive alternative to travel, which is becoming more and more appreciated. Audio conferences can of course not replace all types of meetings, but can help companies to cut ...

Identifying hidden voice and video streams

Science.gov (United States)

Fan, Jieyan; Wu, Dapeng; Nucci, Antonio; Keralapura, Ram; Gao, Lixin

2009-04-01

Given the rising popularity of voice and video services over the Internet, accurately identifying voice and video traffic that traverse their networks has become a critical task for Internet service providers (ISPs). As the number of proprietary applications that deliver voice and video services to end users increases over time, the search for the one methodology that can accurately detect such services while being application independent still remains open. This problem becomes even more complicated when voice and video service providers like Skype, Microsoft, and Google bundle their voice and video services with other services like file transfer and chat. For example, a bundled Skype session can contain both voice stream and file transfer stream in the same layer-3/layer-4 flow. In this context, traditional techniques to identify voice and video streams do not work. In this paper, we propose a novel self-learning classifier, called VVS-I , that detects the presence of voice and video streams in flows with minimum manual intervention. Our classifier works in two phases: training phase and detection phase. In the training phase, VVS-I first extracts the relevant features, and subsequently constructs a fingerprint of a flow using the power spectral density (PSD) analysis. In the detection phase, it compares the fingerprint of a flow to the existing fingerprints learned during the training phase, and subsequently classifies the flow. Our classifier is not only capable of detecting voice and video streams that are hidden in different flows, but is also capable of detecting different applications (like Skype, MSN, etc.) that generate these voice/video streams. We show that our classifier can achieve close to 100% detection rate while keeping the false positive rate to less that 1%.

One Message, Many Voices: Mobile Audio Counselling in Health Education.

Science.gov (United States)

Pimmer, Christoph; Mbvundula, Francis

2018-01-01

Health workers' use of counselling information on their mobile phones for health education is a central but little understood phenomenon in numerous mobile health (mHealth) projects in Sub-Saharan Africa. Drawing on empirical data from an interpretive case study in the setting of the Millennium Villages Project in rural Malawi, this research investigates the ways in which community health workers (CHWs) perceive that audio-counselling messages support their health education practice. Three main themes emerged from the analysis: phone-aided audio counselling (1) legitimises the CHWs' use of mobile phones during household visits; (2) helps CHWs to deliver a comprehensive counselling message; (3) supports CHWs in persuading communities to change their health practices. The findings show the complexity and interplay of the multi-faceted, sociocultural, political, and socioemotional meanings associated with audio-counselling use. Practical implications and the demand for further research are discussed.

Impact of Audio-Coaching on the Position of Lung Tumors

International Nuclear Information System (INIS)

Haasbeek, Cornelis J.A.; Spoelstra, Femke; Lagerwaard, Frank J.; Soernsen de Koste, John R. van; Cuijpers, Johan P.; Slotman, Ben J.; Senan, Suresh

2008-01-01

Purpose: Respiration-induced organ motion is a major source of positional, or geometric, uncertainty in thoracic radiotherapy. Interventions to mitigate the impact of motion include audio-coached respiration-gated radiotherapy (RGRT). To assess the impact of coaching on average tumor position during gating, we analyzed four-dimensional computed tomography (4DCT) scans performed both with and without audio-coaching. Methods and Materials: Our RGRT protocol requires that an audio-coached 4DCT scan is performed when the initial free-breathing 4DCT indicates a potential benefit with gating. We retrospectively analyzed 22 such paired scans in patients with well-circumscribed tumors. Changes in lung volume and position of internal target volumes (ITV) generated in three consecutive respiratory phases at both end-inspiration and end-expiration were analyzed. Results: Audio-coaching increased end-inspiration lung volumes by a mean of 10.2% (range, -13% to +43%) when compared with free breathing (p = 0.001). The mean three-dimensional displacement of the center of ITV was 3.6 mm (SD, 2.5; range, 0.3-9.6mm), mainly caused by displacement in the craniocaudal direction. Displacement of ITV caused by coaching was more than 5 mm in 5 patients, all of whom were in the subgroup of 9 patients showing total tumor motion of 10 mm or more during both coached and uncoached breathing. Comparable ITV displacements were observed at end-expiration phases of the 4DCT. Conclusions: Differences in ITV position exceeding 5 mm between coached and uncoached 4DCT scans were detected in up to 56% of mobile tumors. Both end-inspiration and end-expiration RGRT were susceptible to displacements. This indicates that the method of audio-coaching should remain unchanged throughout the course of treatment

audio-ultrasonic waves by argon gas discharge

International Nuclear Information System (INIS)

Ragheb, M.S.

2010-01-01

in the present work, wave emission formed by audio-ultrasonic plasma is investigated. the evidence of the magnetic and electric fields presence is performed by experimental technique. comparison between experimental field measurements and several plasma wave methods reveals the plasma audio-ultrasonic radiations mode. this plasma is a symmetrically driven capacitive discharge, consisting of three interactive regions: the electrodes, the sheaths, and the positive column regions . the discharge voltage is up to 900 volts, the discharge current flowing through the plasma attains a value of 360 mA .the frequency of the discharge voltage covers the audio and the ultrasonic range up to 100 khz. the effective plasma working distance has increased to attain the total length of the tube of 40 cm. a non-disturbing method using an external coil is used to measure the electric discharge field in a plane perpendicular to that of the plasma axe tube. this method proves the existence of a current flowing in a direction perpendicular to the plasma axe tube. a system of minute coils sensors proved the existence of two fields in two perpendicular directions . comparison between different observed fields reveals the existence of propagating electromagnetic waves due to the alternating current flowing through the skin plasma tube. the field intensity distribution along the tube draws the discharge current behavior between the two plasma electrodes that can be used to predict the range of the plasma discharge current.

Stretch-minimising stream surfaces

KAUST Repository

Barton, Michael; Kosinka, Jin; Calo, Victor M.

2015-01-01

We study the problem of finding stretch-minimising stream surfaces in a divergence-free vector field. These surfaces are generated by motions of seed curves that propagate through the field in a stretch minimising manner, i.e., they move without stretching or shrinking, preserving the length of their arbitrary arc. In general fields, such curves may not exist. How-ever, the divergence-free constraint gives rise to these 'stretch-free' curves that are locally arc-length preserving when infinitesimally propagated. Several families of stretch-free curves are identified and used as initial guesses for stream surface generation. These surfaces are subsequently globally optimised to obtain the best stretch-minimising stream surfaces in a given divergence-free vector field. Our algorithm was tested on benchmark datasets, proving its applicability to incompressible fluid flow simulations, where our stretch-minimising stream surfaces realistically reflect the flow of a flexible univariate object. © 2015 Elsevier Inc. All rights reserved.

Stretch-minimising stream surfaces

KAUST Repository

Barton, Michael

2015-05-01

We study the problem of finding stretch-minimising stream surfaces in a divergence-free vector field. These surfaces are generated by motions of seed curves that propagate through the field in a stretch minimising manner, i.e., they move without stretching or shrinking, preserving the length of their arbitrary arc. In general fields, such curves may not exist. How-ever, the divergence-free constraint gives rise to these \\'stretch-free\\' curves that are locally arc-length preserving when infinitesimally propagated. Several families of stretch-free curves are identified and used as initial guesses for stream surface generation. These surfaces are subsequently globally optimised to obtain the best stretch-minimising stream surfaces in a given divergence-free vector field. Our algorithm was tested on benchmark datasets, proving its applicability to incompressible fluid flow simulations, where our stretch-minimising stream surfaces realistically reflect the flow of a flexible univariate object. © 2015 Elsevier Inc. All rights reserved.

Comparison of Linear Prediction Models for Audio Signals

Directory of Open Access Journals (Sweden)

2009-03-01

Full Text Available While linear prediction (LP has become immensely popular in speech modeling, it does not seem to provide a good approach for modeling audio signals. This is somewhat surprising, since a tonal signal consisting of a number of sinusoids can be perfectly predicted based on an (all-pole LP model with a model order that is twice the number of sinusoids. We provide an explanation why this result cannot simply be extrapolated to LP of audio signals. If noise is taken into account in the tonal signal model, a low-order all-pole model appears to be only appropriate when the tonal components are uniformly distributed in the Nyquist interval. Based on this observation, different alternatives to the conventional LP model can be suggested. Either the model should be changed to a pole-zero, a high-order all-pole, or a pitch prediction model, or the conventional LP model should be preceded by an appropriate frequency transform, such as a frequency warping or downsampling. By comparing these alternative LP models to the conventional LP model in terms of frequency estimation accuracy, residual spectral flatness, and perceptual frequency resolution, we obtain several new and promising approaches to LP-based audio modeling.

Interactive real-time media streaming with reliable communication

Science.gov (United States)

Pan, Xunyu; Free, Kevin M.

2014-02-01

Streaming media is a recent technique for delivering multimedia information from a source provider to an end- user over the Internet. The major advantage of this technique is that the media player can start playing a multimedia file even before the entire file is transmitted. Most streaming media applications are currently implemented based on the client-server architecture, where a server system hosts the media file and a client system connects to this server system to download the file. Although the client-server architecture is successful in many situations, it may not be ideal to rely on such a system to provide the streaming service as users may be required to register an account using personal information in order to use the service. This is troublesome if a user wishes to watch a movie simultaneously while interacting with a friend in another part of the world over the Internet. In this paper, we describe a new real-time media streaming application implemented on a peer-to-peer (P2P) architecture in order to overcome these challenges within a mobile environment. When using the peer-to-peer architecture, streaming media is shared directly between end-users, called peers, with minimal or no reliance on a dedicated server. Based on the proposed software pɛvμa (pronounced [revma]), named for the Greek word meaning stream, we can host a media file on any computer and directly stream it to a connected partner. To accomplish this, pɛvμa utilizes the Microsoft .NET Framework and Windows Presentation Framework, which are widely available on various types of windows-compatible personal computers and mobile devices. With specially designed multi-threaded algorithms, the application can stream HD video at speeds upwards of 20 Mbps using the User Datagram Protocol (UDP). Streaming and playback are handled using synchronized threads that communicate with one another once a connection is established. Alteration of playback, such as pausing playback or tracking to a

Study of audio speakers containing ferrofluid

Energy Technology Data Exchange (ETDEWEB)

Rosensweig, R E [34 Gloucester Road, Summit, NJ 07901 (United States); Hirota, Y; Tsuda, S [Ferrotec, 1-4-14 Kyobashi, chuo-Ku, Tokyo 104-0031 (Japan); Raj, K [Ferrotec, 33 Constitution Drive, Bedford, NH 03110 (United States)

2008-05-21

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.

Using the Periscope Live Video-Streaming Application for Global Pathology Education: A Brief Introduction.

Science.gov (United States)

Fuller, Maren Y; Mukhopadhyay, Sanjay; Gardner, Jerad M

2016-07-21

Periscope is a live video-streaming smartphone application (app) that allows any individual with a smartphone to broadcast live video simultaneously to multiple smartphone users around the world. The aim of this review is to describe the potential of this emerging technology for global pathology education. To our knowledge, since the launch of the Periscope app (2015), only a handful of educational presentations by pathologists have been streamed as live video via Periscope. This review includes links to these initial attempts, a step-by-step guide for those interested in using the app for pathology education, and a summary of the pros and cons, including ethical/legal issues. We hope that pathologists will appreciate the potential of Periscope for sharing their knowledge, expertise, and research with a live (and potentially large) audience without the barriers associated with traditional video equipment and standard classroom/conference settings.

Response of mercury in an Adirondack (NY, USA) forest stream to watershed lime application

Science.gov (United States)

Millard, Geoffrey D.; Driscoll, Charles T.; Burns, Douglas; Montesdeoca, Mario R.; Murray, Karen

2018-01-01

Surface waters in Europe and North America previously impacted by acid deposition are recovering in conjunction with declining precursor emissions since the 1980s. Lime has been applied to some impacted watersheds to accelerate recovery. The response to liming can be considered a proxy for future recovery from acid deposition. Increases in dissolved organic carbon concentrations have been observed in surface waters in response to increased pH associated with recovery from acid deposition. Although not previously described, recovery-related increases in dissolved organic carbon could drive increases in mercury concentrations and loads because of the affinity of mercury for dissolved organic matter. We used a before–after impact-response approach to describe the response of stream mercury cycling to the application of lime to the watershed of a small stream in the Adirondack Mountains of New York, USA. Dissolved organic carbon, total mercury and methylmercury concentrations increased

Audio engineering 101 a beginner's guide to music production

CERN Document Server

Dittmar, Tim

2013-01-01

Audio Engineering 101 is a real world guide for starting out in the recording industry. If you have the dream, the ideas, the music and the creativity but don't know where to start, then this book is for you!Filled with practical advice on how to navigate the recording world, from an author with first-hand, real-life experience, Audio Engineering 101 will help you succeed in the exciting, but tough and confusing, music industry. Covering all you need to know about the recording process, from the characteristics of sound to a guide to microphones to analog versus digital

GaN Power Stage for Switch-mode Audio Amplification

DEFF Research Database (Denmark)

Ploug, Rasmus Overgaard; Knott, Arnold; Poulsen, Søren Bang

2015-01-01

Gallium Nitride (GaN) based power transistors are gaining more and more attention since the introduction of the enhancement mode eGaN Field Effect Transistor (FET) which makes an adaptation from Metal-Oxide Semiconductor (MOSFET) to eGaN based technology less complex than by using depletion mode Ga......N FETs. This project seeks to investigate the possibilities of using eGaN FETs as the power switching device in a full bridge power stage intended for switch mode audio amplification. A 50 W 1 MHz power stage was built and provided promising audio performance. Future work includes optimization of dead...

RECURSOS TECNOLÓGICOS AUDIOVISUALES DE FORMACIÓN EN RED: SISTEMAS STREAMING MEDIA Y TELEINMERSIVOS TECHNOLOGICAL RESOURCES FOR ON-LINE INSTRUCTION: STREAMING MEDIA AND TELEIMMERSION SYSTEMS

Directory of Open Access Journals (Sweden)

Juan Antonio Juanes Méndez

2010-06-01

Full Text Available Presentamos dos modalidades tecnológicas de enseñanza en red: la consolidada y ampliamente utilizada tecnología video-streaming, y el futuro de la comunicación a distancia, la teleinmersión. La primera permite la transmisión de audio/vídeo por la red para que puede ser vista por el usuario en su ordenador personal, desde cualquier lugar que disponga de una conexión a red. La información será recibida y decodificada por el usuario final utilizando cualquier reproductor de los que existen en el mercado. La teleinmersión, por su parte, permite crear espacios virtuales de colaboración entre profesionales, ofreciendo entornos muy cercanos a la realidad. Esta tecnología revolucionará, sin duda, nuestros sistemas de enseñanza en los próximos años, facilitando la interacción profesor-alumno. Es evidente que la formación e-learning aporta a los alumnos y a los docentes grandes ventajas como: menores tiempos de aprendizaje, flexibilidad de horarios y de ubicación geográfica, entre otras. We describe two technological modes of on-line teaching: the consolidated and widely used video-streaming mode and teleimmersion, the future of distance communications. The former mode allows the transmission of audio/video through the network so that it can be seen by the user on a PC from anywhere harbouring a network connection. The information is received and decoded by the final user using any reproducer available on the market. Teleimmersion allows the creation of virtual spaces for collaboration among professionals, offering venues that are very similar to reality. This technology will undoubtedly revolutionize our teaching systems in the near future, facilitating instructor-student interaction. It is clear that e-learning- instruction offers both students and instructors huge advantages, such as shorter learning times and schedule and geographic flexibility, among others.

A high PSRR Class-D audio amplifier IC based on a self-adjusting voltage reference

OpenAIRE

Huffenus , Alexandre; Pillonnet , Gaël; Abouchi , Nacer; Goutti , Frédéric; Rabary , Vincent; Cittadini , Robert

2010-01-01

International audience; In a wide range of applications, audio amplifiers require a large Power Supply Rejection Ratio (PSRR) that the current Class-D architecture cannot reach. This paper proposes a self-adjusting internal voltage reference scheme that sets the bias voltages of the amplifier without losing on output dynamics. This solution relaxes the constraints on gain and feedback resistors matching that were previously the limiting factor for the PSRR. Theory of operation, design and IC ...

Audio- and TV-products. Power consumption reduction in audio- and TV-products. Final report; Audio- og TV-produkter. Effektminimering i audio- og TV-produkter: Afsluttende rapport

Energy Technology Data Exchange (ETDEWEB)

Kierkegaard, P.

1998-10-01

The project concerning the audio products resulted in energy savings of 90-97% at efficiencies of 91-96% with full effect and stand-by losses of 0.4-3 W. It is especially new epoch-making methods for pulse modulation (called Controlled Oscillation Modulator, COM and Phase Shifted Carrier Pulse Width Modulation, PSCPWM) and error for correction in the effect conversion (called Multivariable Enhanced Cascade Control, MECC and Pulse Edge Delay Error Correction, PEDEC), which has made the breakthrough. Two patents have been applied for, and new digital amplifiers will be introduced in all the relevant products. The project concerning TV products has shown that a loss reduction in deflecting circuits of ca.20 % may be obtained. (EHS)

Small signal audio design

CERN Document Server

Self, Douglas

2014-01-01

Learn to use inexpensive and readily available parts to obtain state-of-the-art performance in all the vital parameters of noise, distortion, crosstalk and so on. With ample coverage of preamplifiers and mixers and a new chapter on headphone amplifiers, this practical handbook provides an extensive repertoire of circuits that can be put together to make almost any type of audio system.A resource packed full of valuable information, with virtually every page revealing nuggets of specialized knowledge not found elsewhere. Essential points of theory that bear on practical performance are lucidly

Computerized Audio-Visual Instructional Sequences (CAVIS): A Versatile System for Listening Comprehension in Foreign Language Teaching.

Science.gov (United States)

Aleman-Centeno, Josefina R.

1983-01-01

Discusses the development and evaluation of CAVIS, which consists of an Apple microcomputer used with audiovisual dialogs. Includes research on the effects of three conditions: (1) computer with audio and visual, (2) computer with audio alone and (3) audio alone in short-term and long-term recall. (EKN)

Audio-Visual and Meaningful Semantic Context Enhancements in Older and Younger Adults.

Directory of Open Access Journals (Sweden)

Kirsten E Smayda

Full Text Available Speech perception is critical to everyday life. Oftentimes noise can degrade a speech signal; however, because of the cues available to the listener, such as visual and semantic cues, noise rarely prevents conversations from continuing. The interaction of visual and semantic cues in aiding speech perception has been studied in young adults, but the extent to which these two cues interact for older adults has not been studied. To investigate the effect of visual and semantic cues on speech perception in older and younger adults, we recruited forty-five young adults (ages 18-35 and thirty-three older adults (ages 60-90 to participate in a speech perception task. Participants were presented with semantically meaningful and anomalous sentences in audio-only and audio-visual conditions. We hypothesized that young adults would outperform older adults across SNRs, modalities, and semantic contexts. In addition, we hypothesized that both young and older adults would receive a greater benefit from a semantically meaningful context in the audio-visual relative to audio-only modality. We predicted that young adults would receive greater visual benefit in semantically meaningful contexts relative to anomalous contexts. However, we predicted that older adults could receive a greater visual benefit in either semantically meaningful or anomalous contexts. Results suggested that in the most supportive context, that is, semantically meaningful sentences presented in the audiovisual modality, older adults performed similarly to young adults. In addition, both groups received the same amount of visual and meaningful benefit. Lastly, across groups, a semantically meaningful context provided more benefit in the audio-visual modality relative to the audio-only modality, and the presence of visual cues provided more benefit in semantically meaningful contexts relative to anomalous contexts. These results suggest that older adults can perceive speech as well as younger

Publicación de materiales audiovisuales a través de un servidor de video-streaming Publication of audio-visual materials through a streaming video server

Directory of Open Access Journals (Sweden)

Acevedo Clavijo Edwin Jovanny

2010-07-01

Full Text Available Esta propuesta tiene como objetivo estudiar varias alternativas de servidores Streaming para determinar la mejor herramienta para el desarrollo de la publicación de material audiovisual educativo. Se evaluaron las plataformas más utilizadas teniendo en cuenta sus características y beneficios que tiene cada servidor entre las los cuales están: Hélix Universal Server, Windows Media Server de Microsoft, Peer Cast y Darwin Server. implementando un servidor con mayores capacidades y beneficios para la publicación de videos con fines académicos a través de la intranet de la Universidad Cooperativa de Colombia seccional Barrancabermeja This proposal has as an principal objective to study different alternatives for streaming servers to determine the best tool in the project’s development. Platforms most used were evaluated features and benefits in each served such as: Helix Universal Server, Microsoft Windows Media Server, Peer Cast and Darwin Server. Implementing a server with more capabilities and benefits for the publication of videos for academic purposes through the intranet of the Cooperative University of Colombia Barrancabermeja’s sectional

Carrier Distortion in Hysteretic Self-Oscillating Class-D Audio Power:Amplifiers: Analysis and Optimization

OpenAIRE

Høyerby, Mikkel Christian Kofod; Andersen, Michael A. E.

2009-01-01

An important distortion mechanism in hysteretic self-oscillating (SO) class-D (switch mode) power amplifiers-–carrier distortion-–is analyzed and an optimization method is proposed. This mechanism is an issue in any power amplifier application where a high degree of proportionality between input and output is required, such as in audio power amplifiers or xDSL drivers. From an average-mode point of view, carrier distortion is shown to be caused by nonlinear variation of the hysteretic compara...

A Novel Audio Cryptosystem Using Chaotic Maps and DNA Encoding

Directory of Open Access Journals (Sweden)

S. J. Sheela

2017-01-01

Full Text Available Chaotic maps have good potential in security applications due to their inherent characteristics relevant to cryptography. This paper introduces a new audio cryptosystem based on chaotic maps, hybrid chaotic shift transform (HCST, and deoxyribonucleic acid (DNA encoding rules. The scheme uses chaotic maps such as two-dimensional modified Henon map (2D-MHM and standard map. The 2D-MHM which has sophisticated chaotic behavior for an extensive range of control parameters is used to perform HCST. DNA encoding technology is used as an auxiliary tool which enhances the security of the cryptosystem. The performance of the algorithm is evaluated for various speech signals using different encryption/decryption quality metrics. The simulation and comparison results show that the algorithm can achieve good encryption results and is able to resist several cryptographic attacks. The various types of analysis revealed that the algorithm is suitable for narrow band radio communication and real-time speech encryption applications.

Delay in catchment nitrogen load to streams following restrictions on fertilizer application

DEFF Research Database (Denmark)

Vervloet, Lidwien S. C.; Binning, Philip John; Borgesen, Christen D.

2018-01-01

A MIKE SHE hydrological-solute transport model including nitrate reduction is employed to evaluate the delayed response in nitrogen loads in catchment streams following the implementation of nitrogen mitigation measures since the 1980s. The nitrate transport lag times between the root zone...... and the streams for the period 1950-2011 were simulated for two catchments in Denmark and compared with observational data. Results include nitrogen concentration and mass discharge to streams. By automated baseflow separation, stream discharge was separated into baseflow and drain flow components...

Neutron streaming analysis for shield design of FMIT Facility

International Nuclear Information System (INIS)

Carter, L.L.

1980-12-01

Applications of the Monte Carlo method have been summarized relevant to neutron streaming problems of interest in the shield design for the FMIT Facility. An improved angular biasing method has been implemented to further optimize the calculation of streaming and this method has been applied to calculate streaming within a double bend pipe

Potential applications for Flettner rotors and Turbosails in tidal stream turbines

Energy Technology Data Exchange (ETDEWEB)

NONE

2005-07-01

Oreada reports on its studies of two novel lifting devices, namely Flettner Rotors and Turbosails, for application in powering tidal stream generators. Through computer modelling, the power generated by the lift devices has been compared with that of a conventional hydrofoil. The mathematical model assumes the base-case configuration for the turbine to be four parallel lift devices at a constant radius from the centre of the turbine and simulates a vertical axis turbine. Adjacent lift devices subtend an angle of ninety degrees at the centre of the turbine. The theoretical study indicated that the planned second part of the project involving bench tests should not go ahead. The study was largely funded by the DTI.

Methodological considerations in the use of audio diaries in work psychology: Adding to the qualitative toolkit.

Science.gov (United States)

Crozier, Sarah E; Cassell, Catherine M

2016-06-01

The use of longitudinal methodology as a means of capturing the intricacies in complex organizational phenomena is well documented, and many different research strategies for longitudinal designs have been put forward from both a qualitative and quantitative stance. This study explores a specific emergent qualitative methodology, audio diaries, and assesses their utility for work psychology research drawing on the findings from a four-stage study addressing transient working patterns and stress in UK temporary workers. Specifically, we explore some important methodological, analytical and technical issues for practitioners and researchers who seek to use these methods and explain how this type of methodology has much to offer when studying stress and affective experiences at work. We provide support for the need to implement pluralistic and complementary methodological approaches in unearthing the depth in sense-making and assert their capacity to further illuminate the process orientation of stress. This study illustrates the importance of verbalization in documenting stress and affective experience as a mechanism for accessing cognitive processes in making sense of such experience.This study compares audio diaries with more traditional qualitative methods to assess applicability to different research contexts.This study provides practical guidance and a methodological framework for the design of audio diary research and design, taking into account challenges and solutions for researchers and practitioners.

Java parallel secure stream for grid computing

International Nuclear Information System (INIS)

Chen, J.; Akers, W.; Chen, Y.; Watson, W.

2001-01-01

The emergence of high speed wide area networks makes grid computing a reality. However grid applications that need reliable data transfer still have difficulties to achieve optimal TCP performance due to network tuning of TCP window size to improve the bandwidth and to reduce latency on a high speed wide area network. The authors present a pure Java package called JPARSS (Java Parallel Secure Stream) that divides data into partitions that are sent over several parallel Java streams simultaneously and allows Java or Web applications to achieve optimal TCP performance in a gird environment without the necessity of tuning the TCP window size. Several experimental results are provided to show that using parallel stream is more effective than tuning TCP window size. In addition X.509 certificate based single sign-on mechanism and SSL based connection establishment are integrated into this package. Finally a few applications using this package will be discussed

Streams with Strahler Stream Order

Data.gov (United States)

Minnesota Department of Natural Resources — Stream segments with Strahler stream order values assigned. As of 01/08/08 the linework is from the DNR24K stream coverages and will not match the updated...

Transcript of Audio Narrative Portion of: Scandinavian Heritage. A Set of Five Audio-Visual Film Strip/Cassette Presentations.

Science.gov (United States)

Anderson, Gerald D.; Olson, David B.

The document presents the transcript of the audio narrative portion of approximately 100 interviews with first and second generation Scandinavian immigrants to the United States. The document is intended for use by secondary school classroom teachers as they develop and implement educational programs related to the Scandinavian heritage in…

Open-Loop Audio-Visual Stimulation (AVS): A Useful Tool for Management of Insomnia?

Science.gov (United States)

Tang, Hsin-Yi Jean; Riegel, Barbara; McCurry, Susan M; Vitiello, Michael V

2016-03-01

Audio Visual Stimulation (AVS), a form of neurofeedback, is a non-pharmacological intervention that has been used for both performance enhancement and symptom management. We review the history of AVS, its two sub-types (close- and open-loop), and discuss its clinical implications. We also describe a promising new application of AVS to improve sleep, and potentially decrease pain. AVS research can be traced back to the late 1800s. AVS's efficacy has been demonstrated for both performance enhancement and symptom management. Although AVS is commonly used in clinical settings, there is limited literature evaluating clinical outcomes and mechanisms of action. One of the challenges to AVS research is the lack of standardized terms, which makes systematic review and literature consolidation difficult. Future studies using AVS as an intervention should; (1) use operational definitions that are consistent with the existing literature, such as AVS, Audio-visual Entrainment, or Light and Sound Stimulation, (2) provide a clear rationale for the chosen training frequency modality, (3) use a randomized controlled design, and (4) follow the Consolidated Standards of Reporting Trials and/or related guidelines when disseminating results.

The Transfer of Learning Associated with Audio Feedback on Written Work

Directory of Open Access Journals (Sweden)

Tanya Martini

2014-11-01

Full Text Available This study examined whether audio feedback provided to undergraduates (N=51 about one paper would prove beneficial in terms of improving their grades on another, unrelated paper of the same type. We examined this issue both in terms of student beliefs about learning transfer, as well as their actual ability to transfer what had been learned on one assignment to another, subsequent assignment. Results indicated that students believed that they would be able to transfer what they had learned via audio feedback. Moreover, results also suggested that students actually did generalize the overarching comments about content and structure made in the audio files to a subsequent paper, the content of which differed substantially from the initial one. Both students and teaching assistants demonstrated very favourable responses to this type of feedback, suggesting that it was both clear and comprehensive.

Guided Expectations: A Case Study of a Sound Collage Audio Guide

DEFF Research Database (Denmark)

Laursen, Ditte

This paper is a user evaluation of a mobile phone audio guide developed for visitors to use at the National Gallery of Denmark. The audio guide is offered as a downloadable MP3 file to every incoming visitor who is carrying a mobile phone with an open Bluetooth connection. The guide itself...... according to personal interest, and a conflict between the expectation of a learning experience rather than an aesthetic experience. Results indicate that most visitors are able to make sense of the guide and to use it successfully, in different ways, to enrich their visit. Evaluation also shows...... that visitors are fond of using their own mobile phones - but they have several problems with their phones in downloading the MP3 file. Read more: Guided Expectations: A Case Study of a Sound Collage Audio Guide | conference.archimuse.com...

Cambridge English First 2 audio CDs : authentic examination papers

CERN Document Server

2016-01-01

Four authentic Cambridge English Language Assessment examination papers for the Cambridge English: First (FCE) exam. These examination papers for the Cambridge English: First (FCE) exam provide the most authentic exam preparation available, allowing candidates to familiarise themselves with the content and format of the exam and to practise useful exam techniques. The Audio CDs contain the recorded material to allow thorough preparation for the Listening paper and are designed to be used with the Student's Book. A Student's Book with or without answers and a Student's Book with answers and downloadable Audio are available separately. These tests are also available as Cambridge English: First Tests 5-8 on Testbank.org.uk

Audio-visual materials usage preference among agricultural ...

African Journals Online (AJOL)

It was found that respondents preferred radio, television, poster, advert, photographs, specimen, bulletin, magazine, cinema, videotape, chalkboard, and bulletin board as audio-visual materials for extension work. These are the materials that can easily be manipulated and utilized for extension work. Nigerian Journal of ...

An Analog I/O Interface Board for Audio Arduino Open Sound Card System

DEFF Research Database (Denmark)

Dimitrov, Smilen; Serafin, Stefania

2011-01-01

AudioArduino [1] is a system consisting of an ALSA (Advanced Linux Sound Architecture) audio driver and corresponding microcontroller code; that can demonstrate full-duplex, mono, 8-bit, 44.1 kHz soundcard behavior on an FTDI based Arduino. While the basic operation as a soundcard can...

Minimizing Crosstalk in Self Oscillating Switch Mode Audio Power Amplifiers

DEFF Research Database (Denmark)

Knott, Arnold; Ploug, Rasmus Overgaard

2012-01-01

a method to minimize this phenomenon by improving the integrity of the various power distribution systems of the amplifier. The method is then applied to an amplifier built for this investigation. The results show that the crosstalk is suppressed with 30 dB, but is not entirely eliminated......The varying switching frequencies of self oscillating switch mode audio amplifiers have been known to cause interchannel intermodulation disturbances in multi channel configurations. This crosstalk phenomenon has a negative impact on the audio performance. The goal of this paper is to present...

Multilevel tracking power supply for switch-mode audio power amplifiers

DEFF Research Database (Denmark)

Iversen, Niels Elkjær; Lazarevic, Vladan; Vasic, Miroslav

2018-01-01

to the power supply in order to improve efficiency. A 100 W prototype system was designed. Measured results show that systems employing envelope tracking can improve system efficiency from 2% to 12%, i.e. a factor of 6. The temperature rise is strongly reduced, especially for the switching power MOSFETs where......Switch-mode technology is the common choice for high efficiency audio power amplifiers. The dynamic nature of real audio reduces efficiency as less continuous output power can be achieved. Based on methods used for RF amplifiers this paper proposes to employ envelope tracking techniques...

Convolution-based classification of audio and symbolic representations of music

DEFF Research Database (Denmark)

Velarde, Gissel; Cancino Chacón, Carlos; Meredith, David

2018-01-01

We present a novel convolution-based method for classification of audio and symbolic representations of music, which we apply to classification of music by style. Pieces of music are first sampled to pitch–time representations (piano-rolls or spectrograms) and then convolved with a Gaussian filter......-class composer identification, methods specialised for classifying symbolic representations of music are more effective. We also performed experiments on symbolic representations, synthetic audio and two different recordings of The Well-Tempered Clavier by J. S. Bach to study the method’s capacity to distinguish...

Probabilistic Graphical Models for the Analysis and Synthesis of Musical Audio

Science.gov (United States)

Hoffmann, Matthew Douglas

Content-based Music Information Retrieval (MIR) systems seek to automatically extract meaningful information from musical audio signals. This thesis applies new and existing generative probabilistic models to several content-based MIR tasks: timbral similarity estimation, semantic annotation and retrieval, and latent source discovery and separation. In order to estimate how similar two songs sound to one another, we employ a Hierarchical Dirichlet Process (HDP) mixture model to discover a shared representation of the distribution of timbres in each song. Comparing songs under this shared representation yields better query-by-example retrieval quality and scalability than previous approaches. To predict what tags are likely to apply to a song (e.g., "rap," "happy," or "driving music"), we develop the Codeword Bernoulli Average (CBA) model, a simple and fast mixture-of-experts model. Despite its simplicity, CBA performs at least as well as state-of-the-art approaches at automatically annotating songs and finding to what songs in a database a given tag most applies. Finally, we address the problem of latent source discovery and separation by developing two Bayesian nonparametric models, the Shift-Invariant HDP and Gamma Process NMF. These models allow us to discover what sounds (e.g. bass drums, guitar chords, etc.) are present in a song or set of songs and to isolate or suppress individual source. These models' ability to decide how many latent sources are necessary to model the data is particularly valuable in this application, since it is impossible to guess a priori how many sounds will appear in a given song or set of songs. Once they have been fit to data, probabilistic models can also be used to drive the synthesis of new musical audio, both for creative purposes and to qualitatively diagnose what information a model does and does not capture. We also adapt the SIHDP model to create new versions of input audio with arbitrary sample sets, for example, to create

The audio and visual communication systems for suited engineering activities on JET

International Nuclear Information System (INIS)

Pearce, R.J.H.; Bruce, J.; Callaghan, C.; Hart, M.; Martin, P.; Middleton, R.; Tait, J.

2001-01-01

The beryllium and/or tritium contamination of the JET tokamak and auxiliary systems necessitates that many activities are carried out in air line fed pressurised suits. To enable often complex engineering activities to be performed, a number of novel audio and visual and communications systems have been designed. The paper describes these systems which give freedom of visual and audio communication between suited personnel, supervisors, operators and engineers. The system enhances the safety of the working environment as well as helping to minimise the radiation dose to personnel. It is concluded, from a number of years experience of using the audio and visual communications systems for suited operations, that safety and the progress of complex engineering tasks have been significantly enhanced

The audio and visual communication systems for suited engineering activities on JET

Energy Technology Data Exchange (ETDEWEB)

Pearce, R.J.H. E-mail: robert.pearce@jet.uk; Bruce, J.; Callaghan, C.; Hart, M.; Martin, P.; Middleton, R.; Tait, J

2001-11-01

The beryllium and/or tritium contamination of the JET tokamak and auxiliary systems necessitates that many activities are carried out in air line fed pressurised suits. To enable often complex engineering activities to be performed, a number of novel audio and visual and communications systems have been designed. The paper describes these systems which give freedom of visual and audio communication between suited personnel, supervisors, operators and engineers. The system enhances the safety of the working environment as well as helping to minimise the radiation dose to personnel. It is concluded, from a number of years experience of using the audio and visual communications systems for suited operations, that safety and the progress of complex engineering tasks have been significantly enhanced.

Procedural Audio in Computer Games Using Motion Controllers: An Evaluation on the Effect and Perception

Directory of Open Access Journals (Sweden)

Niels Böttcher

2013-01-01

Full Text Available A study has been conducted into whether the use of procedural audio affects players in computer games using motion controllers. It was investigated whether or not (1 players perceive a difference between detailed and interactive procedural audio and prerecorded audio, (2 the use of procedural audio affects their motor-behavior, and (3 procedural audio affects their perception of control. Three experimental surveys were devised, two consisting of game sessions and the third consisting of watching videos of gameplay. A skiing game controlled by a Nintendo Wii balance board and a sword-fighting game controlled by a Wii remote were implemented with two versions of sound, one sample based and the other procedural based. The procedural models were designed using a perceptual approach and by alternative combinations of well-known synthesis techniques. The experimental results showed that, when being actively involved in playing or purely observing a video recording of a game, the majority of participants did not notice any difference in sound. Additionally, it was not possible to show that the use of procedural audio caused any consistent change in the motor behavior. In the skiing experiment, a portion of players perceived the control of the procedural version as being more sensitive.

Agency Video, Audio and Imagery Library