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...

ESA personal communications and digital audio broadcasting systems based on non-geostationary satellites

Science.gov (United States)

Logalbo, P.; Benedicto, J.; Viola, R.

1993-01-01

Personal Communications and Digital Audio Broadcasting are two new services that the European Space Agency (ESA) is investigating for future European and Global Mobile Satellite systems. ESA is active in promoting these services in their various mission options including non-geostationary and geostationary satellite systems. A Medium Altitude Global Satellite System (MAGSS) for global personal communications at L and S-band, and a Multiregional Highly inclined Elliptical Orbit (M-HEO) system for multiregional digital audio broadcasting at L-band are described. Both systems are being investigated by ESA in the context of future programs, such as Archimedes, which are intended to demonstrate the new services and to develop the technology for future non-geostationary mobile communication and broadcasting satellites.

Understanding the Effect of Audio Communication Delay on Distributed Team Interaction

Science.gov (United States)

2013-06-01

means for members to socialize and learn about each other, engenders development cooperative relationships, and lays a foundation for future interaction...length will result in increases in task completion time and mental workload. 3. Audiovisual technology will moderate the effect of communication...than audio alone. 4. Audiovisual technology will moderate the effect of communication delays such that task completion time and mental workload will

Amplitude Modulated Sinusoidal Signal Decomposition for Audio Coding

DEFF Research Database (Denmark)

Christensen, M. G.; Jacobson, A.; Andersen, S. V.

2006-01-01

In this paper, we present a decomposition for sinusoidal coding of audio, based on an amplitude modulation of sinusoids via a linear combination of arbitrary basis vectors. The proposed method, which incorporates a perceptual distortion measure, is based on a relaxation of a nonlinear least......-squares minimization. Rate-distortion curves and listening tests show that, compared to a constant-amplitude sinusoidal coder, the proposed decomposition offers perceptually significant improvements in critical transient signals....

Subband coding of digital audio signals without loss of quality

NARCIS (Netherlands)

Veldhuis, Raymond N.J.; Breeuwer, Marcel; van de Waal, Robbert

1989-01-01

A subband coding system for high quality digital audio signals is described. To achieve low bit rates at a high quality level, it exploits the simultaneous masking effect of the human ear. It is shown how this effect can be used in an adaptive bit-allocation scheme. The proposed approach has been

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…

Recursive nearest neighbor search in a sparse and multiscale domain for comparing audio signals

DEFF Research Database (Denmark)

Sturm, Bob L.; Daudet, Laurent

2011-01-01

We investigate recursive nearest neighbor search in a sparse domain at the scale of audio signals. Essentially, to approximate the cosine distance between the signals we make pairwise comparisons between the elements of localized sparse models built from large and redundant multiscale dictionaries...

Joint evaluation of communication quality and user experience in an audio-visual virtual reality meeting

DEFF Research Database (Denmark)

Møller, Anders Kalsgaard; Hoffmann, Pablo F.; Carrozzino, Marcello

2013-01-01

The state-of-the-art speech intelligibility tests are created with the purpose of evaluating acoustic communication devices and not for evaluating audio-visual virtual reality systems. This paper present a novel method to evaluate a communication situation based on both the speech intelligibility...

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.

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,

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...

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.

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

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.

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...

Measuring 3D Audio Localization Performance and Speech Quality of Conferencing Calls for a Multiparty Communication System

Directory of Open Access Journals (Sweden)

Mansoor Hyder

2013-07-01

Full Text Available Communication systems which support 3D (Three Dimensional audio offer a couple of advantages to the users/customers. Firstly, within the virtual acoustic environments all participants could easily be recognized through their placement/sitting positions. Secondly, all participants can turn their focus on any particular talker when multiple participants start talking at the same time by taking advantage of the natural listening tendency which is called the Cocktail Party Effect. On the other hand, 3D audio is known as a decreasing factor for overall speech quality because of the commencement of reverberations and echoes within the listening environment. In this article, we study the tradeoff between speech quality and human natural ability of localizing audio events/or talkers within our three dimensional audio supported telephony and teleconferencing solution. Further, we performed subjective user studies by incorporating two different HRTFs (Head Related Transfer Functions, different placements of the teleconferencing participants and different layouts of the virtual environments. Moreover, subjective user studies results for audio event localization and subjective speech quality are presented in this article. This subjective user study would help the research community to optimize the existing 3D audio systems and to design new 3D audio supported teleconferencing solutions based on the quality of experience requirements of the users/customers for agriculture personal in particular and for all potential users in general.

Measuring 3D Audio Localization Performance and Speech Quality of Conferencing Calls for a Multiparty Communication System

International Nuclear Information System (INIS)

Hyder, M.; Menghwar, G.D.; Qureshi, A.

2013-01-01

Communication systems which support 3D (Three Dimensional) audio offer a couple of advantages to the users/customers. Firstly, within the virtual acoustic environments all participants could easily be recognized through their placement/sitting positions. Secondly, all participants can turn their focus on any particular talker when multiple participants start talking at the same time by taking advantage of the natural listening tendency which is called the Cocktail Party Effect. On the other hand, 3D audio is known as a decreasing factor for overall speech quality because of the commencement of reverberations and echoes within the listening environment. In this article, we study the tradeoff between speech quality and human natural ability of localizing audio events/or talkers within our three dimensional audio supported telephony and teleconferencing solution. Further, we performed subjective user studies by incorporating two different HRTFs (Head Related Transfer Functions), different placements of the teleconferencing participants and different layouts of the virtual environments. Moreover, subjective user studies results for audio event localization and subjective speech quality are presented in this article. This subjective user study would help the research community to optimize the existing 3D audio systems and to design new 3D audio supported teleconferencing solutions based on the quality of experience requirements of the users/customers for agriculture personal in particular and for all potential users in general. (author)

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...

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.

Digital communication device

DEFF Research Database (Denmark)

2005-01-01

The invention concerns a digital communication device like a hearing aid or a headset. The hearing aid or headset has a power supply, a signal processing device, means for receiving a wireless signal and a receiver or loudspeaker, which produces an audio signal based on a modulated pulsed signal...... point is provided which is in electrical contact with the metal of the metal box and whereby this third connection point is connected to the electric circuitry of the communication device at a point having a stable and well defined electrical potential. In this way the electro-and magnetic radiation...

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…

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

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...

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

Analysis of musical expression in audio signals

Science.gov (United States)

Dixon, Simon

2003-01-01

In western art music, composers communicate their work to performers via a standard notation which specificies the musical pitches and relative timings of notes. This notation may also include some higher level information such as variations in the dynamics, tempo and timing. Famous performers are characterised by their expressive interpretation, the ability to convey structural and emotive information within the given framework. The majority of work on audio content analysis focusses on retrieving score-level information; this paper reports on the extraction of parameters describing the performance, a task which requires a much higher degree of accuracy. Two systems are presented: BeatRoot, an off-line beat tracking system which finds the times of musical beats and tracks changes in tempo throughout a performance, and the Performance Worm, a system which provides a real-time visualisation of the two most important expressive dimensions, tempo and dynamics. Both of these systems are being used to process data for a large-scale study of musical expression in classical and romantic piano performance, which uses artificial intelligence (machine learning) techniques to discover fundamental patterns or principles governing expressive performance.

Detection and Correction of Under-/Overexposed Optical Soundtracks by Coupling Image and Audio Signal Processing

Directory of Open Access Journals (Sweden)

Etienne Decenciere

2008-10-01

Full Text Available Film restoration using image processing, has been an active research field during the last years. However, the restoration of the soundtrack has been mainly performed in the sound domain, using signal processing methods, despite the fact that it is recorded as a continuous image between the images of the film and the perforations. While the very few published approaches focus on removing dust particles or concealing larger corrupted areas, no published works are devoted to the restoration of soundtracks degraded by substantial underexposure or overexposure. Digital restoration of optical soundtracks is an unexploited application field and, besides, scientifically rich, because it allows mixing both image and signal processing approaches. After introducing the principles of optical soundtrack recording and playback, this contribution focuses on our first approaches to detect and cancel the effects of under and overexposure. We intentionally choose to get a quantification of the effect of bad exposure in the 1D audio signal domain instead of 2D image domain. Our measurement is sent as feedback value to an image processing stage where the correction takes place, building up a “digital image and audio signal” closed loop processing. The approach is validated on both simulated alterations and real data.

Noise-Canceling Helmet Audio System

Science.gov (United States)

Seibert, Marc A.; Culotta, Anthony J.

2007-01-01

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

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)

The presentation of expert testimony via live audio-visual communication.

Science.gov (United States)

Miller, R D

1991-01-01

As part of a national effort to improve efficiency in court procedures, the American Bar Association has recommended, on the basis of a number of pilot studies, increased use of current audio-visual technology, such as telephone and live video communication, to eliminate delays caused by unavailability of participants in both civil and criminal procedures. Although these recommendations were made to facilitate court proceedings, and for the convenience of attorneys and judges, they also have the potential to save significant time for clinical expert witnesses as well. The author reviews the studies of telephone testimony that were done by the American Bar Association and other legal research groups, as well as the experience in one state forensic evaluation and treatment center. He also reviewed the case law on the issue of remote testimony. He then presents data from a national survey of state attorneys general concerning the admissibility of testimony via audio-visual means, including video depositions. Finally, he concludes that the option to testify by telephone provides a significant savings in precious clinical time for forensic clinicians in public facilities, and urges that such clinicians work actively to convince courts and/or legislatures in states that do not permit such testimony (currently the majority), to consider accepting it, to improve the effective use of scarce clinical resources in public facilities.

Novel communication scheme based on chaotic Roessler circuits

International Nuclear Information System (INIS)

GarcIa-Lopez, J H; Jaimes-Reategui, R; Pisarchik, A N; MurguIa-Hernandez, A; Medina-Gutierrez, C; Valdivia-Hernadez, R; Villafana-Rauda, E

2005-01-01

We present a novel synchronization scheme for secure communication with two chaotic unidirectionally coupled Roessler circuits. The circuits are synchronized via one of the variables, while a signal is transmitted through another variable. We show that this scheme allows more stable communications. The system dynamics is studied numerically and experimentally in a wide range of a control parameter. The possibility of secure communications with an audio signal is demonstrated

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

Interactive video audio system: communication server for INDECT portal

Science.gov (United States)

Mikulec, Martin; Voznak, Miroslav; Safarik, Jakub; Partila, Pavol; Rozhon, Jan; Mehic, Miralem

2014-05-01

The paper deals with presentation of the IVAS system within the 7FP EU INDECT project. The INDECT project aims at developing the tools for enhancing the security of citizens and protecting the confidentiality of recorded and stored information. It is a part of the Seventh Framework Programme of European Union. We participate in INDECT portal and the Interactive Video Audio System (IVAS). This IVAS system provides a communication gateway between police officers working in dispatching centre and police officers in terrain. The officers in dispatching centre have capabilities to obtain information about all online police officers in terrain, they can command officers in terrain via text messages, voice or video calls and they are able to manage multimedia files from CCTV cameras or other sources, which can be interesting for officers in terrain. The police officers in terrain are equipped by smartphones or tablets. Besides common communication, they can reach pictures or videos sent by commander in office and they can respond to the command via text or multimedia messages taken by their devices. Our IVAS system is unique because we are developing it according to the special requirements from the Police of the Czech Republic. The IVAS communication system is designed to use modern Voice over Internet Protocol (VoIP) services. The whole solution is based on open source software including linux and android operating systems. The technical details of our solution are presented in the paper.

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

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.

Digital signal processing

CERN Document Server

O'Shea, Peter; Hussain, Zahir M

2011-01-01

In three parts, this book contributes to the advancement of engineering education and that serves as a general reference on digital signal processing. Part I presents the basics of analog and digital signals and systems in the time and frequency domain. It covers the core topics: convolution, transforms, filters, and random signal analysis. It also treats important applications including signal detection in noise, radar range estimation for airborne targets, binary communication systems, channel estimation, banking and financial applications, and audio effects production. Part II considers sel

Mobile video-to-audio transducer and motion detection for sensory substitution

Directory of Open Access Journals (Sweden)

Maxime eAmbard

2015-10-01

Full Text Available Visuo-auditory sensory substitution systems are augmented reality devices that translate a video stream into an audio stream in order to help the blind in daily tasks requiring visuo-spatial information. In this work, we present both a new mobile device and a transcoding method specifically designed to sonify moving objects. Frame differencing is used to extract spatial features from the video stream and two-dimensional spatial information is converted into audio cues using pitch, interaural time difference and interaural level difference. Using numerical methods, we attempt to reconstruct visuo-spatial information based on audio signals generated from various video stimuli. We show that despite a contrasted visual background and a highly lossy encoding method, the information in the audio signal is sufficient to allow object localization, object trajectory evaluation, object approach detection, and spatial separation of multiple objects. We also show that this type of audio signal can be interpreted by human users by asking ten subjects to discriminate trajectories based on generated audio signals.

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.

WLAN Technologies for Audio Delivery

Directory of Open Access Journals (Sweden)

Nicolas-Alexander Tatlas

2007-01-01

Full Text Available Audio delivery and reproduction for home or professional applications may greatly benefit from the adoption of digital wireless local area network (WLAN technologies. The most challenging aspect of such integration relates the synchronized and robust real-time streaming of multiple audio channels to multipoint receivers, for example, wireless active speakers. Here, it is shown that current WLAN solutions are susceptible to transmission errors. A detailed study of the IEEE802.11e protocol (currently under ratification is also presented and all relevant distortions are assessed via an analytical and experimental methodology. A novel synchronization scheme is also introduced, allowing optimized playback for multiple receivers. The perceptual audio performance is assessed for both stereo and 5-channel applications based on either PCM or compressed audio signals.

Communication with diode laser: short distance line of sight communication using fiber optics

International Nuclear Information System (INIS)

Mirza, A.H.

1999-01-01

The objective of this project is to carry audio signal from transmitting station to a short distance receiving station along line of sight and also communication through fiber optics is performed, using diode laser light as carrier. In this project optical communication system, modulation techniques, basics of laser and causes of using diode laser are discussed briefly. Transmitter circuit and receiver circuit are fully described. Communication was performed using pulse width modulation technique. Optical fiber communication have many advantages over other type of conventional communication techniques. This report contains the description of optical fiber communication and compared with other communication systems. (author)

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.

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 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...

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.

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...

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

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

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

Chaotic signals in digital communications

CERN Document Server

Eisencraft, Marcio; Suyama, Ricardo

2013-01-01

Chaotic Signals in Digital Communications combines fundamental background knowledge with state-of-the-art methods for using chaotic signals and systems in digital communications. The book builds a bridge between theoretical works and practical implementation to help researchers attain consistent performance in realistic environments. It shows the possible shortcomings of the chaos-based communication systems proposed in the literature, particularly when they are subjected to non-ideal conditions. It also presents a toolbox of techniques for researchers working to actually implement such system

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.

Interactive Teaching of Adaptive Signal Processing

OpenAIRE

Stewart, R W; Harteneck, M; Weiss, S

2000-01-01

Over the last 30 years adaptive digital signal processing has progressed from being a strictly graduate level advanced class in signal processing theory to a topic that is part of the core curriculum for many undergraduate signal processing classes. The key reason is the continued advance of communications technology, with its need for echo control and equalisation, and the widespread use of adaptive filters in audio, biomedical, and control applications. In this paper we will review the basi...

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 ...

Communication Signals in Lizards.

Science.gov (United States)

Carpenter, Charles C.

1983-01-01

Discusses mechanisms and functional intent of visual communication signals in iguanid/agamid lizards. Demonstrated that lizards communicate with each other by using pushups and head nods and that each species does this in its own way, conveying different types of information. (JN)

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...

Calcium signal communication in the central nervous system.

Science.gov (United States)

Braet, Katleen; Cabooter, Liesbet; Paemeleire, Koen; Leybaert, Luc

2004-02-01

The communication of calcium signals between cells is known to be operative between neurons where these signals integrate intimately with electrical and chemical signal communication at synapses. Recently, it has become clear that glial cells also exchange calcium signals between each other in cultures and in brain slices. This communication pathway has received utmost attention since it is known that astrocytic calcium signals can be induced by neuronal stimulation and can be communicated back to the neurons to modulate synaptic transmission. In addition to this, cells that are generally not considered as brain cells become progressively incorporated in the picture, as astrocytic calcium signals are reported to be communicated to endothelial cells of the vessel wall and can affect smooth muscle cell tone to influence the vessel diameter and thus blood flow. We review the available evidence for calcium signal communication in the central nervous system, taking into account a basic functional unit -the brain cell tripartite- consisting of neurons, glial cells and vascular cells and with emphasis on glial-vascular calcium signaling aspects.

Underdetermined Blind Audio Source Separation Using Modal Decomposition

Directory of Open Access Journals (Sweden)

Abdeldjalil Aïssa-El-Bey

2007-03-01

Full Text Available This paper introduces new algorithms for the blind separation of audio sources using modal decomposition. Indeed, audio signals and, in particular, musical signals can be well approximated by a sum of damped sinusoidal (modal components. Based on this representation, we propose a two-step approach consisting of a signal analysis (extraction of the modal components followed by a signal synthesis (grouping of the components belonging to the same source using vector clustering. For the signal analysis, two existing algorithms are considered and compared: namely the EMD (empirical mode decomposition algorithm and a parametric estimation algorithm using ESPRIT technique. A major advantage of the proposed method resides in its validity for both instantaneous and convolutive mixtures and its ability to separate more sources than sensors. Simulation results are given to compare and assess the performance of the proposed algorithms.

Underdetermined Blind Audio Source Separation Using Modal Decomposition

Directory of Open Access Journals (Sweden)

Aïssa-El-Bey Abdeldjalil

2007-01-01

Full Text Available This paper introduces new algorithms for the blind separation of audio sources using modal decomposition. Indeed, audio signals and, in particular, musical signals can be well approximated by a sum of damped sinusoidal (modal components. Based on this representation, we propose a two-step approach consisting of a signal analysis (extraction of the modal components followed by a signal synthesis (grouping of the components belonging to the same source using vector clustering. For the signal analysis, two existing algorithms are considered and compared: namely the EMD (empirical mode decomposition algorithm and a parametric estimation algorithm using ESPRIT technique. A major advantage of the proposed method resides in its validity for both instantaneous and convolutive mixtures and its ability to separate more sources than sensors. Simulation results are given to compare and assess the performance of the proposed algorithms.

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.

An Analysis/Synthesis System of Audio Signal with Utilization of an SN Model

Directory of Open Access Journals (Sweden)

G. Rozinaj

2004-12-01

Full Text Available An SN (sinusoids plus noise model is a spectral model, in which theperiodic components of the sound are represented by sinusoids withtime-varying frequencies, amplitudes and phases. The remainingnon-periodic components are represented by a filtered noise. Thesinusoidal model utilizes physical properties of musical instrumentsand the noise model utilizes the human inability to perceive the exactspectral shape or the phase of stochastic signals. SN modeling can beapplied in a compression, transformation, separation of sounds, etc.The designed system is based on methods used in the SN modeling. Wehave proposed a model that achieves good results in audio perception.Although many systems do not save phases of the sinusoids, they areimportant for better modelling of transients, for the computation ofresidual and last but not least for stereo signals, too. One of thefundamental properties of the proposed system is the ability of thesignal reconstruction not only from the amplitude but from the phasepoint of view, as well.

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...

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...

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...

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

Science.gov (United States)

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

2014-02-01

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

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.

Semantic Labeling of Nonspeech Audio Clips

Directory of Open Access Journals (Sweden)

Xiaojuan Ma

2010-01-01

Full Text Available Human communication about entities and events is primarily linguistic in nature. While visual representations of information are shown to be highly effective as well, relatively little is known about the communicative power of auditory nonlinguistic representations. We created a collection of short nonlinguistic auditory clips encoding familiar human activities, objects, animals, natural phenomena, machinery, and social scenes. We presented these sounds to a broad spectrum of anonymous human workers using Amazon Mechanical Turk and collected verbal sound labels. We analyzed the human labels in terms of their lexical and semantic properties to ascertain that the audio clips do evoke the information suggested by their pre-defined captions. We then measured the agreement with the semantically compatible labels for each sound clip. Finally, we examined which kinds of entities and events, when captured by nonlinguistic acoustic clips, appear to be well-suited to elicit information for communication, and which ones are less discriminable. Our work is set against the broader goal of creating resources that facilitate communication for people with some types of language loss. Furthermore, our data should prove useful for future research in machine analysis/synthesis of audio, such as computational auditory scene analysis, and annotating/querying large collections of sound effects.

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

A New Principle for a High Efficiency Power Audio Amplifier for Use with a Digital Preamplifier

DEFF Research Database (Denmark)

Jensen, Jørgen Arendt

1986-01-01

The use of class-B and class-D amlifiers for converting digital audio signals to analog signals is discussed. It is shown that the class-D amplifier is unsuitable due to distortion. Therefore, a new principle involving a switch-mode power supply and a class-B amplifier is suggested. By regulating...... the supply voltage to the amplifier according to the amplitude of the audio signal, a higher efficiency than can be obtained by the current principles is achieved. The regulation can be done very efficiently by generating the control signal to the power supply in advance of the audio signal, made possible...

A new principle for a high-efficiency power audio amplifier for use with a digital preamplifier

DEFF Research Database (Denmark)

Jensen, Jørgen Arendt

1987-01-01

The use of class-B and class-D amplifiers for converting digital audio signals to analog signals is discussed. It is shown that the class-D amplifier is unsuitable due to distortion. Therefore a new principle involving a switch-mode power supply and a class-B amplifier is suggested. By regulating...... the supply voltage to the amplifier according to the amplitude of the audio signal, a higher efficiency than can be obtained by the usual principles is achieved. The regulation can be done very efficiently by generating the control signal to the power supply in advance of the audio signal, made possible...

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.

Estimation of the energy ratio between primary and ambience components in stereo audio data

NARCIS (Netherlands)

Harma, A.S.

2011-01-01

Stereo audio signal is often modeled as a mixture of instantaneously mixed primary components and uncorrelated ambience components. This paper focuses on the estimation of the primary-to-ambience energy ratio, PAR. This measure is useful for signal decomposition in stereo and multichannel audio

A High Performance Approach to Minimizing Interactions between Inbound and Outbound Signals in Helmet, Phase I

Data.gov (United States)

National Aeronautics and Space Administration — We propose a high performance approach to enhancing communications between astronauts. In the new generation of NASA audio systems for astronauts, inbound signals...

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

Real-time Loudspeaker Distance Estimation with Stereo Audio

DEFF Research Database (Denmark)

Nielsen, Jesper Kjær; Gaubitch, Nikolay; Heusdens, Richard

2015-01-01

Knowledge on how a number of loudspeakers are positioned relative to a listening position can be used to enhance the listening experience. Usually, these loudspeaker positions are estimated using calibration signals, either audible or psycho-acoustically hidden inside the desired audio signal...

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.

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...

Cepstral domain modification of audio signals for data embedding: preliminary results

Science.gov (United States)

Gopalan, Kaliappan

2004-06-01

A method of embedding data in an audio signal using cepstral domain modification is described. Based on successful embedding in the spectral points of perceptually masked regions in each frame of speech, first the technique was extended to embedding in the log spectral domain. This extension resulted at approximately 62 bits /s of embedding with less than 2 percent of bit error rate (BER) for a clean cover speech (from the TIMIT database), and about 2.5 percent for a noisy speech (from an air traffic controller database), when all frames - including silence and transition between voiced and unvoiced segments - were used. Bit error rate increased significantly when the log spectrum in the vicinity of a formant was modified. In the next procedure, embedding by altering the mean cepstral values of two ranges of indices was studied. Tests on both a noisy utterance and a clean utterance indicated barely noticeable perceptual change in speech quality when lower range of cepstral indices - corresponding to vocal tract region - was modified in accordance with data. With an embedding capacity of approximately 62 bits/s - using one bit per each frame regardless of frame energy or type of speech - initial results showed a BER of less than 1.5 percent for a payload capacity of 208 embedded bits using the clean cover speech. BER of less than 1.3 percent resulted for the noisy host with a capacity was 316 bits. When the cepstrum was modified in the region of excitation, BER increased to over 10 percent. With quantization causing no significant problem, the technique warrants further studies with different cepstral ranges and sizes. Pitch-synchronous cepstrum modification, for example, may be more robust to attacks. In addition, cepstrum modification in regions of speech that are perceptually masked - analogous to embedding in frequency masked regions - may yield imperceptible stego audio with low BER.

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...

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…

Automatic modulation recognition of communication signals

CERN Document Server

Azzouz, Elsayed Elsayed

1996-01-01

Automatic modulation recognition is a rapidly evolving area of signal analysis. In recent years, interest from the academic and military research institutes has focused around the research and development of modulation recognition algorithms. Any communication intelligence (COMINT) system comprises three main blocks: receiver front-end, modulation recogniser and output stage. Considerable work has been done in the area of receiver front-ends. The work at the output stage is concerned with information extraction, recording and exploitation and begins with signal demodulation, that requires accurate knowledge about the signal modulation type. There are, however, two main reasons for knowing the current modulation type of a signal; to preserve the signal information content and to decide upon the suitable counter action, such as jamming. Automatic Modulation Recognition of Communications Signals describes in depth this modulation recognition process. Drawing on several years of research, the authors provide a cr...

Active Electromagnetic Interference Cancelation for Automotive Switch-Mode Audio Power Amplifiers

DEFF Research Database (Denmark)

Knott, Arnold; Pfaffinger, Gerhard; Andersen, Michael A. E.

2009-01-01

Recent trends in the automotive audio industry have shown the importance of active noise cancelation (ANC) for major improvements in mobile entertainment environments. These approaches target the acoustical noise in the cabin and superimpose an inverse noise signal to cancel disturbances. Electro......Recent trends in the automotive audio industry have shown the importance of active noise cancelation (ANC) for major improvements in mobile entertainment environments. These approaches target the acoustical noise in the cabin and superimpose an inverse noise signal to cancel disturbances...

Concurrent signal combining and channel estimation in digital communications

Science.gov (United States)

Ormesher, Richard C [Albuquerque, NM; Mason, John J [Albuquerque, NM

2011-08-30

In the reception of digital information transmitted on a communication channel, a characteristic exhibited by the communication channel during transmission of the digital information is estimated based on a communication signal that represents the digital information and has been received via the communication channel. Concurrently with the estimating, the communication signal is used to decide what digital information was transmitted.

Signals in Communication Engineering History

Science.gov (United States)

Consonni, Denise; Silva, Magno T. M.

2010-01-01

This paper is a study of various electric signals, which have been employed throughout the history of communication engineering in its two main landmarks: the telegraph and the telephone. The signals are presented in their time and frequency domain representations. The historical order has been followed in the presentation: wired systems, spark…

Advances in preventive monitoring of machinery through audio and vibration signals

OpenAIRE

Henríquez Rodríguez, Patricia

2016-01-01

Programa de doctorado: Sistemas Inteligentes y Aplicaciones Numéricas en Ingeniería. La fecha de publicación es la fecha de lectura. El objetivo de la presente Tesis es la mejora de los sistemas de monitorización de maquinaria en diagnóstico e identificación de fallo usando señales de vibración y de audio en dos aplicaciones (cojinetes y bombas centrífugas) con especial énfasis en la etapa de extracción de características y en la utilización del audio como fuente de información. En el caso...

Silent communication: toward using brain signals.

Science.gov (United States)

Pei, Xiaomei; Hill, Jeremy; Schalk, Gerwin

2012-01-01

From the 1980s movie Firefox to the more recent Avatar, popular science fiction has speculated about the possibility of a persons thoughts being read directly from his or her brain. Such braincomputer interfaces (BCIs) might allow people who are paralyzed to communicate with and control their environment, and there might also be applications in military situations wherever silent user-to-user communication is desirable. Previous studies have shown that BCI systems can use brain signals related to movements and movement imagery or attention-based character selection. Although these systems have successfully demonstrated the possibility to control devices using brain function, directly inferring which word a person intends to communicate has been elusive. A BCI using imagined speech might provide such a practical, intuitive device. Toward this goal, our studies to date addressed two scientific questions: (1) Can brain signals accurately characterize different aspects of speech? (2) Is it possible to predict spoken or imagined words or their components using brain signals?

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 ...

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...

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.

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.

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

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.

Design and Implementation of a Video-Zoom Driven Digital Audio-Zoom System for Portable Digital Imaging Devices

Science.gov (United States)

Park, Nam In; Kim, Seon Man; Kim, Hong Kook; Kim, Ji Woon; Kim, Myeong Bo; Yun, Su Won

In this paper, we propose a video-zoom driven audio-zoom algorithm in order to provide audio zooming effects in accordance with the degree of video-zoom. The proposed algorithm is designed based on a super-directive beamformer operating with a 4-channel microphone system, in conjunction with a soft masking process that considers the phase differences between microphones. Thus, the audio-zoom processed signal is obtained by multiplying an audio gain derived from a video-zoom level by the masked signal. After all, a real-time audio-zoom system is implemented on an ARM-CORETEX-A8 having a clock speed of 600 MHz after different levels of optimization are performed such as algorithmic level, C-code, and memory optimizations. To evaluate the complexity of the proposed real-time audio-zoom system, test data whose length is 21.3 seconds long is sampled at 48 kHz. As a result, it is shown from the experiments that the processing time for the proposed audio-zoom system occupies 14.6% or less of the ARM clock cycles. It is also shown from the experimental results performed in a semi-anechoic chamber that the signal with the front direction can be amplified by approximately 10 dB compared to the other directions.

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.

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...

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.

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-visuel Integre" et Communication(s) ("Integrated Audiovisual" and Communication)

Science.gov (United States)

Moirand, Sophie

1974-01-01

This article examines the usefullness of the audiovisual method in teaching communication competence, and calls for research in audiovisual methods as well as in communication theory for improvement in these areas. (Text is in French.) (AM)

Harmonic Enhancement in Low Bitrate Audio Coding Using an Efficient Long-Term Predictor

Directory of Open Access Journals (Sweden)

Song Jeongook

2010-01-01

Full Text Available This paper proposes audio coding using an efficient long-term prediction method to enhance the perceptual quality of audio codecs to speech input signals at low bit-rates. The MPEG-4 AAC-LTP exploited a similar concept, but its improvement was not significant because of small prediction gain due to long prediction lags and aliased components caused by the transformation with a time-domain aliasing cancelation (TDAC technique. The proposed algorithm increases the prediction gain by employing a deharmonizing predictor and a long-term compensation filter. The look-back memory elements are first constructed by applying the de-harmonizing predictor to the input signal, then the prediction residual is encoded and decoded by transform audio coding. Finally, the long-term compensation filter is applied to the updated look-back memory of the decoded prediction residual to obtain synthesized signals. Experimental results show that the proposed algorithm has much lower spectral distortion and higher perceptual quality than conventional approaches especially for harmonic signals, such as voiced speech.

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.

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...

All-optical signal processing data communication and storage applications

CERN Document Server

Eggleton, Benjamin

2015-01-01

This book provides a comprehensive review of the state-of-the art of optical signal processing technologies and devices. It presents breakthrough solutions for enabling a pervasive use of optics in data communication and signal storage applications. It presents presents optical signal processing as solution to overcome the capacity crunch in communication networks. The book content ranges from the development of innovative materials and devices, such as graphene and slow light structures, to the use of nonlinear optics for secure quantum information processing and overcoming the classical Shannon limit on channel capacity and microwave signal processing. Although it holds the promise for a substantial speed improvement, today’s communication infrastructure optics remains largely confined to the signal transport layer, as it lags behind electronics as far as signal processing is concerned. This situation will change in the near future as the tremendous growth of data traffic requires energy efficient and ful...

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...

2012 Proceedings of the International Conference on Communications, Signal Processing, and Systems

CERN Document Server

Wang, Wei; Mu, Jiasong; Liang, Jing; Zhang, Baoju; Pi, Yiming; Zhao, Chenglin

2012-01-01

Communications, Signal Processing, and Systems is a collection of contributions coming out of the International Conference on Communications, Signal Processing, and Systems (CSPS) held October 2012. This book provides the state-of-art developments of Communications, Signal Processing, and Systems, and their interactions in multidisciplinary fields, such as Smart Grid. The book also examines Radar Systems, Sensor Networks, Radar Signal Processing, Design and Implementation of Signal Processing Systems and Applications. Written by experts and students in the fields of Communications, Signal Processing, and Systems.

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.

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.

Advanced Signal Processing for Wireless Multimedia Communications

Directory of Open Access Journals (Sweden)

Xiaodong Wang

2000-01-01

Full Text Available There is at present a worldwide effort to develop next-generation wireless communication systems. It is envisioned that many of the future wireless systems will incorporate considerable signal-processing intelligence in order to provide advanced services such as multimedia transmission. In general, wireless channels can be very hostile media through which to communicate, due to substantial physical impediments, primarily radio-frequency interference and time-arying nature of the channel. The need of providing universal wireless access at high data-rate (which is the aim of many merging wireless applications presents a major technical challenge, and meeting this challenge necessitates the development of advanced signal processing techniques for multiple-access communications in non-stationary interference-rich environments. In this paper, we present some key advanced signal processing methodologies that have been developed in recent years for interference suppression in wireless networks. We will focus primarily on the problem of jointly suppressing multiple-access interference (MAI and intersymbol interference (ISI, which are the limiting sources of interference for the high data-rate wireless systems being proposed for many emerging application areas, such as wireless multimedia. We first present a signal subspace approach to blind joint suppression of MAI and ISI. We then discuss a powerful iterative technique for joint interference suppression and decoding, so-called Turbo multiuser detection, that is especially useful for wireless multimedia packet communications. We also discuss space-time processing methods that employ multiple antennas for interference rejection and signal enhancement. Finally, we touch briefly on the problems of suppressing narrowband interference and impulsive ambient noise, two other sources of radio-frequency interference present in wireless multimedia networks.

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.

The Most Common Feedback Themes in Communication Skills Training in an Internal Medicine Residency Program: Lessons from the Resident Audio-Recording Project.

Science.gov (United States)

Han, Heeyoung; Papireddy, Muralidhar Reddy; Hingle, Susan T; Ferguson, Jacqueline Anne; Koschmann, Timothy; Sandstrom, Steve

2018-07-01

Individualized structured feedback is an integral part of a resident's learning in communication skills. However, it is not clear what feedback residents receive for their communication skills development in real patient care. We will identify the most common feedback topics given to residents regarding communication skills during Internal Medicine residency training. We analyzed Resident Audio-recording Project feedback data from 2008 to 2013 by using a content analysis approach. Using open coding and an iterative categorization process, we identified 15 emerging themes for both positive and negative feedback. The most recurrent feedback topics were Patient education, Thoroughness, Organization, Questioning strategy, and Management. The residents were guided to improve their communication skills regarding Patient education, Thoroughness, Management, and Holistic exploration of patient's problem. Thoroughness and Communication intelligibility were newly identified themes that were rarely discussed in existing frameworks. Assessment rubrics serve as a lens through which we assess the adequacy of the residents' communication skills. Rather than sticking to a specific rubric, we chose to let the rubric evolve through our experience.

Modeling binaural signal detection

NARCIS (Netherlands)

Breebaart, D.J.

2001-01-01

With the advent of multimedia technology and powerful signal processing systems, audio processing and reproduction has gained renewed interest. Examples of products that have been developed are audio coding algorithms to efficiently store and transmit music and speech, or audio reproduction systems

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.

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

Directory of Open Access Journals (Sweden)

Jose Juan Garcia-Hernandez

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

Acoustic signal recovery by thermal demodulation

Science.gov (United States)

Boullosa, R. R.; Santillán, Arturo O.

2006-10-01

One operating mode of recently developed thermoacoustic transducers is as an audio speaker that uses an input superimposed on a direct current; as a result, the audio signal occurs at the same frequency as the input signal. To extend the potential applications of these kinds of sources, the authors propose an alternative driving mode in which a simple thermoacoustic device, consisting of a metal film over a substrate and a heat sink, is excited with a high frequency sinusoid that is amplitude modulated by a lower frequency signal. They show that the modulating signal is recovered in the radiated waves due to a mechanism that is inherent to this type of thermoacoustic process. If the frequency of the carrier is higher than 30kHz and any modulating signal (the one of interest) is in the audio frequency range, only this signal will be heard. Thus, the thermoacoustic device operates as an audio-band, self-demodulating speaker.

Can audio recording of outpatient consultations improve patients recall and understanding?

DEFF Research Database (Denmark)

Wolderslund, Maiken

of the dialogue between the patient and the clinician via the telephone in the consultation room. By dialing a dedicated number, patients can get access to an audio recording of their consultation by entering their social security number along with a PIN. The primary objective of this study is to determine......Introduction Information provided in an outpatient consultation concerns medication, diagnostic tests, treatment and rehabilitation, all of which are crucial knowledge with regards to patient compliance, decision making and general patient satisfaction. Despite good communication skills among...... clinicians, the communication is challenged by the fact that patients tend to forget or misunderstand parts of the information given. Thus we have designed a study which gives the patients a possibility to hear their consultation again. An Interactive Voice Response platform enables an audio recording...

Catecholaminergic contributions to vocal communication signals.

Science.gov (United States)

Matheson, Laura E; Sakata, Jon T

2015-05-01

Social context affects behavioral displays across a variety of species. For example, social context acutely influences the acoustic and temporal structure of vocal communication signals such as speech and birdsong. Despite the prevalence and importance of such social influences, little is known about the neural mechanisms underlying the social modulation of communication. Catecholamines are implicated in the regulation of social behavior and motor control, but the degree to which catecholamines influence vocal communication signals remains largely unknown. Using a songbird, the Bengalese finch, we examined the extent to which the social context in which song is produced affected immediate early gene expression (EGR-1) in catecholamine-synthesising neurons in the midbrain. Further, we assessed the degree to which administration of amphetamine, which increases catecholamine concentrations in the brain, mimicked the effect of social context on vocal signals. We found that significantly more catecholaminergic neurons in the ventral tegmental area and substantia nigra (but not the central grey, locus coeruleus or subcoeruleus) expressed EGR-1 in birds that were exposed to females and produced courtship song than in birds that produced non-courtship song in isolation. Furthermore, we found that amphetamine administration mimicked the effects of social context and caused many aspects of non-courtship song to resemble courtship song. Specifically, amphetamine increased the stereotypy of syllable structure and sequencing, the repetition of vocal elements and the degree of sequence completions. Taken together, these data highlight the conserved role of catecholamines in vocal communication across species, including songbirds and humans. © 2015 Federation of European Neuroscience Societies and John Wiley & Sons Ltd.

Mixed-Signal Architectures for High-Efficiency and Low-Distortion Digital Audio Processing and Power Amplification

Directory of Open Access Journals (Sweden)

Pierangelo Terreni

2010-01-01

Full Text Available The paper addresses the algorithmic and architectural design of digital input power audio amplifiers. A modelling platform, based on a meet-in-the-middle approach between top-down and bottom-up design strategies, allows a fast but still accurate exploration of the mixed-signal design space. Different amplifier architectures are configured and compared to find optimal trade-offs among different cost-functions: low distortion, high efficiency, low circuit complexity and low sensitivity to parameter changes. A novel amplifier architecture is derived; its prototype implements digital processing IP macrocells (oversampler, interpolating filter, PWM cross-point deriver, noise shaper, multilevel PWM modulator, dead time compensator on a single low-complexity FPGA while off-chip components are used only for the power output stage (LC filter and power MOS bridge; no heatsink is required. The resulting digital input amplifier features a power efficiency higher than 90% and a total harmonic distortion down to 0.13% at power levels of tens of Watts. Discussions towards the full-silicon integration of the mixed-signal amplifier in embedded devices, using BCD technology and targeting power levels of few Watts, are also reported.

Demystifying communication signal lost for network redundancy ...

African Journals Online (AJOL)

These studies report on the communication signal lost factors that were analyzed and supported by evidences on coverage analysis activities for Automatic Meter Reading (AMR) systems. We have categorized the influential signal lost factors into four core elements that were concluded based on our field measurement ...

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

Unified communications

OpenAIRE

Kravos, Urban

2011-01-01

In the modern business world, communication are becoming more and more complex. As a solution to this problem unified communications occurred. Using a single communication approach unified communications are the integration of various communication technologies (eg, telephony, unified messaging, audio, video and web conferencing and collaboration tools). Unified Messaging, which represents only part of the unified communications means the integration of different non real time communication t...

Visible light communications modulation and signal processing

CERN Document Server

Wang, Zhaocheng; Huang, Wei; Xu, Zhengyuan

2018-01-01

This informative new book on state-of-the-art visible light communication (VLC) provides, for the first time, a systematical and advanced treatment of modulation and signal processing for VLC. Visible Light Communications: Modulation and Signal Processing offers a practical guide to designing VLC, linking academic research with commercial applications. In recent years, VLC has attracted attention from academia and industry since it has many advantages over the traditional radio frequency, including wide unregulated bandwidth, high security, and low cost. It is a promising complementary technique in 5G and beyond wireless communications, especially in indoor applications. However, lighting constraints have not been fully considered in the open literature when considering VLC system design, and its importance has been underestimated. That’s why this book—written by a team of experts with both academic research experience and industrial development experience in the field—is so welcome. To help readers u...

Sinusoidal Analysis-Synthesis of Audio Using Perceptual Criteria

Science.gov (United States)

Painter, Ted; Spanias, Andreas

2003-12-01

This paper presents a new method for the selection of sinusoidal components for use in compact representations of narrowband audio. The method consists of ranking and selecting the most perceptually relevant sinusoids. The idea behind the method is to maximize the matching between the auditory excitation pattern associated with the original signal and the corresponding auditory excitation pattern associated with the modeled signal that is being represented by a small set of sinusoidal parameters. The proposed component-selection methodology is shown to outperform the maximum signal-to-mask ratio selection strategy in terms of subjective quality.

Signal Conditioning An Introduction to Continuous Wave Communication and Signal Processing

CERN Document Server

Das, Apurba

2012-01-01

"Signal Conditioning” is a comprehensive introduction to electronic signal processing. The book presents the mathematical basics including the implications of various transformed domain representations in signal synthesis and analysis in an understandable and lucid fashion and illustrates the theory through many applications and examples from communication systems. The ease to learn is supported by well-chosen exercises which give readers the flavor of the subject. Supplementary electronic materials available on http://extras.springer.com including MATLAB codes illuminating applications in the domain of one dimensional electrical signal processing, image processing and speech processing. The book is an introduction for students with a basic understanding in engineering or natural sciences.

Fractal Complexity-Based Feature Extraction Algorithm of Communication Signals

Science.gov (United States)

Wang, Hui; Li, Jingchao; Guo, Lili; Dou, Zheng; Lin, Yun; Zhou, Ruolin

How to analyze and identify the characteristics of radiation sources and estimate the threat level by means of detecting, intercepting and locating has been the central issue of electronic support in the electronic warfare, and communication signal recognition is one of the key points to solve this issue. Aiming at accurately extracting the individual characteristics of the radiation source for the increasingly complex communication electromagnetic environment, a novel feature extraction algorithm for individual characteristics of the communication radiation source based on the fractal complexity of the signal is proposed. According to the complexity of the received signal and the situation of environmental noise, use the fractal dimension characteristics of different complexity to depict the subtle characteristics of the signal to establish the characteristic database, and then identify different broadcasting station by gray relation theory system. The simulation results demonstrate that the algorithm can achieve recognition rate of 94% even in the environment with SNR of -10dB, and this provides an important theoretical basis for the accurate identification of the subtle features of the signal at low SNR in the field of information confrontation.

Characteristics of communication with older people in home care: A qualitative analysis of audio recordings of home care visits.

Science.gov (United States)

Kristensen, Dorte V; Sundler, Annelie J; Eide, Hilde; Hafskjold, Linda; Ruud, Iren; Holmström, Inger K

2017-12-01

To describe the characteristics of communication practice in home care visits between older people (over 65 years old) and nurse assistants and to discuss the findings from a person-centered perspective. The older population is increasing worldwide, along with the need for healthcare services in the person's home. To achieve a high-quality care, person-centered communication is crucial. A descriptive design with a qualitative inductive approach was used. Fifteen audio recordings of naturally occurring conversations between 12 nurse assistants and 13 older people in Norway were analysed by qualitative content analysis. Four categories were revealed through analysis: (i) supporting older people's connection to everyday life; (ii) supporting older people's involvement in their own care; (iii) attention to older people's bodily and existential needs; and (iv) the impact of continuity and predictability on older people's well-being. The communication between the older people and the nurse assistants during home care visits was mainly task-oriented, but also related to the person. The older people were involved in the tasks to be carried out and humour was part of the communication. Greater attention was paid to bodily than existential needs. The communication was connected with the older people's everyday life in several ways. Time frames and interruptions concern the older people; hearing and speech impairments were a challenge to communication. To enhance person-centred communication, further studies are needed, especially intervention studies for healthcare professionals and students. Being responsive to older people's subjective experiences is important in meeting their needs in home care. Communication that addresses the need for trust and predictability is important for older people. Responding to existential needs require more attention. The home care setting has an impact on communication. © 2017 John Wiley & Sons Ltd.

Anthropomorphic Coding of Speech and Audio: A Model Inversion Approach

Directory of Open Access Journals (Sweden)

W. Bastiaan Kleijn

2005-06-01

Full Text Available Auditory modeling is a well-established methodology that provides insight into human perception and that facilitates the extraction of signal features that are most relevant to the listener. The aim of this paper is to provide a tutorial on perceptual speech and audio coding using an invertible auditory model. In this approach, the audio signal is converted into an auditory representation using an invertible auditory model. The auditory representation is quantized and coded. Upon decoding, it is then transformed back into the acoustic domain. This transformation converts a complex distortion criterion into a simple one, thus facilitating quantization with low complexity. We briefly review past work on auditory models and describe in more detail the components of our invertible model and its inversion procedure, that is, the method to reconstruct the signal from the output of the auditory model. We summarize attempts to use the auditory representation for low-bit-rate coding. Our approach also allows the exploitation of the inherent redundancy of the human auditory system for the purpose of multiple description (joint source-channel coding.

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.

Plant signalling: the opportunities and dangers of chemical communication.

Science.gov (United States)

Adler, Frederick R

2011-04-23

The notion of chemical communication between plants and other organisms has gone from being viewed as a fringe idea to an accepted ecological phenomenon only recently. An Organized Oral Session at the August 2010 Ecological Society of America meeting in Pittsburgh examined the role of plant signalling both within and between plants, with speakers addressing the remarkably wide array of effects that plant signals have on plant physiology, species interactions and entire communities. In addition to the familiar way that plants communicate with mutualists like pollinators and fruit dispersers through both chemical and visual cues, speakers at this session described how plants communicate with themselves, with each other, with herbivores and with predators of those herbivores. These plant signals create a complex odour web superimposed upon the more classical food web itself, with its own dynamics in the face of exotic species and rapid community assembly and disassembly.

The Role of Audio Media in the Lives of Children.

Science.gov (United States)

Christenson, Peter G.; Lindlof, Thomas R.

Mass communication researchers have largely ignored the role of audio media and popular music in the lives of children, yet the available evidence shows that children do listen. Extant studies yield a consistent developmental portrait of childrens' listening frequency, but there is a notable lack of programatic research over the past decade, one…

Flash signal evolution in Photinus fireflies: character displacement and signal exploitation in a visual communication system.

Science.gov (United States)

Stanger-Hall, Kathrin F; Lloyd, James E

2015-03-01

Animal communication is an intriguing topic in evolutionary biology. In this comprehensive study of visual signal evolution, we used a phylogenetic approach to study the evolution of the flash communication system of North American fireflies. The North American firefly genus Photinus contains 35 described species with simple ON-OFF visual signals, and information on habitat types, sympatric congeners, and predators. This makes them an ideal study system to test hypotheses on the evolution of male and female visual signal traits. Our analysis of 34 Photinus species suggests two temporal pattern generators: one for flash duration and one for flash intervals. Reproductive character displacement was a main factor for signal divergence in male flash duration among sympatric Photinus species. Male flash pattern intervals (i.e., the duration of the dark periods between signals) were positively correlated with the number of sympatric Photuris fireflies, which include predators of Photinus. Females of different Photinus species differ in their response preferences to male traits. As in other communication systems, firefly male sexual signals seem to be a compromise between optimizing mating success (sexual selection) and minimizing predation risk (natural selection). An integrative model for Photinus signal evolution is proposed. © 2015 The Author(s).

Realisierung eines verzerrungsarmen Open-Loop Klasse-D Audio-Verstärkers mit SB-ZePoC

Directory of Open Access Journals (Sweden)

O. Schnick

2007-06-01

Full Text Available In den letzten Jahren hat die Entwicklung von Klasse-D Verstärkern für Audio-Anwendungen ein vermehrtes Interesse auf sich gezogen. Eine Motivation hierfür liegt in der mit dieser Technik extrem hohen erzielbaren Effizienz von über 90%. Die Signale, die Klasse-D Verstärker steuern, sind binär. Immer mehr Audio-Signale werden entweder digital gespeichert (CD, DVD, MP3 oder digital übermittelt (Internet, DRM, DAB, DVB-T, DVB-S, GMS, UMTS, weshalb eine direkte Umsetzung dieser Daten in ein binäres Steuersignal ohne vorherige konventionelle D/A-Wandlung erstrebenswert erscheint.

Die klassischen Pulsweitenmodulationsverfahren führen zu Aliasing-Komponenten im Audio-Basisband. Diese Verzerrungen können nur durch eine sehr hohe Schaltfrequenz auf ein akzeptables Maß reduziert werden. Durch das von der Forschungsgruppe um Prof. Mathis vorgestellte SB-ZePoC Verfahren (Zero Position Coding with Separated Baseband wird diese Art der Signalverzerrung durch Generierung eines separierten Basisbands verhindert. Deshalb können auch niedrige Schaltfrequenzen gewählt werden. Dadurch werden nicht nur die Schaltverluste, sondern auch Timing-Verzerrungen verringert, die durch die nichtideale Schaltendstufe verursacht werden. Diese tragen einen großen Anteil zu den gesamten Verzerrungen eines Klasse-D Verstärkers bei. Mit dem SB-ZePoC Verfahren lassen sich verzerrungsarme Open-Loop Klasse-D Audio-Verstärker realisieren, die ohne aufwändige Gegenkopplungsschleifen auskommen.

Class-D amplifiers are suiteble for amplification of audio signals. One argument is their high efficiency of 90% and more. Today most of the audio signals are stored or transmitted in digital form. A digitally controlled Class-D amplifier can be directly driven with coded (modulated data. No separate D/A conversion is needed. Classical modulation schemes like Pulse-Width-Modulation (PWM cause aliasing. So a very high switching rate is required to minimize the

Non Audio-Video gesture recognition system

DEFF Research Database (Denmark)

Craciunescu, Razvan; Mihovska, Albena Dimitrova; Kyriazakos, Sofoklis

2016-01-01

Gesture recognition is a topic in computer science and language technology with the goal of interpreting human gestures via mathematical algorithms. Gestures can originate from any bodily motion or state but commonly originate from the face or hand. Current research focus includes on the emotion...... recognition from the face and hand gesture recognition. Gesture recognition enables humans to communicate with the machine and interact naturally without any mechanical devices. This paper investigates the possibility to use non-audio/video sensors in order to design a low-cost gesture recognition device...

Audio-visual speech timing sensitivity is enhanced in cluttered conditions.

Directory of Open Access Journals (Sweden)

Warrick Roseboom

2011-04-01

Full Text Available Events encoded in separate sensory modalities, such as audition and vision, can seem to be synchronous across a relatively broad range of physical timing differences. This may suggest that the precision of audio-visual timing judgments is inherently poor. Here we show that this is not necessarily true. We contrast timing sensitivity for isolated streams of audio and visual speech, and for streams of audio and visual speech accompanied by additional, temporally offset, visual speech streams. We find that the precision with which synchronous streams of audio and visual speech are identified is enhanced by the presence of additional streams of asynchronous visual speech. Our data suggest that timing perception is shaped by selective grouping processes, which can result in enhanced precision in temporally cluttered environments. The imprecision suggested by previous studies might therefore be a consequence of examining isolated pairs of audio and visual events. We argue that when an isolated pair of cross-modal events is presented, they tend to group perceptually and to seem synchronous as a consequence. We have revealed greater precision by providing multiple visual signals, possibly allowing a single auditory speech stream to group selectively with the most synchronous visual candidate. The grouping processes we have identified might be important in daily life, such as when we attempt to follow a conversation in a crowded room.

Human sensorimotor communication: a theory of signaling in online social interactions.

Science.gov (United States)

Pezzulo, Giovanni; Donnarumma, Francesco; Dindo, Haris

2013-01-01

Although the importance of communication is recognized in several disciplines, it is rarely studied in the context of online social interactions and joint actions. During online joint actions, language and gesture are often insufficient and humans typically use non-verbal, sensorimotor forms of communication to send coordination signals. For example, when playing volleyball, an athlete can exaggerate her movements to signal her intentions to her teammates (say, a pass to the right) or to feint an adversary. Similarly, a person who is transporting a table together with a co-actor can push the table in a certain direction to signal where and when he intends to place it. Other examples of "signaling" are over-articulating in noisy environments and over-emphasizing vowels in child-directed speech. In all these examples, humans intentionally modify their action kinematics to make their goals easier to disambiguate. At the moment no formal theory exists of these forms of sensorimotor communication and signaling. We present one such theory that describes signaling as a combination of a pragmatic and a communicative action, and explains how it simplifies coordination in online social interactions. We cast signaling within a "joint action optimization" framework in which co-actors optimize the success of their interaction and joint goals rather than only their part of the joint action. The decision of whether and how much to signal requires solving a trade-off between the costs of modifying one's behavior and the benefits in terms of interaction success. Signaling is thus an intentional strategy that supports social interactions; it acts in concert with automatic mechanisms of resonance, prediction, and imitation, especially when the context makes actions and intentions ambiguous and difficult to read. Our theory suggests that communication dynamics should be studied within theories of coordination and interaction rather than only in terms of the maximization of information

Human sensorimotor communication: a theory of signaling in online social interactions.

Directory of Open Access Journals (Sweden)

Giovanni Pezzulo

Full Text Available Although the importance of communication is recognized in several disciplines, it is rarely studied in the context of online social interactions and joint actions. During online joint actions, language and gesture are often insufficient and humans typically use non-verbal, sensorimotor forms of communication to send coordination signals. For example, when playing volleyball, an athlete can exaggerate her movements to signal her intentions to her teammates (say, a pass to the right or to feint an adversary. Similarly, a person who is transporting a table together with a co-actor can push the table in a certain direction to signal where and when he intends to place it. Other examples of "signaling" are over-articulating in noisy environments and over-emphasizing vowels in child-directed speech. In all these examples, humans intentionally modify their action kinematics to make their goals easier to disambiguate. At the moment no formal theory exists of these forms of sensorimotor communication and signaling. We present one such theory that describes signaling as a combination of a pragmatic and a communicative action, and explains how it simplifies coordination in online social interactions. We cast signaling within a "joint action optimization" framework in which co-actors optimize the success of their interaction and joint goals rather than only their part of the joint action. The decision of whether and how much to signal requires solving a trade-off between the costs of modifying one's behavior and the benefits in terms of interaction success. Signaling is thus an intentional strategy that supports social interactions; it acts in concert with automatic mechanisms of resonance, prediction, and imitation, especially when the context makes actions and intentions ambiguous and difficult to read. Our theory suggests that communication dynamics should be studied within theories of coordination and interaction rather than only in terms of the

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...

Sound localization with head movement: implications for 3-d audio displays.

Directory of Open Access Journals (Sweden)

Ken Ian McAnally

2014-08-01

Full Text Available Previous studies have shown that the accuracy of sound localization is improved if listeners are allowed to move their heads during signal presentation. This study describes the function relating localization accuracy to the extent of head movement in azimuth. Sounds that are difficult to localize were presented in the free field from sources at a wide range of azimuths and elevations. Sounds remained active until the participants’ heads had rotated through windows ranging in width of 2°, 4°, 8°, 16°, 32°, or 64° of azimuth. Error in determining sound-source elevation and the rate of front/back confusion were found to decrease with increases in azimuth window width. Error in determining sound-source lateral angle was not found to vary with azimuth window width. Implications for 3-d audio displays: The utility of a 3-d audio display for imparting spatial information is likely to be improved if operators are able to move their heads during signal presentation. Head movement may compensate in part for a paucity of spectral cues to sound-source location resulting from limitations in either the audio signals presented or the directional filters (i.e., head-related transfer functions used to generate a display. However, head movements of a moderate size (i.e., through around 32° of azimuth may be required to ensure that spatial information is conveyed with high accuracy.

Signals and cues in the evolution of plant-microbe communication.

Science.gov (United States)

Padje, Anouk Van't; Whiteside, Matthew D; Kiers, E Toby

2016-08-01

Communication has played a key role in organismal evolution. If sender and receiver have a shared interest in propagating reliable information, such as when they are kin relatives, then effective communication can bring large fitness benefits. However, interspecific communication (among different species) is more prone to dishonesty. Over the last decade, plants and their microbial root symbionts have become a model system for studying interspecific molecular crosstalk. However, less is known about the evolutionary stability of plant-microbe communication. What prevents partners from hijacking or manipulating information to their own benefit? Here, we focus on communication between arbuscular mycorrhizal fungi and their host plants. We ask how partners use directed signals to convey specific information, and highlight research on the problem of dishonest signaling. Copyright © 2016 Elsevier Ltd. All rights reserved.

47 CFR 25.214 - Technical requirements for space stations in the satellite digital audio radio service and...

Science.gov (United States)

2010-10-01

... 47 Telecommunication 2 2010-10-01 2010-10-01 false Technical requirements for space stations in the satellite digital audio radio service and associated terrestrial repeaters. 25.214 Section 25.214 Telecommunication FEDERAL COMMUNICATIONS COMMISSION (CONTINUED) COMMON CARRIER SERVICES SATELLITE COMMUNICATIONS...

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...

Evolution of Signaling in a Multi-Robot System: Categorization and Communication

Science.gov (United States)

Ampatzis, Christos; Tuci, Elio; Trianni, Vito; Dorigo, Marco

We use Evolutionary Robotics to design robot controllers in which decision-making mechanisms to switch from solitary to social behavior are integrated with the mechanisms that underpin the sensory-motor repertoire of the robots. In particular, we study the evolution of behavioral and communicative skills in a categorization task. The individual decision-making structures are based on the integration over time of sensory information. The mechanisms for switching from solitary to social behavior and the ways in which the robots can affect each other's behavior are not predetermined by the experimenter, but are aspects of our model designed by artificial evolution. Our results show that evolved robots manage to cooperate and collectively discriminate between different environments by developing a simple communication protocol based on sound signaling. Communication emerges in the absence of explicit selective pressure coded in the fitness function. The evolution of communication is neither trivial nor obvious; for a meaningful signaling system to evolve, evolution must produce both appropriate signals and appropriate reactions to signals. The use of communication proves to be adaptive for the group, even if, in principle, non-cooperating robots can be equally successful with cooperating robots.

A Review on Human Body Communication: Signal Propagation Model, Communication Performance, and Experimental Issues

Directory of Open Access Journals (Sweden)

Jian Feng Zhao

2017-01-01

Full Text Available Human body communication (HBC, which uses the human body tissue as the transmission medium to transmit health informatics, serves as a promising physical layer solution for the body area network (BAN. The human centric nature of HBC offers an innovative method to transfer the healthcare data, whose transmission requires low interference and reliable data link. Therefore, the deployment of HBC system obtaining good communication performance is required. In this regard, a tutorial review on the important issues related to HBC data transmission such as signal propagation model, channel characteristics, communication performance, and experimental considerations is conducted. In this work, the development of HBC and its first attempts are firstly reviewed. Then a survey on the signal propagation models is introduced. Based on these models, the channel characteristics are summarized; the communication performance and selection of transmission parameters are also investigated. Moreover, the experimental issues, such as electrodes and grounding strategies, are also discussed. Finally, the recommended future studies are provided.

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.

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....

Design and Implementation of a linear-phase equalizer in digital audio signal processing

NARCIS (Netherlands)

Slump, Cornelis H.; van Asma, C.G.M.; Barels, J.K.P.; Barels, J.K.P.; Brunink, W.J.A; Drenth, F.B.; Pol, J.V.; Schouten, D.S.; Samsom, M.M.; Samsom, M.M.; Herrmann, O.E.

1992-01-01

This contribution presents the four phases of a project aiming at the realization in VLSI of a digital audio equalizer with a linear phase characteristic. The first step includes the identification of the system requirements, based on experience and (psycho-acoustical) literature. Secondly, the

Regulation of cellular communication by signaling microdomains in the blood vessel wall.

Science.gov (United States)

Billaud, Marie; Lohman, Alexander W; Johnstone, Scott R; Biwer, Lauren A; Mutchler, Stephanie; Isakson, Brant E

2014-01-01

It has become increasingly clear that the accumulation of proteins in specific regions of the plasma membrane can facilitate cellular communication. These regions, termed signaling microdomains, are found throughout the blood vessel wall where cellular communication, both within and between cell types, must be tightly regulated to maintain proper vascular function. We will define a cellular signaling microdomain and apply this definition to the plethora of means by which cellular communication has been hypothesized to occur in the blood vessel wall. To that end, we make a case for three broad areas of cellular communication where signaling microdomains could play an important role: 1) paracrine release of free radicals and gaseous molecules such as nitric oxide and reactive oxygen species; 2) role of ion channels including gap junctions and potassium channels, especially those associated with the endothelium-derived hyperpolarization mediated signaling, and lastly, 3) mechanism of exocytosis that has considerable oversight by signaling microdomains, especially those associated with the release of von Willebrand factor. When summed, we believe that it is clear that the organization and regulation of signaling microdomains is an essential component to vessel wall function.

Regulation of Cellular Communication by Signaling Microdomains in the Blood Vessel Wall

Science.gov (United States)

Billaud, Marie; Lohman, Alexander W.; Johnstone, Scott R.; Biwer, Lauren A.; Mutchler, Stephanie; Isakson, Brant E.

2014-01-01

It has become increasingly clear that the accumulation of proteins in specific regions of the plasma membrane can facilitate cellular communication. These regions, termed signaling microdomains, are found throughout the blood vessel wall where cellular communication, both within and between cell types, must be tightly regulated to maintain proper vascular function. We will define a cellular signaling microdomain and apply this definition to the plethora of means by which cellular communication has been hypothesized to occur in the blood vessel wall. To that end, we make a case for three broad areas of cellular communication where signaling microdomains could play an important role: 1) paracrine release of free radicals and gaseous molecules such as nitric oxide and reactive oxygen species; 2) role of ion channels including gap junctions and potassium channels, especially those associated with the endothelium-derived hyperpolarization mediated signaling, and lastly, 3) mechanism of exocytosis that has considerable oversight by signaling microdomains, especially those associated with the release of von Willebrand factor. When summed, we believe that it is clear that the organization and regulation of signaling microdomains is an essential component to vessel wall function. PMID:24671377

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

Digital signal processing for wireless communication using Matlab

CERN Document Server

Gopi, E S

2016-01-01

This book examines signal processing techniques used in wireless communication illustrated by using the Matlab program. The author discusses these techniques as they relate to Doppler spread; delay spread; Rayleigh and Rician channel modeling; rake receiver; diversity techniques; MIMO and OFDM -based transmission techniques; and array signal processing. Related topics such as detection theory, link budget, multiple access techniques, and spread spectrum are also covered.   ·         Illustrates signal processing techniques involved in wireless communication using Matlab ·         Discusses multiple access techniques such as Frequency division multiple access, Time division multiple access, and Code division multiple access ·         Covers band pass modulation techniques such as Binary phase shift keying, Differential phase shift keying, Quadrature phase shift keying, Binary frequency shift keying, Minimum shift keying, and Gaussian minimum shift keying.

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.

Advances in Computer, Communication, Control and Automation

CERN Document Server

011 International Conference on Computer, Communication, Control and Automation

2012-01-01

The volume includes a set of selected papers extended and revised from the 2011 International Conference on Computer, Communication, Control and Automation (3CA 2011). 2011 International Conference on Computer, Communication, Control and Automation (3CA 2011) has been held in Zhuhai, China, November 19-20, 2011. This volume  topics covered include signal and Image processing, speech and audio Processing, video processing and analysis, artificial intelligence, computing and intelligent systems, machine learning, sensor and neural networks, knowledge discovery and data mining, fuzzy mathematics and Applications, knowledge-based systems, hybrid systems modeling and design, risk analysis and management, system modeling and simulation. We hope that researchers, graduate students and other interested readers benefit scientifically from the proceedings and also find it stimulating in the process.

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.

A Survey of Open Source Products for Building a SIP Communication Platform

Directory of Open Access Journals (Sweden)

Pavel Segec

2011-01-01

Full Text Available The Session Initiation Protocol (SIP is a multimedia signalling protocol that has evolved into a widely adopted communication standard. The integration of SIP into existing IP networks has fostered IP networks becoming a convergence platform for both real-time and non-real-time multimedia communications. This converged platform integrates data, voice, video, presence, messaging, and conference services into a single network that offers new communication experiences for users. The open source community has contributed to SIP adoption through the development of open source software for both SIP clients and servers. In this paper, we provide a survey on open SIP systems that can be built using publically available software. We identify SIP features for service development and programming, services and applications of a SIP-converged platform, and the most important technologies supporting SIP functionalities. We propose an advanced converged IP communication platform that uses SIP for service delivery. The platform supports audio and video calls, along with media services such as audio conferences, voicemail, presence, and instant messaging. Using SIP Application Programming Interfaces (APIs, the platform allows the deployment of advanced integrated services. The platform is implemented with open source software. Architecture components run on standardized hardware with no need for special purpose investments.

An audio FIR-DAC in a BCD process for high power Class-D amplifiers

NARCIS (Netherlands)

Doorn, T.S.; van Tuijl, Adrianus Johannes Maria; Schinkel, Daniel; Annema, Anne J.; Berkhout, M.; Berkhout, M.; Nauta, Bram

A 322 coefficient semi-digital FIR-DAC using a 1-bit PWM input signal was designed and implemented in a high voltage, audio power bipolar CMOS DMOS (BCD) process. This facilitates digital input signals for an analog class-D amplifier in BCD. The FIR-DAC performance depends on the ISI-resistant

Explaining evolution of plant communication by airborne signals.

Science.gov (United States)

Heil, Martin; Karban, Richard

2010-03-01

In spite of initial doubts about the reality of 'talking trees', plant resistance expression mediated by volatile compounds that come from neighboring plants is now well described. Airborne signals usually improve the resistance of the receiver, but without obvious benefits for the emitter, thus making the evolutionary explanation of this phenomenon problematic. Here, we discuss four possible non-exclusive explanations involving the role of volatiles: in direct defense, as within-plant signals, as traits that synergistically interact with other defenses, and as cues among kin. Unfortunately, there is a lack of knowledge on the fitness consequences of plant communication for both emitter and receiver. This information is crucial to understanding the ecology and evolution of plant communication via airborne cues.

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....

A Comparison of Television and Audio Presentations of the MLA French Listening Examination

Science.gov (United States)

Stallings, William M.

1972-01-01

Although nonverbal cues are often available in real-life communication, listening is usually tested by aural stimuli broadcast from an audio-tape. It would seem that testing listening comprehension might be improved by using television to offer nonverbal cues in addition to aural stimuli. (Author)

4th International Conference on Communications, Signal Processing, and Systems

CERN Document Server

Mu, Jiasong; Wang, Wei; Zhang, Baoju

2016-01-01

This book brings together papers presented at the 4th International Conference on Communications, Signal Processing, and Systems, which provides a venue to disseminate the latest developments and to discuss the interactions and links between these multidisciplinary fields. Spanning topics ranging from Communications, Signal Processing and Systems, this book is aimed at undergraduate and graduate students in Electrical Engineering, Computer Science and Mathematics, researchers and engineers from academia and industry as well as government employees (such as NSF, DOD, DOE, etc).

A simple clockless Network-on-Chip for a commercial audio DSP chip

DEFF Research Database (Denmark)

Stensgaard, Mikkel Bystrup; Bjerregaard, Tobias; Sparsø, Jens

2006-01-01

We design a very small, packet-switched, clockless Network-on-Chip (NoC) as a replacement for the existing crossbar-based communication infrastructure in a commercial audio DSP chip. Both solutions are laid out in a 0.18 um process, and compared in terms of area, power consumption and routing...... to the existing crossbar, it allows all blocks to communicate. The total wire length is decreased by 22% which eases the layout process and makes the design less prone to routing congestion. Not least, the communicating blocks are decoupled by means of the NoC, providing a Globally-Asynchronous, Locally...

Retrograde Signals: Integrators of Interorganellar Communication and Orchestrators of Plant Development.

Science.gov (United States)

de Souza, Amancio; Wang, Jin-Zheng; Dehesh, Katayoon

2017-04-28

Interorganellar cooperation maintained via exquisitely controlled retrograde-signaling pathways is an evolutionary necessity for maintenance of cellular homeostasis. This signaling feature has therefore attracted much research attention aimed at improving understanding of the nature of these communication signals, how the signals are sensed, and ultimately the mechanism by which they integrate targeted processes that collectively culminate in organellar cooperativity. The answers to these questions will provide insight into how retrograde-signal-mediated regulatory mechanisms are recruited and which biological processes are targeted, and will advance our understanding of how organisms balance metabolic investments in growth against adaptation to environmental stress. This review summarizes the present understanding of the nature and the functional complexity of retrograde signals as integrators of interorganellar communication and orchestrators of plant development, and offers a perspective on the future of this critical and dynamic area of research.

Finding the Correspondence of Audio-Visual Events by Object Manipulation

Science.gov (United States)

Nishibori, Kento; Takeuchi, Yoshinori; Matsumoto, Tetsuya; Kudo, Hiroaki; Ohnishi, Noboru

A human being understands the objects in the environment by integrating information obtained by the senses of sight, hearing and touch. In this integration, active manipulation of objects plays an important role. We propose a method for finding the correspondence of audio-visual events by manipulating an object. The method uses the general grouping rules in Gestalt psychology, i.e. “simultaneity” and “similarity” among motion command, sound onsets and motion of the object in images. In experiments, we used a microphone, a camera, and a robot which has a hand manipulator. The robot grasps an object like a bell and shakes it or grasps an object like a stick and beat a drum in a periodic, or non-periodic motion. Then the object emits periodical/non-periodical events. To create more realistic scenario, we put other event source (a metronome) in the environment. As a result, we had a success rate of 73.8 percent in finding the correspondence between audio-visual events (afferent signal) which are relating to robot motion (efferent signal).

Radio Science from an Optical Communications Signal

Science.gov (United States)

Moision, Bruce; Asmar, Sami; Oudrhiri, Kamal

2013-01-01

NASA is currently developing the capability to deploy deep space optical communications links. This creates the opportunity to utilize the optical link to obtain range, doppler, and signal intensity estimates. These may, in turn, be used to complement or extend the capabilities of current radio science. In this paper we illustrate the achievable precision in estimating range, doppler, and received signal intensity of an non-coherent optical link (the current state-of-the-art for a deep-space link). We provide a joint estimation algorithm with performance close to the bound. We draw comparisons to estimates based on a coherent radio frequency signal, illustrating that large gains in either precision or observation time are possible with an optical link.

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…

A Method to Detect AAC Audio Forgery

Directory of Open Access Journals (Sweden)

Qingzhong Liu

2015-08-01

Full Text Available Advanced Audio Coding (AAC, a standardized lossy compression scheme for digital audio, which was designed to be the successor of the MP3 format, generally achieves better sound quality than MP3 at similar bit rates. While AAC is also the default or standard audio format for many devices and AAC audio files may be presented as important digital evidences, the authentication of the audio files is highly needed but relatively missing. In this paper, we propose a scheme to expose tampered AAC audio streams that are encoded at the same encoding bit-rate. Specifically, we design a shift-recompression based method to retrieve the differential features between the re-encoded audio stream at each shifting and original audio stream, learning classifier is employed to recognize different patterns of differential features of the doctored forgery files and original (untouched audio files. Experimental results show that our approach is very promising and effective to detect the forgery of the same encoding bit-rate on AAC audio streams. Our study also shows that shift recompression-based differential analysis is very effective for detection of the MP3 forgery at the same bit rate.

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

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.

Hierarchical structure for audio-video based semantic classification of sports video sequences

Science.gov (United States)

Kolekar, M. H.; Sengupta, S.

2005-07-01

A hierarchical structure for sports event classification based on audio and video content analysis is proposed in this paper. Compared to the event classifications in other games, those of cricket are very challenging and yet unexplored. We have successfully solved cricket video classification problem using a six level hierarchical structure. The first level performs event detection based on audio energy and Zero Crossing Rate (ZCR) of short-time audio signal. In the subsequent levels, we classify the events based on video features using a Hidden Markov Model implemented through Dynamic Programming (HMM-DP) using color or motion as a likelihood function. For some of the game-specific decisions, a rule-based classification is also performed. Our proposed hierarchical structure can easily be applied to any other sports. Our results are very promising and we have moved a step forward towards addressing semantic classification problems in general.

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...

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)

A centralized audio presentation manager

Energy Technology Data Exchange (ETDEWEB)

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

1994-05-16

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

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

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

Text messaging as a strategy to address the limits of audio-based communication during mass-gathering events with high ambient noise.

Science.gov (United States)

Lund, Adam; Wong, Daniel; Lewis, Kerrie; Turris, Sheila A; Vaisler, Sean; Gutman, Samuel

2013-02-01

The provision of medical care in environments with high levels of ambient noise (HLAN), such as concerts or sporting events, presents unique communication challenges. Audio transmissions can be incomprehensible to the receivers. Text-based communications may be a valuable primary and/or secondary means of communication in this type of setting. To evaluate the usability of text-based communications in parallel with standard two-way radio communications during mass-gathering (MG) events in the context of HLAN. This Canadian study used outcome survey methods to evaluate the performance of communication devices during MG events. Ten standard commercially available handheld smart phones loaded with basic voice and data plans were assigned to health care providers (HCPs) for use as an adjunct to the medical team's typical radio-based communication. Common text messaging and chat platforms were trialed. Both efficacy and provider satisfaction were evaluated. During a 23-month period, the smart phones were deployed at 17 events with HLAN for a total of 40 event days or approximately 460 hours of active use. Survey responses from health care providers (177) and dispatchers (26) were analyzed. The response rate was unknown due to the method of recruitment. Of the 155 HCP responses to the question measuring difficulty of communication in environments with HLAN, 68.4% agreed that they "occasionally" or "frequently" found it difficult to clearly understand voice communications via two-way radio. Similarly, of the 23 dispatcher responses to the same item, 65.2% of the responses indicated that "occasionally" or "frequently" HLAN negatively affected the ability to communicate clearly with team members. Of the 168 HCP responses to the item assessing whether text-based communication improved the ability to understand and respond to calls when compared to radio alone, 86.3% "agreed" or "strongly agreed" that this was the case. The dispatcher responses (n = 21) to the same item also

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.

Biochar and microbial signaling: production conditions determine effects on microbial communication

Science.gov (United States)

Masiello, Caroline A.; Chen, Ye; Gao, Xiaodong; Liu, Shirley; Cheng, Hsiao-Ying; Bennett, Matthew R.; Rudgers, Jennifer A.; Wagner, Daniel S.; Zygourakis, Kyriacos; Silberg, Jonathan J.

2013-01-01

Charcoal has a long soil residence time, which has resulted in its production and use as a carbon sequestration technique (biochar). A range of biological effects can be triggered by soil biochar that can positively and negatively influence carbon storage, such as changing the decomposition rate of organic matter and altering plant biomass production. Sorption of cellular signals has been hypothesized to underlie some of these effects, but it remains unknown whether the binding of biochemical signals occurs, and if so, on time scales relevant to microbial growth and communication. We examined biochar sorption of N-3-oxo-dodecanoyl-L-homoserine lactone, an acyl-homoserine lactone (AHL) intercellular signaling molecule used by many gram-negative soil microbes to regulate gene expression. We show that wood biochars disrupt communication within a growing multicellular system that is made up of sender cells that synthesize AHL and receiver cells that express green fluorescent protein in response to an AHL signal. However, biochar inhibition of AHL-mediated cell-cell communication varied, with the biochar prepared at 700°C (surface area of 301 m2/g) inhibiting cellular communication 10-fold more than an equivalent mass of biochar prepared at 300°C (surface area of 3 m2/g). These findings provide the first direct evidence that biochars elicit a range of effects on gene expression dependent on intercellular signaling, implicating the method of biochar preparation as a parameter that could be tuned to regulate microbial-dependent soil processes, like nitrogen fixation and pest attack of root crops. PMID:24066613

Biochar and microbial signaling: production conditions determine effects on microbial communication.

Science.gov (United States)

Masiello, Caroline A; Chen, Ye; Gao, Xiaodong; Liu, Shirley; Cheng, Hsiao-Ying; Bennett, Matthew R; Rudgers, Jennifer A; Wagner, Daniel S; Zygourakis, Kyriacos; Silberg, Jonathan J

2013-10-15

Charcoal has a long soil residence time, which has resulted in its production and use as a carbon sequestration technique (biochar). A range of biological effects can be triggered by soil biochar that can positively and negatively influence carbon storage, such as changing the decomposition rate of organic matter and altering plant biomass production. Sorption of cellular signals has been hypothesized to underlie some of these effects, but it remains unknown whether the binding of biochemical signals occurs, and if so, on time scales relevant to microbial growth and communication. We examined biochar sorption of N-3-oxo-dodecanoyl-L-homoserine lactone, an acyl-homoserine lactone (AHL) intercellular signaling molecule used by many gram-negative soil microbes to regulate gene expression. We show that wood biochars disrupt communication within a growing multicellular system that is made up of sender cells that synthesize AHL and receiver cells that express green fluorescent protein in response to an AHL signal. However, biochar inhibition of AHL-mediated cell-cell communication varied, with the biochar prepared at 700 °C (surface area of 301 m(2)/g) inhibiting cellular communication 10-fold more than an equivalent mass of biochar prepared at 300 °C (surface area of 3 m(2)/g). These findings provide the first direct evidence that biochars elicit a range of effects on gene expression dependent on intercellular signaling, implicating the method of biochar preparation as a parameter that could be tuned to regulate microbial-dependent soil processes, like nitrogen fixation and pest attack of root crops.

Audio frequency in vivo optical coherence elastography

Science.gov (United States)

Adie, Steven G.; Kennedy, Brendan F.; Armstrong, Julian J.; Alexandrov, Sergey A.; Sampson, David D.

2009-05-01

We present a new approach to optical coherence elastography (OCE), which probes the local elastic properties of tissue by using optical coherence tomography to measure the effect of an applied stimulus in the audio frequency range. We describe the approach, based on analysis of the Bessel frequency spectrum of the interferometric signal detected from scatterers undergoing periodic motion in response to an applied stimulus. We present quantitative results of sub-micron excitation at 820 Hz in a layered phantom and the first such measurements in human skin in vivo.

Audio frequency in vivo optical coherence elastography

International Nuclear Information System (INIS)

Adie, Steven G; Kennedy, Brendan F; Armstrong, Julian J; Alexandrov, Sergey A; Sampson, David D

2009-01-01

We present a new approach to optical coherence elastography (OCE), which probes the local elastic properties of tissue by using optical coherence tomography to measure the effect of an applied stimulus in the audio frequency range. We describe the approach, based on analysis of the Bessel frequency spectrum of the interferometric signal detected from scatterers undergoing periodic motion in response to an applied stimulus. We present quantitative results of sub-micron excitation at 820 Hz in a layered phantom and the first such measurements in human skin in vivo.

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.

[Voix d'Or, an audio tool to revive memories].

Science.gov (United States)

Braunschweig, Lina

2010-01-01

Voix d'Or is an audio tool designed to awaken the affective memory of elderly people and particularly those suffering from Alzheimer's disease. Every month it offers new radio programmes to initiate or facilitate leisure and entertainment activities, memory workshops or provide the basis of quiet moments. The tool has a double objective: to procure well-being, boost the individual's self-esteem and recognise his/her history and to facilitate exchange and communication between the residents and the staff of a care home.

Data communication equipment

International Nuclear Information System (INIS)

Kim, Hak Seon; Lee, Sang Mok

1998-02-01

The contents of this book are introduction of data communication on definition, purpose and history, information terminal about data communication system and data transmission system, data transmit equipment of summary, transmission cable, data port, concentrator and front-end processor, audio communication equipment like phones, radio communication equipment of summary on foundation of electromagnetic waves, AM transmitter, AM receiver, FM receiver and FM transmitter, a satellite and mobile communication equipment such as earth station, TT and C and Cellular phone, video telephone and new media apparatus.

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...

Hysteretic self-oscillating bandpass current mode control for Class D audio amplifiers driving capacitive transducers

DEFF Research Database (Denmark)

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

2013-01-01

A hysteretic self-oscillating bandpass current mode control (BPCM) scheme for Class D audio amplifiers driving capacitive transducers are presented. The scheme provides excellent stability margins and low distortion over a wide range of operating conditions. Small-signal behavior of the amplifier...... the rules of electrostatics have been known as very interesting alternatives to the traditional inefficient electrodynamic transducers. When driving capacitive transducers from a Class D audio amplifier the high impedance nature of the load represents a key challenge. The BPCM control scheme ensures a flat...

A scheme for racquet sports video analysis with the combination of audio-visual information

Science.gov (United States)

Xing, Liyuan; Ye, Qixiang; Zhang, Weigang; Huang, Qingming; Yu, Hua

2005-07-01

As a very important category in sports video, racquet sports video, e.g. table tennis, tennis and badminton, has been paid little attention in the past years. Considering the characteristics of this kind of sports video, we propose a new scheme for structure indexing and highlight generating based on the combination of audio and visual information. Firstly, a supervised classification method is employed to detect important audio symbols including impact (ball hit), audience cheers, commentator speech, etc. Meanwhile an unsupervised algorithm is proposed to group video shots into various clusters. Then, by taking advantage of temporal relationship between audio and visual signals, we can specify the scene clusters with semantic labels including rally scenes and break scenes. Thirdly, a refinement procedure is developed to reduce false rally scenes by further audio analysis. Finally, an exciting model is proposed to rank the detected rally scenes from which many exciting video clips such as game (match) points can be correctly retrieved. Experiments on two types of representative racquet sports video, table tennis video and tennis video, demonstrate encouraging results.

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.

Audio-visual identification of place of articulation and voicing in white and babble noise.

Science.gov (United States)

Alm, Magnus; Behne, Dawn M; Wang, Yue; Eg, Ragnhild

2009-07-01

Research shows that noise and phonetic attributes influence the degree to which auditory and visual modalities are used in audio-visual speech perception (AVSP). Research has, however, mainly focused on white noise and single phonetic attributes, thus neglecting the more common babble noise and possible interactions between phonetic attributes. This study explores whether white and babble noise differentially influence AVSP and whether these differences depend on phonetic attributes. White and babble noise of 0 and -12 dB signal-to-noise ratio were added to congruent and incongruent audio-visual stop consonant-vowel stimuli. The audio (A) and video (V) of incongruent stimuli differed either in place of articulation (POA) or voicing. Responses from 15 young adults show that, compared to white noise, babble resulted in more audio responses for POA stimuli, and fewer for voicing stimuli. Voiced syllables received more audio responses than voiceless syllables. Results can be attributed to discrepancies in the acoustic spectra of both the noise and speech target. Voiced consonants may be more auditorily salient than voiceless consonants which are more spectrally similar to white noise. Visual cues contribute to identification of voicing, but only if the POA is visually salient and auditorily susceptible to the noise type.

Digital communication communication, multimedia, security

CERN Document Server

Meinel, Christoph

2014-01-01

The authors give a detailed summary about the fundamentals and the historical background of digital communication. This includes an overview of the encoding principles and algorithms of textual information, audio information, as well as images, graphics, and video in the Internet. Furthermore the fundamentals of computer networking, digital security and cryptography are covered. Thus, the book provides a well-founded access to communication technology of computer networks, the internet and the WWW. Numerous pictures and images, a subject-index and a detailed list of historical personalities in

[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.

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...

Impact of a bone conduction communication channel on multichannel communication system effectiveness.

Science.gov (United States)

Blue, Misty; McBride, Maranda; Weatherless, Rachel; Letowski, Tomasz

2013-04-01

In this study, the impact of including a bone conduction transducer in a three-channel spatialized communication system was investigated. Several military and security forces situations require concurrent listening to three or more radio channels. In such radio systems, spatial separation between three concurrent radio channels can be achieved by delivering separate signals to the left and right earphone independently and both earphones simultaneously. This method appears to be effective; however, the use of bone conduction as one channel may provide both operational and performance benefits. Three three-channel communication systems were used to collect speech intelligibility data from 18 listeners (System I, three loudspeakers; System 2, stereo headphones; System 3, stereo headphones and a bone conduction vibrator). Each channel presented signals perceived to originate from separate locations. Volunteers listened to three sets of competing sentences and identified a number, color, and object spoken in the target sentence. Each listener participated in three trials (one per system). Each trial consisted of 48 competing sentence sets. Systems 2 and 3 were more intelligible than System I. Systems 2 and 3 were overall equally intelligible; however, the intelligibility of all three channels was significantly more balanced in System 3. Replacing an air conduction transducer with a bone conduction transducer in a multichannel audio device can provide a more effective and balanced simultaneous monitoring auditory environment. These results have important design and implementation implications for spatial auditory communication equipment.

Design guidelines for the use of audio cues in computer interfaces

Energy Technology Data Exchange (ETDEWEB)

Sumikawa, D.A.; Blattner, M.M.; Joy, K.I.; Greenberg, R.M.

1985-07-01

A logical next step in the evolution of the computer-user interface is the incorporation of sound thereby using our senses of ''hearing'' in our communication with the computer. This allows our visual and auditory capacities to work in unison leading to a more effective and efficient interpretation of information received from the computer than by sight alone. In this paper we examine earcons, which are audio cues, used in the computer-user interface to provide information and feedback to the user about computer entities (these include messages and functions, as well as states and labels). The material in this paper is part of a larger study that recommends guidelines for the design and use of audio cues in the computer-user interface. The complete work examines the disciplines of music, psychology, communication theory, advertising, and psychoacoustics to discover how sound is utilized and analyzed in those areas. The resulting information is organized according to the theory of semiotics, the theory of signs, into the syntax, semantics, and pragmatics of communication by sound. Here we present design guidelines for the syntax of earcons. Earcons are constructed from motives, short sequences of notes with a specific rhythm and pitch, embellished by timbre, dynamics, and register. Compound earcons and family earcons are introduced. These are related motives that serve to identify a family of related cues. Examples of earcons are given.

A Novel Chewing Detection System Based on PPG, Audio, and Accelerometry.

Science.gov (United States)

Papapanagiotou, Vasileios; Diou, Christos; Zhou, Lingchuan; van den Boer, Janet; Mars, Monica; Delopoulos, Anastasios

2017-05-01

In the context of dietary management, accurate monitoring of eating habits is receiving increased attention. Wearable sensors, combined with the connectivity and processing of modern smartphones, can be used to robustly extract objective and real-time measurements of human behavior. In particular, for the task of chewing detection, several approaches based on an in-ear microphone can be found in the literature, while other types of sensors have also been reported, such as strain sensors. In this paper, performed in the context of the SPLENDID project, we propose to combine an in-ear microphone with a photoplethysmography (PPG) sensor placed in the ear concha, in a new high accuracy and low sampling rate prototype chewing detection system. We propose a pipeline that initially processes each sensor signal separately, and then fuses both to perform the final detection. Features are extracted from each modality, and support vector machine (SVM) classifiers are used separately to perform snacking detection. Finally, we combine the SVM scores from both signals in a late-fusion scheme, which leads to increased eating detection accuracy. We evaluate the proposed eating monitoring system on a challenging, semifree living dataset of 14 subjects, which includes more than 60 h of audio and PPG signal recordings. Results show that fusing the audio and PPG signals significantly improves the effectiveness of eating event detection, achieving accuracy up to 0.938 and class-weighted accuracy up to 0.892.

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.

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...

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...

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

Science.gov (United States)

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

2013-01-01

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

Visual communication and the content and style of conversation.

Science.gov (United States)

Rutter, D R; Stephenson, G M; Dewey, M E

1981-02-01

Previous research suggests that visual communication plays a number of important roles in social interaction. In particular, it appears to influence the content of what people say in discussions, the style of their speech, and the outcomes they reach. However, the findings are based exclusively on comparisons between face-to-face conversations and audio conversations, in which subjects sit in separate rooms and speak over a microphone-headphone intercom which precludes visual communication. Interpretation is difficult, because visual communication is confounded with physical presence, which itself makes available certain cues denied to audio subjects. The purpose of this paper is to report two experiments in which the variables were separated and content and style were re-examined. The first made use of blind subjects, and again compared the face-to-face and audio conditions. The second returned to sighted subjects, and examined four experimental conditions: face-to-face; audio; a curtain condition in which subjects sat in the same room but without visual communication; and a video condition in which they sat in separate rooms and communicated over a television link. Neither visual communication nor physical presence proved to be critical variable. Instead, the two sources of cues combined, such that content and style were influenced by the aggregate of available cues. The more cueless the settings, the more task-oriented, depersonalized and unspontaneous the conversation. The findings also suggested that the primary effect of cuelessness is to influence verbal content, and that its influence on both style and outcome occurs indirectly, through the mediation of content.

Computerized J-H loop tracer for soft magnetic thick films in the audio frequency range

Directory of Open Access Journals (Sweden)

Loizos G.

2014-07-01

Full Text Available A computerized J-H loop tracer for soft magnetic thick films in the audio frequency range is described. It is a system built on a PXI platform combining PXI modules for control signal generation and data acquisition. The physiscal signals are digitized and the respective data strems are processed, presented and recorded in LabVIEW 7.0.

Visualising the environmental appearance of audio products

Energy Technology Data Exchange (ETDEWEB)

Stilma, M. [Univ. of Twente, Enschede (Netherlands); Stevels, A. [Delft Univ. of Technology, Delft (Netherlands)]|[Philips Consumer Electronics, Eindhoven (Netherlands); Christiaans, H.; Kandachar, P. [Delft Univ. of Technology, Delft (Netherlands)

2004-07-01

Can environmental friendliness be communicated by the design style and appearance of products? (such as form, colour, style or material)? Consumers are interested in buying environmental products and design styles might be used as communicative tools. However, current 'green' products show something else. Environmental aspects are chiefly promoted by marketing programs based on technical items like the use of materials, hazardous substances, energy consumption, etc. By a qualitative and exploratory research the environmental design styles according to consumers' opinions were analysed with larger audio products as case study. Visible distinctive differences can be identified between the most and the least environmental rated products. A 'Green flagship', which claims to be environmentally orientated, wasn't recognised as such by consumers. And women and men perceive environmental friendliness in another way. From this research can be concluded that more attention is needed to visualise the good technical environmental performance of products. (orig.)

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.

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

Web Audio/Video Streaming Tool

Science.gov (United States)

Guruvadoo, Eranna K.

2003-01-01

In order to promote NASA-wide educational outreach program to educate and inform the public of space exploration, NASA, at Kennedy Space Center, is seeking efficient ways to add more contents to the web by streaming audio/video files. This project proposes a high level overview of a framework for the creation, management, and scheduling of audio/video assets over the web. To support short-term goals, the prototype of a web-based tool is designed and demonstrated to automate the process of streaming audio/video files. The tool provides web-enabled users interfaces to manage video assets, create publishable schedules of video assets for streaming, and schedule the streaming events. These operations are performed on user-defined and system-derived metadata of audio/video assets stored in a relational database while the assets reside on separate repository. The prototype tool is designed using ColdFusion 5.0.

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.

Exploring Meaning Negotiation Patterns in Synchronous Audio and Video Conferencing English Classes in China

Science.gov (United States)

Li, Chenxi; Wu, Ligao; Li, Chen; Tang, Jinlan

2017-01-01

This work-in-progress doctoral research project aims to identify meaning negotiation patterns in synchronous audio and video Computer-Mediated Communication (CMC) environments based on the model of CMC text chat proposed by Smith (2003). The study was conducted in the Institute of Online Education at Beijing Foreign Studies University. Four dyads…

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; ·     �...

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...

Intensity position modulation for free-space laser communication system

Science.gov (United States)

Jangjoo, Alireza; Faghihi, F.

2004-12-01

In this research a novel modulation technique for free-space laser communication system called Intensity Position Modulation (IPM) is carried out. According to TEM00 mode of a laser beam and by linear fitting on the Gaussian function as an approximation, the variation of linear part on the reverse biased pn photodiode produced alternating currents which contain the information. Here, no characteristic property of the beam as intensity or frequency is changed and only the beam position moves laterally. We demonstrated that in this method no bandwidth is required, so it is possible to reduce the background radiation noise by narrowband filtering of the carrier. The fidelity of the analog voice communication system which is made upon the IPM is satisfactory and we are able to transmit the audio signals up to 1Km.

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...

Communicative Signals Promote Object Recognition Memory and Modulate the Right Posterior STS.

Science.gov (United States)

Redcay, Elizabeth; Ludlum, Ruth S; Velnoskey, Kayla R; Kanwal, Simren

2016-01-01

Detection of communicative signals is thought to facilitate knowledge acquisition early in life, but less is known about the role these signals play in adult learning or about the brain systems supporting sensitivity to communicative intent. The current study examined how ostensive gaze cues and communicative actions affect adult recognition memory and modulate neural activity as measured by fMRI. For both the behavioral and fMRI experiments, participants viewed a series of videos of an actress acting on one of two objects in front of her. Communicative context in the videos was manipulated in a 2 × 2 design in which the actress either had direct gaze (Gaze) or wore a visor (NoGaze) and either pointed at (Point) or reached for (Reach) one of the objects (target) in front of her. Participants then completed a recognition memory task with old (target and nontarget) objects and novel objects. Recognition memory for target objects in the Gaze conditions was greater than NoGaze, but no effects of gesture type were seen. Similarly, the fMRI video-viewing task revealed a significant effect of Gaze within right posterior STS (pSTS), but no significant effects of Gesture. Furthermore, pSTS sensitivity to Gaze conditions was related to greater memory for objects viewed in Gaze, as compared with NoGaze, conditions. Taken together, these results demonstrate that the ostensive, communicative signal of direct gaze preceding an object-directed action enhances recognition memory for attended items and modulates the pSTS response to object-directed actions. Thus, establishment of a communicative context through ostensive signals remains an important component of learning and memory into adulthood, and the pSTS may play a role in facilitating this type of social learning.

Reconstruction of chaotic signals with applications to chaos-based communications

CERN Document Server

Feng, Jiu Chao

2008-01-01

This book provides a systematic review of the fundamental theory of signal reconstruction and the practical techniques used in reconstructing chaotic signals. Specific applications of signal reconstruction methods in chaos-based communications are expounded in full detail, along with examples illustrating the various problems associated with such applications.The book serves as an advanced textbook for undergraduate and graduate courses in electronic and information engineering, automatic control, physics and applied mathematics. It is also highly suited for general nonlinear scientists who wi

Improving Communicative Competence through Synchronous Communication in Computer-Supported Collaborative Learning Environments: A Systematic Review

OpenAIRE

Xi Huang

2018-01-01

Computer-supported collaborative learning facilitates the extension of second language acquisition into social practice. Studies on its achievement effects speak directly to the pedagogical notion of treating communicative practice in synchronous computer-mediated communication (SCMC): real-time communication that takes place between human beings via the instrumentality of computers in forms of text, audio and video communication, such as live chat and chatrooms as socially-oriented meaning c...

Shaping communicative colour signals over evolutionary time

Science.gov (United States)

Oyola Morales, José R.; Vital-García, Cuauhcihuatl; Hews, Diana K.; Martins, Emília P.

2016-01-01

Many evolutionary forces can shape the evolution of communicative signals, and the long-term impact of each force may depend on relative timing and magnitude. We use a phylogenetic analysis to infer the history of blue belly patches of Sceloporus lizards, and a detailed spectrophotometric analysis of four species to explore the specific forces shaping evolutionary change. We find that the ancestor of Sceloporus had blue patches. We then focus on four species; the first evolutionary shift (captured by comparison of S. merriami and S. siniferus) represents an ancient loss of the belly patch by S. siniferus, and the second evolutionary shift, bounded by S. undulatus and S. virgatus, represents a more recent loss of blue belly patch by S. virgatus. Conspicuousness measurements suggest that the species with the recent loss (S. virgatus) is the least conspicuous. Results for two other species (S. siniferus and S. merriami) suggest that over longer periods of evolutionary time, new signal colours have arisen which minimize absolute contrast with the habitat while maximizing conspicuousness to a lizard receiver. Specifically, males of the species representing an ancient loss of blue patch (S. siniferus) are more conspicuous than are females in the UV, whereas S. merriami males have evolved a green element that makes their belly patches highly sexually dimorphic but no more conspicuous than the white bellies of S. merriami females. Thus, our results suggest that natural selection may act more immediately to reduce conspicuousness, whereas sexual selection may have a more complex impact on communicative signals through the introduction of new colours. PMID:28018661

Synthesis of audio spectra using a diffraction model.

Science.gov (United States)

Vijayakumar, V; Eswaran, C

2006-12-01

It is shown that the intensity variations of an audio signal in the frequency domain can be obtained by using a mathematical function containing a series of weighted complex Bessel functions. With proper choice of values for two parameters, this function can transform an input spectrum of discrete frequencies of unit intensity into the known spectra of different musical instruments. Specific examples of musical instruments are considered for evaluating the performance of this method. It is found that this function yields musical spectra with a good degree of accuracy.

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.

[Audio-visual communication in the history of psychiatry].

Science.gov (United States)

Farina, B; Remoli, V; Russo, F

1993-12-01

The authors analyse the evolution of visual communication in the history of psychiatry. From the 18th century oil paintings to the first dagherrotic prints until the cinematography and the modern audiovisual systems they observed an increasing diffusion of the new communication techniques in psychiatry, and described the use of the different techniques in psychiatric practice. The article ends with a brief review of the current applications of the audiovisual in therapy, training, teaching, and research.

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...

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

Intrateam Communication and Performance in Doubles Tennis

Science.gov (United States)

Lausic, Domagoj; Tennebaum, Gershon; Eccles, David; Jeong, Allan; Johnson, Tristan

2009-01-01

Verbal and nonverbal communication is a critical mediator of performance in team sports and yet there is little extant research in sports that involves direct measures of communication. Our study explored communication within NCAA Division I female tennis doubles teams. Video and audio recordings of players during doubles tennis matches captured…

NASA Bluetooth Wireless Communications

Science.gov (United States)

Miller, Robert D.

2007-01-01

NASA has been interested in wireless communications for many years, especially when the crew size of the International Space Station (ISS) was reduced to two members. NASA began a study to find ways to improve crew efficiency to make sure the ISS could be maintained with limited crew capacity and still be a valuable research testbed in Low-Earth Orbit (LEO). Currently the ISS audio system requires astronauts to be tethered to the audio system, specifically a device called the Audio Terminal Unit (ATU). Wireless communications would remove the tether and allow astronauts to freely float from experiment to experiment without having to worry about moving and reconnecting the associated cabling or finding the space equivalent of an extension cord. A wireless communication system would also improve safety and reduce system susceptibility to Electromagnetic Interference (EMI). Safety would be improved because a crewmember could quickly escape a fire while maintaining communications with the ground and other crewmembers at any location. In addition, it would allow the crew to overcome the volume limitations of the ISS ATU. This is especially important to the Portable Breathing Apparatus (PBA). The next generation of space vehicles and habitats also demand wireless attention. Orion will carry up to six crewmembers in a relatively small cabin. Yet, wireless could become a driving factor to reduce launch weight and increase habitable volume. Six crewmembers, each tethered to a panel, could result in a wiring mess even in nominal operations. In addition to Orion, research is being conducted to determine if Bluetooth is appropriate for Lunar Habitat applications.

Effect of Ionosphere on Geostationary Communication Satellite Signals

Science.gov (United States)

Erdem, Esra; Arikan, Feza; Gulgonul, Senol

2016-07-01

Geostationary orbit (GEO) communications satellites allow radio, television, and telephone transmissions to be sent live anywhere in the world. They are extremely important in daily life and also for military applications. Since, satellite communication is an expensive technology addressing crowd of people, it is critical to improve the performance of this technology. GEO satellites are at 35,786 kilometres from Earth's surface situated directly over the equator. A satellite in a geostationary orbit (GEO) appears to stand still in the sky, in a fixed position with respect to an observer on the earth, because the satellite's orbital period is the same as the rotation rate of the Earth. The advantage of this orbit is that ground antennas can be fixed to point towards to satellite without their having to track the satellite's motion. Radio frequency ranges used in satellite communications are C, X, Ku, Ka and even EHG and V-band. Satellite signals are disturbed by atmospheric effects on the path between the satellite and the receiver antenna. These effects are mostly rain, cloud and gaseous attenuation. It is expected that ionosphere has a minor effect on the satellite signals when the ionosphere is quiet. But there are anomalies and perturbations on the structure of ionosphere with respect to geomagnetic field and solar activity and these conditions may cause further affects on the satellite signals. In this study IONOLAB-RAY algorithm is adopted to examine the effect of ionosphere on satellite signals. IONOLAB-RAY is developed to calculate propagation path and characteristics of high frequency signals. The algorithm does not have any frequency limitation and models the plasmasphere up to 20,200 km altitude, so that propagation between a GEO satellite and antenna on Earth can be simulated. The algorithm models inhomogeneous, anisotropic and time dependent structure of the ionosphere with a 3-D spherical grid geometry and calculates physical parameters of the

Organelle communication: signaling crossroads between homeostasis and disease.

Science.gov (United States)

Bravo-Sagua, Roberto; Torrealba, Natalia; Paredes, Felipe; Morales, Pablo E; Pennanen, Christian; López-Crisosto, Camila; Troncoso, Rodrigo; Criollo, Alfredo; Chiong, Mario; Hill, Joseph A; Simmen, Thomas; Quest, Andrew F; Lavandero, Sergio

2014-05-01

Cellular organelles do not function as isolated or static units, but rather form dynamic contacts between one another that can be modulated according to cellular needs. The physical interfaces between organelles are important for Ca2+ and lipid homeostasis, and serve as platforms for the control of many essential functions including metabolism, signaling, organelle integrity and execution of the apoptotic program. Emerging evidence also highlights the importance of organelle communication in disorders such as Alzheimer's disease, pulmonary arterial hypertension, cancer, skeletal and cardiac muscle dysfunction. Here, we provide an overview of the current literature on organelle communication and the link to human pathologies. Copyright © 2014 Elsevier Ltd. All rights reserved.

New audio applications of beryllium metal

International Nuclear Information System (INIS)

Sato, M.

1977-01-01

The major applications of beryllium metal in the field of audio appliances are for the vibrating cones for the two types of speakers 'TWITTER' for high range sound and 'SQUAWKER' for mid range sound, and also for beryllium cantilever tube assembled in stereo cartridge. These new applications are based on the characteristic property of beryllium having high ratio of modulus of elasticity to specific gravity. The production of these audio parts is described, and the audio response is shown. (author)

Aggressive Bimodal Communication in Domestic Dogs, Canis familiaris.

Science.gov (United States)

Déaux, Éloïse C; Clarke, Jennifer A; Charrier, Isabelle

2015-01-01

Evidence of animal multimodal signalling is widespread and compelling. Dogs' aggressive vocalisations (growls and barks) have been extensively studied, but without any consideration of the simultaneously produced visual displays. In this study we aimed to categorize dogs' bimodal aggressive signals according to the redundant/non-redundant classification framework. We presented dogs with unimodal (audio or visual) or bimodal (audio-visual) stimuli and measured their gazing and motor behaviours. Responses did not qualitatively differ between the bimodal and two unimodal contexts, indicating that acoustic and visual signals provide redundant information. We could not further classify the signal as 'equivalent' or 'enhancing' as we found evidence for both subcategories. We discuss our findings in relation to the complex signal framework, and propose several hypotheses for this signal's function.

Communications and media services

Science.gov (United States)

Mcculla, James W.; Kukowski, James F.

1990-01-01

NASA's internal and external communication methods are reviewed. NASA information services for the media, for the public, and for employees are discussed. Consideration is given to electron information distribution, the NASA TV-audio system, the NASA broadcast news service, astronaut appearances, technology and information exhibits, speaker services, and NASA news reports for internal communications. Also, the NASA worldwide electronic mail network is described and trends for future NASA communications and media services are outlined.

DOA and Pitch Estimation of Audio Sources using IAA-based Filtering

DEFF Research Database (Denmark)

Jensen, Jesper Rindom; Christensen, Mads Græsbøll

2014-01-01

For decades, it has been investigated how to separately solve the problems of both direction-of-arrival (DOA) and pitch estimation. Recently, it was found that estimating these parameters jointly from multichannel recordings of audio can be extremely beneficial. Many joint estimators are based...... on knowledge of the inverse sample covariance matrix. Typically, this covariance is estimated using the sample covariance matrix, but for this estimate to be full rank, many temporal samples are needed. In cases with non-stationary signals, this is a serious limitation. We therefore investigate how a recent...... joint DOA and pitch filtering-based estimator can be combined with the iterative adaptive approach to circumvent this limitation in joint DOA and pitch estimation of audio sources. Simulations show a clear improvement compared to when using the sample covariance matrix and the considered approach also...

Evaluating the Peruvian Rural Communication Services Project.

Science.gov (United States)

Mayo, John

1988-01-01

Reviews the Peruvian Rural Communication Services (PRCS) Project and outlines selected findings. Topics discussed include a brief description of Peru's economic and social conditions; satellite communication systems; audio teleconferencing; telephone service; planning and administration; research design features; data collection; and project…

A description of communication patterns during CPR in ICU.

Science.gov (United States)

Taylor, Katherine L; Ferri, Susan; Yavorska, Tatyana; Everett, Tobias; Parshuram, Christopher

2014-10-01

Deficiencies in communication in health care are a common source of medical error. Preferred communication patterns are a component of resuscitation teaching. We audio-recorded resuscitations in a mixed paediatric medical and surgical ICU to describe communication. In the intensive care unit, resuscitation events were prospectively audio-recorded by two trained observers (using handheld recorders). Recordings were transcribed and anonymised within 24h. We grouped utterances regarding the same subject matter from beginning (irrespective of response) as a communication epoch. For each epoch, we describe the initiator, audience and content of message. Teamwork behaviours were described using Anesthesia Nontechnical Skills framework (ANTS), a behavioural marker system for crisis-resource management. Consent rates from staff were 139/140 (99%) and parents were 67/92 (73%). We analysed 36min 57s of audio dialogue from 4 cardiac arrest events in 363h of prospective screening. There were 180 communication epochs (1 every 12s): 100 (56%) from the team-leader and 80 (44%) from non-team-leader(s). Team-leader epochs were to give or confirm orders or assert authority (61%), clarify patient history (14%) and provide clinical updates (25%). Non-team-leader epochs were more often directed to the team (65%) than the team-leader (35%). Audio-recordings provided information for 80% of the ANTS component elements with scores of 2-4. Communication epochs were frequent, most from the team-leader. We identified an 'outer loop' of communication between team members not including the team-leader, responsible for 44% of all communication events. We discuss difficulties in this research methodology. Future work includes exploring the process of the 'outer loop' by resuscitation team members to evaluate the optimal balance between single leader and team suggestions, the content of the outer loop discussions and in-event communication strategies to improve outcomes. Crown Copyright © 2014

Programmable delay circuit for sparker signal analysis

Digital Repository Service at National Institute of Oceanography (India)

Pathak, D.

The sparker echo signal had been recorded along with the EPC recorder trigger on audio cassettes in a dual channel analog recorder. The sparker signal in the analog form had to be digitised for further signal processing techniques to be performed...

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....

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.

Adenylate Kinase and AMP Signaling Networks: Metabolic Monitoring, Signal Communication and Body Energy Sensing

Directory of Open Access Journals (Sweden)

Andre Terzic

2009-04-01

Full Text Available Adenylate kinase and downstream AMP signaling is an integrated metabolic monitoring system which reads the cellular energy state in order to tune and report signals to metabolic sensors. A network of adenylate kinase isoforms (AK1-AK7 are distributed throughout intracellular compartments, interstitial space and body fluids to regulate energetic and metabolic signaling circuits, securing efficient cell energy economy, signal communication and stress response. The dynamics of adenylate kinase-catalyzed phosphotransfer regulates multiple intracellular and extracellular energy-dependent and nucleotide signaling processes, including excitation-contraction coupling, hormone secretion, cell and ciliary motility, nuclear transport, energetics of cell cycle, DNA synthesis and repair, and developmental programming. Metabolomic analyses indicate that cellular, interstitial and blood AMP levels are potential metabolic signals associated with vital functions including body energy sensing, sleep, hibernation and food intake. Either low or excess AMP signaling has been linked to human disease such as diabetes, obesity and hypertrophic cardiomyopathy. Recent studies indicate that derangements in adenylate kinase-mediated energetic signaling due to mutations in AK1, AK2 or AK7 isoforms are associated with hemolytic anemia, reticular dysgenesis and ciliary dyskinesia. Moreover, hormonal, food and antidiabetic drug actions are frequently coupled to alterations of cellular AMP levels and associated signaling. Thus, by monitoring energy state and generating and distributing AMP metabolic signals adenylate kinase represents a unique hub within the cellular homeostatic network.

Classroom Audio Distribution in the Postsecondary Setting: A Story of Universal Design for Learning

Science.gov (United States)

Flagg-Williams, Joan B.; Bokhorst-Heng, Wendy D.

2016-01-01

Classroom Audio Distribution Systems (CADS) consist of amplification technology that enhances the teacher's, or sometimes the student's, vocal signal above the background noise in a classroom. Much research has supported the benefits of CADS for student learning, but most of it has focused on elementary school classrooms. This study investigated…

Radiation-induced perturbation of cell-to-cell signalling and communication

International Nuclear Information System (INIS)

Mariotti, L.; Facoetti, A.; Bertolotti, A.; Ranza, E.; Alloni, D.; Ottolenghi, A.

2011-01-01

The investigation of the bystander phenomena (i.e. the induction of damage in cells not directly traversed by radiation) is strictly related to the study of the mechanisms of intercellular communication and of the perturbative effects of radiation. A new possible way to try to solve the bystander puzzle is through a 'systems radiation biology' approach with the total integration of experimental and theoretical activities. In particular, this contribution will focus on: (1) 'ad hoc' experiments designed to quantify key parameters involved in intercellular signalling (focusing, as a pilot study, on release, decay and internalization of interleukin-6 molecules, their modulation by radiation, and possible differences between in vivo/in vitro behaviour); (2) the implementation and the development of two different modelling approaches: a stochastic model (based on a Monte Carlo code) that takes account of the local mechanisms of release and internalization of signalling molecules (e.g. cytokines) and an analytical model where signal molecules are treated as a population and their temporal behaviour is described by differential equations. This approach provided instruments to investigate the complex phenomena of signal transmission and the role of cell communication to guarantee (maintain) the robustness of the in vitro experimental systems against the effects of perturbations. (authors)

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.

Aggressive Bimodal Communication in Domestic Dogs, Canis familiaris.

Directory of Open Access Journals (Sweden)

Éloïse C Déaux

Full Text Available Evidence of animal multimodal signalling is widespread and compelling. Dogs' aggressive vocalisations (growls and barks have been extensively studied, but without any consideration of the simultaneously produced visual displays. In this study we aimed to categorize dogs' bimodal aggressive signals according to the redundant/non-redundant classification framework. We presented dogs with unimodal (audio or visual or bimodal (audio-visual stimuli and measured their gazing and motor behaviours. Responses did not qualitatively differ between the bimodal and two unimodal contexts, indicating that acoustic and visual signals provide redundant information. We could not further classify the signal as 'equivalent' or 'enhancing' as we found evidence for both subcategories. We discuss our findings in relation to the complex signal framework, and propose several hypotheses for this signal's function.

Signal Processing

International Nuclear Information System (INIS)

Anon.

1992-01-01

Signal processing techniques, extensively used nowadays to maximize the performance of audio and video equipment, have been a key part in the design of hardware and software for high energy physics detectors since pioneering applications in the UA1 experiment at CERN in 1979

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.

Molecular diagnostics reveal spiders that exploit prey vibrational signals used in sexual communication.

Science.gov (United States)

Virant-Doberlet, Meta; King, R Andrew; Polajnar, Jernej; Symondson, William O C

2011-05-01

Vibrational signalling is a widespread form of animal communication and, in the form of sexual communication, has been generally regarded as inherently short-range and a private communication channel, free from eavesdropping by generalist predators. A combination of fieldwork and laboratory experiments was used to test the hypothesis that predators can intercept and exploit such signals. First, we developed and characterized PCR primers specific for leafhoppers of the genus Aphrodes and specifically for the species Aphrodes makarovi. Spiders were collected from sites where leafhoppers were present and screened with these primers to establish which spider species were significant predators of this species during the mating period of these leafhoppers. Analysis using PCR of the gut contents of tangle-web spiders, Enoplognatha ovata (Theridiidae), showed that they consume leafhoppers in the field at a greater rate when signalling adults were present than when nymphs were dominant, suggesting that the spiders were using these vibrations signals to find their prey. Playback and microcosm experiments then showed that E. ovata can use the vibrational signals of male leafhoppers as a cue during foraging and, as a result, killed significantly more male than female A. makarovi. Our results show, for the first time, that arthropod predators can exploit prey vibrational communication to obtain information about prey availability and use this information to locate and capture prey. This may be a widespread mechanism for prey location, one that is likely to be a major unrecognized driver of evolution in both predators and prey. © 2011 Blackwell Publishing Ltd.

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

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.

Male Diaphorina citri searching responses to vibrational communication signals

Science.gov (United States)

Prototype devices have been developed that mimic D. citri female replies to male communication signals and lure males to a trap. The objective is to trap a high proportion of males that have landed on a host tree and have begun searching for females. This presentation describes the construction and ...

Intercellular signal communication among odontoblasts and trigeminal ganglion neurons via glutamate.

Science.gov (United States)

Nishiyama, A; Sato, M; Kimura, M; Katakura, A; Tazaki, M; Shibukawa, Y

2016-11-01

Various stimuli to the exposed surface of dentin induce changes in the hydrodynamic force inside the dentinal tubules resulting in dentinal pain. Recent evidences indicate that mechano-sensor channels, such as the transient receptor potential channels, in odontoblasts receive these hydrodynamic forces and trigger the release of ATP to the pulpal neurons, to generate dentinal pain. A recent study, however, has shown that odontoblasts also express glutamate receptors (GluRs). This implies that cells in the dental pulp tissue have the ability to release glutamate, which acts as a functional intercellular mediator to establish inter-odontoblast and odontoblast-trigeminal ganglion (TG) neuron signal communication. To investigate the intercellular signal communication, we applied mechanical stimulation to odontoblasts and measured the intracellular free Ca 2+ concentration ([Ca 2+ ] i ). During mechanical stimulation in the presence of extracellular Ca 2+ , we observed a transient [Ca 2+ ] i increase not only in single stimulated odontoblasts, but also in adjacent odontoblasts. We could not observe these responses in the absence of extracellular Ca 2+ . [Ca 2+ ] i increases in the neighboring odontoblasts during mechanical stimulation of single odontoblasts were inhibited by antagonists of metabotropic glutamate receptors (mGluRs) as well as glutamate-permeable anion channels. In the odontoblast-TG neuron coculture, we observed an increase in [Ca 2+ ] i in the stimulated odontoblasts and TG neurons, in response to direct mechanical stimulation of single odontoblasts. These [Ca 2+ ] i increases in the neighboring TG neurons were inhibited by antagonists for mGluRs. The [Ca 2+ ] i increases in the stimulated odontoblasts were also inhibited by mGluRs antagonists. We further confirmed that the odontoblasts express group I, II, and III mGluRs. However, we could not record any currents evoked from odontoblasts near the mechanically stimulated odontoblast, with or without

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.

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.

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...

The effect of visual information on verbal communication process in remote conversation

OpenAIRE

國田, 祥子; 中條, 和光

2005-01-01

This article examined how visual information affects verbal communication process in remote communication. In the experiment twenty pairs of subjects performed a collaborative task remotely via video and audio links or audio link only. During the task used in this experiment one of a pair (an instruction-giver) gave direction with a map to the other of the pair (an instruction-receiver). We recorded and analyzed contents of utterances. Consequently, the existence of visual information did not...

Honest signaling and oxidative stress: the special case of avian acoustic communication

Directory of Open Access Journals (Sweden)

Stefania eCasagrande

2016-05-01

Full Text Available Much research on animal communication has addressed how costs or constraints determined by the oxidative status of an individual can assure the honesty of visual signals, such as sexually selected color ornaments. However, acoustic communication has been largely overlooked in this respect. Here, we describe the few available studies that have considered the role of oxidative status in mediating vocal behavior in adult and nestling birds. Further, we discuss the theoretical principles of how the honesty of avian acoustic signals may be maintained by an organism’s oxidative status. We here distinguish between studies that considered songs and begging calls as indicators of oxidative status and studies where vocalizations were assumed to be the source of oxidative costs. We outline experimental and methodological issues related to the study of bird vocalizations and oxidative stress and describe opportunities for future work in this field of research. Investigating the interactions between acoustic signals and redox state may help address some unresolved questions in avian vocalization, thereby increasing our understanding of the evolutionary pressures shaping animal communication. Finally, we argue that it will be important to extend this line of research beyond birds and include other taxa as well.

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

Digital nonlinearity compensation in high-capacity optical communication systems considering signal spectral broadening effect.

Science.gov (United States)

Xu, Tianhua; Karanov, Boris; Shevchenko, Nikita A; Lavery, Domaniç; Liga, Gabriele; Killey, Robert I; Bayvel, Polina

2017-10-11

Nyquist-spaced transmission and digital signal processing have proved effective in maximising the spectral efficiency and reach of optical communication systems. In these systems, Kerr nonlinearity determines the performance limits, and leads to spectral broadening of the signals propagating in the fibre. Although digital nonlinearity compensation was validated to be promising for mitigating Kerr nonlinearities, the impact of spectral broadening on nonlinearity compensation has never been quantified. In this paper, the performance of multi-channel digital back-propagation (MC-DBP) for compensating fibre nonlinearities in Nyquist-spaced optical communication systems is investigated, when the effect of signal spectral broadening is considered. It is found that accounting for the spectral broadening effect is crucial for achieving the best performance of DBP in both single-channel and multi-channel communication systems, independent of modulation formats used. For multi-channel systems, the degradation of DBP performance due to neglecting the spectral broadening effect in the compensation is more significant for outer channels. Our work also quantified the minimum bandwidths of optical receivers and signal processing devices to ensure the optimal compensation of deterministic nonlinear distortions.

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...

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...

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.

Audio Query by Example Using Similarity Measures between Probability Density Functions of Features

Directory of Open Access Journals (Sweden)

Marko Helén

2010-01-01

Full Text Available This paper proposes a query by example system for generic audio. We estimate the similarity of the example signal and the samples in the queried database by calculating the distance between the probability density functions (pdfs of their frame-wise acoustic features. Since the features are continuous valued, we propose to model them using Gaussian mixture models (GMMs or hidden Markov models (HMMs. The models parametrize each sample efficiently and retain sufficient information for similarity measurement. To measure the distance between the models, we apply a novel Euclidean distance, approximations of Kullback-Leibler divergence, and a cross-likelihood ratio test. The performance of the measures was tested in simulations where audio samples are automatically retrieved from a general audio database, based on the estimated similarity to a user-provided example. The simulations show that the distance between probability density functions is an accurate measure for similarity. Measures based on GMMs or HMMs are shown to produce better results than that of the existing methods based on simpler statistics or histograms of the features. A good performance with low computational cost is obtained with the proposed Euclidean distance.

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.

Modulation Classification of Satellite Communication Signals Using Cumulants and Neural Networks

Science.gov (United States)

Smith, Aaron; Evans, Michael; Downey, Joseph

2017-01-01

National Aeronautics and Space Administration (NASA)'s future communication architecture is evaluating cognitive technologies and increased system intelligence. These technologies are expected to reduce the operational complexity of the network, increase science data return, and reduce interference to self and others. In order to increase situational awareness, signal classification algorithms could be applied to identify users and distinguish sources of interference. A significant amount of previous work has been done in the area of automatic signal classification for military and commercial applications. As a preliminary step, we seek to develop a system with the ability to discern signals typically encountered in satellite communication. Proposed is an automatic modulation classifier which utilizes higher order statistics (cumulants) and an estimate of the signal-to-noise ratio. These features are extracted from baseband symbols and then processed by a neural network for classification. The modulation types considered are phase-shift keying (PSK), amplitude and phase-shift keying (APSK),and quadrature amplitude modulation (QAM). Physical layer properties specific to the Digital Video Broadcasting - Satellite- Second Generation (DVB-S2) standard, such as pilots and variable ring ratios, are also considered. This paper will provide simulation results of a candidate modulation classifier, and performance will be evaluated over a range of signal-to-noise ratios, frequency offsets, and nonlinear amplifier distortions.

Theoretical perspectives and new practices in audio-graphic conferencing for language learning

OpenAIRE

Hampel, Regine

2003-01-01

This article will start with the situation at the Open University, where languages are taught at a distance. Online tuition using an audio-graphic Internet-based conferencing system called Lyceum is one of the ways used to develop students' communicative skills.\\ud Following Garrett's call for an integration of research and practice at EUROCALL 1997 (Garrett, 1998) – a call which is still valid today – the present article proposes a conceptual framework which can support the use of conferenci...

The Build-Up Course of Visuo-Motor and Audio-Motor Temporal Recalibration

Directory of Open Access Journals (Sweden)

Yoshimori Sugano

2011-10-01

Full Text Available The sensorimotor timing is recalibrated after a brief exposure to a delayed feedback of voluntary actions (temporal recalibration effect: TRE (Heron et al., 2009; Stetson et al., 2006; Sugano et al., 2010. We introduce a new paradigm, namely ‘synchronous tapping’ (ST which allows us to investigate how the TRE builds up during adaptation. In each experimental trial, participants were repeatedly exposed to a constant lag (∼150 ms between their voluntary action (pressing a mouse and a feedback stimulus (a visual flash / an auditory click 10 times. Immediately after that, they performed a ST task with the same stimulus as a pace signal (7 flashes / clicks. A subjective ‘no-delay condition’ (∼50 ms served as control. The TRE manifested itself as a change in the tap-stimulus asynchrony that compensated the exposed lag (eg, after lag adaptation, the tap preceded the stimulus more than in control and built up quickly (∼3–6 trials, ∼23–45 sec in both the visuo- and audio-motor domain. The audio-motor TRE was bigger and built-up faster than the visuo-motor one. To conclude, the TRE is comparable between visuo- and audio-motor domain, though they are slightly different in size and build-up rate.

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 %....

Subjective and Objective Assessment of Perceived Audio Quality of Current Digital Audio Broadcasting Systems and Web-Casting Applications

NARCIS (Netherlands)

Pocta, P.; Beerends, J.G.

2015-01-01

This paper investigates the impact of different audio codecs typically deployed in current digital audio broadcasting (DAB) systems and web-casting applications, which represent a main source of quality impairment in these systems and applications, on the quality perceived by the end user. Both

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.

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

Fractal Communication System Using Digital Signal Processing Starter Kit (DSK TMS320c6713

Directory of Open Access Journals (Sweden)

Arsyad Ramadhan Darlis

2015-12-01

Full Text Available In 1992, Wornell and Oppenheim did research on a modulation which is formed by using wavelet theory. In some other studies, proved that this modulation can survive on a few channels and has reliability in some applications. Because of this modulation using the concept of fractal, then it is called as fractalmodulation. Fractal modulation is formed by inserting information signal into fractal signals that are selffractal similary. This modulation technique has the potential to replace the OFDM (Orthogonal Frequency Division Multiplexing, which is currently used on some of the latest telecommunication technologies. The purpose of this research is to implement the fractal communication system using Digital Signal Processing Starter Kit (DSK TMS320C6713 without using AWGN and Rayleigh channel in order to obtain the ideal performance of the system. From the simulation results using MATLAB7.4. it appears that this communication system has good performance on some channels than any other communication systems. While in terms of implementation by using (DSK via TMS320C6713 Code Composer Studio (CCS, it can be concluded that thefractal communication system has a better execution time on some tests.

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.

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…

Secure communications of CAP-4 and OOK signals over MMF based on electro-optic chaos.

Science.gov (United States)

Ai, Jianzhou; Wang, Lulu; Wang, Jian

2017-09-15

Chaos-based secure communication can provide a high level of privacy in data transmission. Here, we experimentally demonstrate secure signal transmission over two kinds of multimode fiber (MMF) based on electro-optic intensity chaos. High-quality synchronization is achieved in an electro-optic feedback configuration. Both 5  Gbit/s carrier-less amplitude/phase (CAP-4) modulation and 10  Gbit/s on-off key (OOK) signals are recovered efficiently in electro-optic chaos-based communication systems. Degradations of chaos synchronization and communication system due to mismatch of various hardware keys are also discussed.

International Conference on VLSI, Communication, Advanced Devices, Signals & Systems and Networking

CERN Document Server

Shirur, Yasha; Prasad, Rekha

2013-01-01

This book is a collection of papers presented by renowned researchers, keynote speakers and academicians in the International Conference on VLSI, Communication, Analog Designs, Signals and Systems, and Networking (VCASAN-2013), organized by B.N.M. Institute of Technology, Bangalore, India during July 17-19, 2013. The book provides global trends in cutting-edge technologies in electronics and communication engineering. The content of the book is useful to engineers, researchers and academicians as well as industry professionals.

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.

Economic and legal aspects of introducing novel ICT instruments: integrating sound into social media marketing - from audio branding to soundscaping

Directory of Open Access Journals (Sweden)

Daj, A.

2013-12-01

Full Text Available The pervasive expansion and implementation of ICT based marketing instruments imposes a new economic investigation of business models and regulatory solutions. Moreover, the current status of Social Media research indicates that the use of social networking and collaboration technologies is deeply changing the way people communicate, consume and cooperate with each other. Against the backdrop of widespread availability of digital audio-video content and the growing number of “smart” mobile devices, business professionals have developed new strategies for achieving customer involvement and retention through digitally linking audio stimuli to the powerful networking environment of Social Media.

An Analysis of Audio Features to Develop a Human Activity Recognition Model Using Genetic Algorithms, Random Forests, and Neural Networks

Directory of Open Access Journals (Sweden)

Carlos E. Galván-Tejada

2016-01-01

Full Text Available This work presents a human activity recognition (HAR model based on audio features. The use of sound as an information source for HAR models represents a challenge because sound wave analyses generate very large amounts of data. However, feature selection techniques may reduce the amount of data required to represent an audio signal sample. Some of the audio features that were analyzed include Mel-frequency cepstral coefficients (MFCC. Although MFCC are commonly used in voice and instrument recognition, their utility within HAR models is yet to be confirmed, and this work validates their usefulness. Additionally, statistical features were extracted from the audio samples to generate the proposed HAR model. The size of the information is necessary to conform a HAR model impact directly on the accuracy of the model. This problem also was tackled in the present work; our results indicate that we are capable of recognizing a human activity with an accuracy of 85% using the HAR model proposed. This means that minimum computational costs are needed, thus allowing portable devices to identify human activities using audio as an information source.

Digital circuit for the introduction and later removal of dither from an analog signal

Science.gov (United States)

Borgen, Gary S.

1994-05-01

An electronics circuit is presented for accurately digitizing an analog audio or like data signal into a digital equivalent signal by introducing dither into the analog signal and then subsequently removing the dither from the digitized signal prior to its conversion to an analog signal which is a substantial replica of the incoming analog audio or like data signal. The electronics circuit of the present invention is characterized by a first pseudo-random number generator which generates digital random noise signals or dither for addition to the digital equivalent signal and a second pseudo-random number generator which generates subtractive digital random noise signals for the subsequent removal of dither from the digital equivalent signal prior its conversion to the analog replica signal.

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%..

Digital Signal Processing for Optical Coherent Communication Systems

DEFF Research Database (Denmark)

Zhang, Xu

spectrum narrowing tolerance 112-Gb/s DP-QPSK optical coherent systems using digital adaptive equalizer. The demonstrated results show that off-line DSP algorithms are able to reduce the bit error rate (BER) penalty induced by signal spectrum narrowing. Third, we also investigate bi...... wavelength division multiplex (U-DWDM) optical coherent systems based on 10-Gbaud QPSK. We report U-DWDM 1.2-Tb/s QPSK coherent system achieving spectral efficiency of 4.0-bit/s/Hz. In the experimental demonstration, digital decision feed back equalizer (DFE) algorithms and a finite impulse response (FIR......In this thesis, digital signal processing (DSP) algorithms are studied to compensate for physical layer impairments in optical fiber coherent communication systems. The physical layer impairments investigated in this thesis include optical fiber chromatic dispersion, polarization demultiplexing...

Tactile band : accessing gaze signals from the sighted in face-to-face communication

NARCIS (Netherlands)

Qiu, S.; Rauterberg, G.W.M.; Hu, J.

2016-01-01

Gaze signals, frequently used by the sighted in social interactions as visual cues, are hardly accessible for low-vision and blind people. A concept is proposed to help the blind people access and react to gaze signals in face-to-face communication. 20 blind and low-vision participants were

A Simple Semaphore Signaling Technique for Ultra-High Frequency Spacecraft Communications

Science.gov (United States)

Butman, S.; Satorius, E.; Illott, P.

2005-11-01

For planetary lander missions such as the upcoming Phoenix mission to Mars, the most challenging phase of the spacecraft-to-ground communications is during the critical phase termed entry, descent, and landing (EDL). At 8.4 GHz (X-band), the signals received by the largest Deep Space Network (DSN) antennas can be too weak for even 1 bit per second (bps) and therefore not able to communicate critical information to Earth. Fortunately, the lander's ultra-high frequency (UHF) link to an orbiting relay can meet the EDL requirements, but the data rate needs to be low enough to fit the capability of the UHF link during some or all of EDL. On Phoenix, the minimum data rate of the as-built UHF radio is 8 kbps and requires a signal level at the Odyssey orbiter of at least minus 120 dBm. For lower signaling levels, the effective data rate needs to be reduced, but without incurring the cost of rebuilding and requalifying the equipment. To address this scenario, a simple form of frequency-shift keying (FSK) has been devised by appropriately programming the data stream that is input to the UHF transceiver. This article describes this technique and provides performance estimates. Laboratory testing reveals that input signal levels at minus 140 dBm and lower can routinely be demodulated with the proposed signaling scheme, thereby providing a 20-dB and greater margin over the 8-kbps threshold.

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

Use of sparker signal to classify seafloor sediment

Digital Repository Service at National Institute of Oceanography (India)

Pathak, D.; Ranade, G.; Sudhakar, T.

During the cruise 190 of R.V. Gaveshani, the sparker signal was recorded in the analog form on audio cassettes. This signal has been digitized and a statistical computation, viz. the normalized cross-correlation function between successive echoes...

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-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.

Parametric Packet-Layer Model for Evaluation Audio Quality in Multimedia Streaming Services

Science.gov (United States)

Egi, Noritsugu; Hayashi, Takanori; Takahashi, Akira

We propose a parametric packet-layer model for monitoring audio quality in multimedia streaming services such as Internet protocol television (IPTV). This model estimates audio quality of experience (QoE) on the basis of quality degradation due to coding and packet loss of an audio sequence. The input parameters of this model are audio bit rate, sampling rate, frame length, packet-loss frequency, and average burst length. Audio bit rate, packet-loss frequency, and average burst length are calculated from header information in received IP packets. For sampling rate, frame length, and audio codec type, the values or the names used in monitored services are input into this model directly. We performed a subjective listening test to examine the relationships between these input parameters and perceived audio quality. The codec used in this test was the Advanced Audio Codec-Low Complexity (AAC-LC), which is one of the international standards for audio coding. On the basis of the test results, we developed an audio quality evaluation model. The verification results indicate that audio quality estimated by the proposed model has a high correlation with perceived audio quality.

Behavioral and Single-Neuron Sensitivity to Millisecond Variations in Temporally Patterned Communication Signals.

Science.gov (United States)

Baker, Christa A; Ma, Lisa; Casareale, Chelsea R; Carlson, Bruce A

2016-08-24

In many sensory pathways, central neurons serve as temporal filters for timing patterns in communication signals. However, how a population of neurons with diverse temporal filtering properties codes for natural variation in communication signals is unknown. Here we addressed this question in the weakly electric fish Brienomyrus brachyistius, which varies the time intervals between successive electric organ discharges to communicate. These fish produce an individually stereotyped signal called a scallop, which consists of a distinctive temporal pattern of ∼8-12 electric pulses. We manipulated the temporal structure of natural scallops during behavioral playback and in vivo electrophysiology experiments to probe the temporal sensitivity of scallop encoding and recognition. We found that presenting time-reversed, randomized, or jittered scallops increased behavioral response thresholds, demonstrating that fish's electric signaling behavior was sensitive to the precise temporal structure of scallops. Next, using in vivo intracellular recordings and discriminant function analysis, we found that the responses of interval-selective midbrain neurons were also sensitive to the precise temporal structure of scallops. Subthreshold changes in membrane potential recorded from single neurons discriminated natural scallops from time-reversed, randomized, and jittered sequences. Pooling the responses of multiple neurons improved the discriminability of natural sequences from temporally manipulated sequences. Finally, we found that single-neuron responses were sensitive to interindividual variation in scallop sequences, raising the question of whether fish may analyze scallop structure to gain information about the sender. Collectively, these results demonstrate that a population of interval-selective neurons can encode behaviorally relevant temporal patterns with millisecond precision. The timing patterns of action potentials, or spikes, play important roles in representing

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

Audio stream classification for multimedia database search

Science.gov (United States)

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

2013-03-01

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

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)

Automated processing of massive audio/video content using FFmpeg

Directory of Open Access Journals (Sweden)

Kia Siang Hock

2014-01-01

Full Text Available Audio and video content forms an integral, important and expanding part of the digital collections in libraries and archives world-wide. While these memory institutions are familiar and well-versed in the management of more conventional materials such as books, periodicals, ephemera and images, the handling of audio (e.g., oral history recordings and video content (e.g., audio-visual recordings, broadcast content requires additional toolkits. In particular, a robust and comprehensive tool that provides a programmable interface is indispensable when dealing with tens of thousands of hours of audio and video content. FFmpeg is comprehensive and well-established open source software that is capable of the full-range of audio/video processing tasks (such as encode, decode, transcode, mux, demux, stream and filter. It is also capable of handling a wide-range of audio and video formats, a unique challenge in memory institutions. It comes with a command line interface, as well as a set of developer libraries that can be incorporated into applications.

Large-signal modulation characteristics of a GaN-based micro-LED for Gbps visible-light communication

Science.gov (United States)

Tian, Pengfei; Wu, Zhengyuan; Liu, Xiaoyan; Fang, Zhilai; Zhang, Shuailong; Zhou, Xiaolin; Liu, Kefu; Liu, Ming-Gang; Chen, Shu-Jhih; Lee, Chia-Yu; Cong, Chunxiao; Hu, Laigui; Qiu, Zhi-Jun; Zheng, Lirong; Liu, Ran

2018-04-01

The large-signal modulation characteristics of a GaN-based micro-LED have been studied for Gbps visible-light communication. With an increasing signal modulation depth the modulation bandwidth decreases, which matches up with the increase in the sum of the signal rise time and fall time. By simulating the band diagram and the carrier recombination rate of the micro-LED under large-signal modulation, carrier recombination and the carrier sweep-out effect are analyzed and found to be the dominant mechanisms behind the variation of modulation bandwidth. These results give further insight into improving the modulation bandwidth for high-speed visible-light communication.

Vectorial signalling mechanism required for cell-cell communication during sporulation in Bacillus subtilis.

Science.gov (United States)

Diez, Veronica; Schujman, Gustavo E; Gueiros-Filho, Frederico J; de Mendoza, Diego

2012-01-01

Spore formation in Bacillus subtilis takes place in a sporangium consisting of two chambers, the forespore and the mother cell, which are linked by pathways of cell-cell communication. One pathway, which couples the proteolytic activation of the mother cell transcription factor σ(E) to the action of a forespore synthesized signal molecule, SpoIIR, has remained enigmatic. Signalling by SpoIIR requires the protein to be exported to the intermembrane space between forespore and mother cell, where it will interact with and activate the integral membrane protease SpoIIGA. Here we show that SpoIIR signal activity as well as the cleavage of its N-terminal extension is strictly dependent on the prespore fatty acid biosynthetic machinery. We also report that a conserved threonine residue (T27) in SpoIIR is required for processing, suggesting that signalling of SpoIIR is dependent on fatty acid synthesis probably because of acylation of T27. In addition, SpoIIR localization in the forespore septal membrane depends on the presence of SpoIIGA. The orchestration of σ(E) activation in the intercellular space by an acylated signal protein provides a new paradigm to ensure local transmission of a weak signal across the bilayer to control cell-cell communication during development. © 2011 Blackwell Publishing Ltd.

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)

Ambiguity Function Analysis and Processing for Passive Radar Based on CDR Digital Audio Broadcasting

Directory of Open Access Journals (Sweden)

Zhang Qiang

2015-01-01

Full Text Available China Digital Radio (CDR broadcasting is a new standard of digital audio broadcasting of FM frequency (87–108 MHz based on our research and development efforts. It is compatible with the frequency spectrum in analog FM radio and satisfies the requirements for smooth transition from analog to digital signal in FM broadcasting in China. This paper focuses on the signal characteristics and processing methods of radio-based passive radar. The signal characteristics and ambiguity function of a passive radar illumination source are analyzed. The adverse effects on the target detection of the side peaks owing to cyclic prefix, the Doppler ambiguity strips because of signal synchronization, and the range of side peaks resulting from the signal discontinuous spectrum are then studied. Finally, methods for suppressing these side peaks are proposed and their effectiveness is verified by simulations.

Analysis, Synthesis, and Classification of Nonlinear Systems Using Synchronized Swept-Sine Method for Audio Effects

Directory of Open Access Journals (Sweden)

Novak Antonin

2010-01-01

Full Text Available A new method of identification, based on an input synchronized exponential swept-sine signal, is used to analyze and synthesize nonlinear audio systems like overdrive pedals for guitar. Two different pedals are studied; the first one exhibiting a strong influence of the input signal level on its input/output law and the second one exhibiting a weak influence of this input signal level. The Synchronized Swept Sine method leads to a Generalized Polynomial Hammerstein model equivalent to the pedals under test. The behaviors of both pedals are illustrated through model-based resynthesized signals. Moreover, it is also shown that this method leads to a criterion allowing the classification of the nonlinear systems under test, according to the influence of the input signal levels on their input/output law.

Can Synchronous Computer-Mediated Communication (CMC) Help Beginning-Level Foreign Language Learners Speak?

Science.gov (United States)

Ko, Chao-Jung

2012-01-01

This study investigated the possibility that initial-level learners may acquire oral skills through synchronous computer-mediated communication (SCMC). Twelve Taiwanese French as a foreign language (FFL) students, divided into three groups, were required to conduct a variety of tasks in one of the three learning environments (video/audio, audio,…

Video as a technology for interpersonal communications: a new perspective

Science.gov (United States)

Whittaker, Steve

1995-03-01

Some of the most challenging multimedia applications have involved real- time conferencing, using audio and video to support interpersonal communication. Here we re-examine assumptions about the role, importance and implementation of video information in such systems. Rather than focussing on novel technologies, we present evaluation data relevant to both the classes of real-time multimedia applications we should develop and their design and implementation. Evaluations of videoconferencing systems show that previous work has overestimated the importance of video at the expense of audio. This has strong implications for the implementation of bandwidth allocation and synchronization. Furthermore our recent studies of workplace interaction show that prior work has neglected another potentially vital function of visual information: in assessing the communication availability of others. In this new class of application, rather than providing a supplement to audio information, visual information is used to promote the opportunistic communications that are prevalent in face-to-face settings. We discuss early experiments with such connection applications and identify outstanding design and implementation issues. Finally we examine a different class of application 'video-as-data', where the video image is used to transmit information about the work objects themselves, rather than information about interactants.

OPTICAL WIRELESS COMMUNICATION SYSTEM

Directory of Open Access Journals (Sweden)

JOSHUA L.Y. CHIENG

2016-02-01

Full Text Available The growing demand of bandwidth in this modern internet age has been testing the existing telecommunication infrastructures around the world. With broadband speeds moving towards the region of Gbps and Tbps, many researches have begun on the development of using optical wireless technology as feasible and future methods to the current wireless technology. Unlike the existing radio frequency wireless applications, optical wireless uses electromagnetic spectrums that are unlicensed and free. With that, this project aim to understand and gain better understanding of optical wireless communication system by building an experimental and simulated model. The quality of service and system performance will be investigated and reviewed. This project employs laser diode as the propagation medium and successfully transferred audio signals as far as 15 meters. On its quality of service, results of the project model reveal that the bit error rate increases, signal-to-noise ratio and quality factor decreases as the link distance between the transmitter and receiver increases. OptiSystem was used to build the simulated model and MATLAB was used to assist signal-to-noise ratio calculations. By comparing the simulated and experimental receiver’s power output, the experimental model’s efficiency is at 66.3%. Other than the system’s performance, challenges and factors affecting the system have been investigated and discussed. Such challenges include beam divergence, misalignment and particle absorption.

The Personal Hearing System—A Software Hearing Aid for a Personal Communication System

Directory of Open Access Journals (Sweden)

Giso Grimm

2009-01-01

Full Text Available A concept and architecture of a personal communication system (PCS is introduced that integrates audio communication and hearing support for the elderly and hearing-impaired through a personal hearing system (PHS. The concept envisions a central processor connected to audio headsets via a wireless body area network (WBAN. To demonstrate the concept, a prototype PCS is presented that is implemented on a netbook computer with a dedicated audio interface in combination with a mobile phone. The prototype can be used for field-testing possible applications and to reveal possibilities and limitations of the concept of integrating hearing support in consumer audio communication devices. It is shown that the prototype PCS can integrate hearing aid functionality, telephony, public announcement systems, and home entertainment. An exemplary binaural speech enhancement scheme that represents a large class of possible PHS processing schemes is shown to be compatible with the general concept. However, an analysis of hardware and software architectures shows that the implementation of a PCS on future advanced cell phone-like devices is challenging. Because of limitations in processing power, recoding of prototype implementations into fixed point arithmetic will be required and WBAN performance is still a limiting factor in terms of data rate and delay.

The Personal Hearing System—A Software Hearing Aid for a Personal Communication System

Science.gov (United States)

Grimm, Giso; Guilmin, Gwénaël; Poppen, Frank; Vlaming, Marcel S. M. G.; Hohmann, Volker

2009-12-01

A concept and architecture of a personal communication system (PCS) is introduced that integrates audio communication and hearing support for the elderly and hearing-impaired through a personal hearing system (PHS). The concept envisions a central processor connected to audio headsets via a wireless body area network (WBAN). To demonstrate the concept, a prototype PCS is presented that is implemented on a netbook computer with a dedicated audio interface in combination with a mobile phone. The prototype can be used for field-testing possible applications and to reveal possibilities and limitations of the concept of integrating hearing support in consumer audio communication devices. It is shown that the prototype PCS can integrate hearing aid functionality, telephony, public announcement systems, and home entertainment. An exemplary binaural speech enhancement scheme that represents a large class of possible PHS processing schemes is shown to be compatible with the general concept. However, an analysis of hardware and software architectures shows that the implementation of a PCS on future advanced cell phone-like devices is challenging. Because of limitations in processing power, recoding of prototype implementations into fixed point arithmetic will be required and WBAN performance is still a limiting factor in terms of data rate and delay.

Communication efficiency and congestion of signal traffic in large-scale brain networks.

Science.gov (United States)

Mišić, Bratislav; Sporns, Olaf; McIntosh, Anthony R

2014-01-01

The complex connectivity of the cerebral cortex suggests that inter-regional communication is a primary function. Using computational modeling, we show that anatomical connectivity may be a major determinant for global information flow in brain networks. A macaque brain network was implemented as a communication network in which signal units flowed between grey matter nodes along white matter paths. Compared to degree-matched surrogate networks, information flow on the macaque brain network was characterized by higher loss rates, faster transit times and lower throughput, suggesting that neural connectivity may be optimized for speed rather than fidelity. Much of global communication was mediated by a "rich club" of hub regions: a sub-graph comprised of high-degree nodes that are more densely interconnected with each other than predicted by chance. First, macaque communication patterns most closely resembled those observed for a synthetic rich club network, but were less similar to those seen in a synthetic small world network, suggesting that the former is a more fundamental feature of brain network topology. Second, rich club regions attracted the most signal traffic and likewise, connections between rich club regions carried more traffic than connections between non-rich club regions. Third, a number of rich club regions were significantly under-congested, suggesting that macaque connectivity actively shapes information flow, funneling traffic towards some nodes and away from others. Together, our results indicate a critical role of the rich club of hub nodes in dynamic aspects of global brain communication.

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. 

Determination of over current protection thresholds for class D audio amplifiers

DEFF Research Database (Denmark)

Nyboe, Flemming; Risbo, L; Andreani, Pietro

2005-01-01

Monolithic class-D audio amplifiers typically feature built-in over current protection circuitry that shuts down the amplifier in case of a short circuit on the output speaker terminals. To minimize cost, the threshold at which the device shuts down must be set just above the maximum current...... that can flow in the loudspeaker during normal operation. The current required is determined by the complex loudspeaker impedance and properties of the music signals played. This work presents a statistical analysis of peak output currents when playing music on typical loudspeakers for home entertainment....

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...

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....

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...

Dephasing in coherent communication with weak signal states

International Nuclear Information System (INIS)

Jarzyna, Marcin; Banaszek, Konrad; Demkowicz-Dobrzański, Rafał

2014-01-01

We analyse the ultimate quantum limit on the accessible information for an optical communication scheme when time bins carry coherent light pulses prepared in one of several orthogonal modes and the phase undergoes diffusion after each channel use. This scheme, an example of a quantum memory channel, can be viewed as noisy pulse position modulation (PPM) keying with phase fluctuations occurring between consecutive PPM symbols. We derive a general expression for the output states in the Fock basis and implement a numerical procedure to calculate the Holevo quantity. Using asymptotic properties of Toeplitz matrices, we also present an analytic expression for the Holevo quantity valid for very weak signals and sufficiently strong dephasing when the dominant contribution comes from the single-photon sector in the Hilbert space of signal states. Based on numerical results we conjecture an inequality for contributions to the Holevo quantity from multiphoton sectors which implies that in the asymptotic limit of weak signals, for arbitrarily small dephasing the accessible information scales linearly with the average number of photons contained in the pulse. Such behaviour presents a qualitative departure from the fully coherent case. (paper)

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.

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.

A digital input class-D audio amplifier with sixth-order PWM

International Nuclear Information System (INIS)

Luo Shumeng; Li Dongmei

2013-01-01

A digital input class-D audio amplifier with a sixth-order pulse-width modulation (PWM) modulator is presented. This modulator moves the PWM generator into the closed sigma—delta modulator loop. The noise and distortions generated at the PWM generator module are suppressed by the high gain of the forward loop of the sigma—delta modulator. Therefore, at the output of the modulator, a very clean PWM signal is acquired for driving the power stage of the class-D amplifier. A sixth-order modulator is designed to balance the performance and the system clock speed. Fabricated in standard 0.18 μm CMOS technology, this class-D amplifier achieves 110 dB dynamic range, 100 dB signal-to-noise rate, and 0.0056% total harmonic distortion plus noise. (semiconductor integrated circuits)

Penguat Audio Kelas D dengan Umpan Balik Tipe Butterworth

Directory of Open Access Journals (Sweden)

Gunawan Dewantoro

2016-03-01

Full Text Available A class D amplifier would, in ideal sense, amplify signals without any noises and distortions which yield 100% efficiency and 0% Total Harmonic Distortion (THD. However, class D amplifiers have some drawbacks that lead to nonlinearity and increasing THD. Therefore, a feedback mechanism was employed to enhance THD performance of amplifier. Some feedback techniques have been using first order filter in the feedback path to retrieve audio signals. This research proposed a second order filter with Butterworth approach. A power amplifier was realized using full-bridge amplifier with MOSFETs to provide greater power. This class D amplifier was designed to meet following specifications: maximum output power up to 32.6 W with an 8 Ω load, sensitivity of 90 mV/W, frequency response ranging from 20 Hz – 20 kHz with tolerance ± 1 dB, THD as low as 1.1 %, SNR up to 90.16 dB, and efficiency of 82.1 %.

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…

Deep learning, audio adversaries, and music content analysis

DEFF Research Database (Denmark)

Kereliuk, Corey Mose; Sturm, Bob L.; Larsen, Jan

2015-01-01

We present the concept of adversarial audio in the context of deep neural networks (DNNs) for music content analysis. An adversary is an algorithm that makes minor perturbations to an input that cause major repercussions to the system response. In particular, we design an adversary for a DNN...... that takes as input short-time spectral magnitudes of recorded music and outputs a high-level music descriptor. We demonstrate how this adversary can make the DNN behave in any way with only extremely minor changes to the music recording signal. We show that the adversary cannot be neutralised by a simple...... filtering of the input. Finally, we discuss adversaries in the broader context of the evaluation of music content analysis systems....

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.

Optimized Audio Classification and Segmentation Algorithm by Using Ensemble Methods

Directory of Open Access Journals (Sweden)

Saadia Zahid

2015-01-01

Full Text Available Audio segmentation is a basis for multimedia content analysis which is the most important and widely used application nowadays. An optimized audio classification and segmentation algorithm is presented in this paper that segments a superimposed audio stream on the basis of its content into four main audio types: pure-speech, music, environment sound, and silence. An algorithm is proposed that preserves important audio content and reduces the misclassification rate without using large amount of training data, which handles noise and is suitable for use for real-time applications. Noise in an audio stream is segmented out as environment sound. A hybrid classification approach is used, bagged support vector machines (SVMs with artificial neural networks (ANNs. Audio stream is classified, firstly, into speech and nonspeech segment by using bagged support vector machines; nonspeech segment is further classified into music and environment sound by using artificial neural networks and lastly, speech segment is classified into silence and pure-speech segments on the basis of rule-based classifier. Minimum data is used for training classifier; ensemble methods are used for minimizing misclassification rate and approximately 98% accurate segments are obtained. A fast and efficient algorithm is designed that can be used with real-time multimedia applications.

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.

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 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.

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.

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...

Two-way digital communications

Science.gov (United States)

Glenn, William E.; Daly, Ed

1996-03-01

The communications industry has been rapidly converting from analog to digital communications for audio, video, and data. The initial applications have been concentrating on point-to-multipoint transmission. Currently, a new revolution is occurring in which two-way point-to-point transmission is a rapidly growing market. The system designs for video compression developed for point-to-multipoint transmission are unsuitable for this new market as well as for satellite based video encoding. A new system developed by the Space Communications Technology Center has been designed to address both of these newer applications. An update on the system performance and design will be given.

32 CFR 705.4 - Communication directly with private organizations and individuals.

Science.gov (United States)

2010-07-01

... 32 National Defense 5 2010-07-01 2010-07-01 false Communication directly with private... Communication directly with private organizations and individuals. (a) Questions from the public and requests... current date may be purchased from the National Archives. Details are available from: Audio-Visual Branch...

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

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.

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.

AERIS : Eco-Vehicle Speed Control at Signalized Intersections Using I2V Communication

Science.gov (United States)

2012-06-01

This report concentrates on a velocity advisory tool, or decision support system, for vehicles approaching an intersection using communication capabilities between the infrastructure and vehicles. The system uses available signal change information, ...

PERANCANGAN MEDIA PEMBELAJARAN BERBASIS AUDIO VISUAL UNTUK MATA KULIAH TIPOGRAFI PADA PROGRAM STUDI DESAIN KOMUNIKASI VISUAL UNIVERSITAS DIAN NUSWANTORO

Directory of Open Access Journals (Sweden)

Puri Sulistiyawati

2017-02-01

Full Text Available Abstrak Tipografi merupakan salah satu mata kuliah pada bidang desain komunikasi visual yang mengutamakan aspek visual. Namun berdasarkan hasil observasi diketahui bahwa media pembelajaran yang selama ini digunakan kurang efektif karena kurangnya pemanfaatan teknologi informasi, sehingga mahasiswa kurang maksimal dalam memahami materi kuliah yang disampaikan oleh pengajar. Perkembangan teknologi informasi saat ini banyak memberikan dampak positif bagi kemajuan bidang pendidikan diantaranya dapat digunakan untuk mendukung media dalam proses pembelajaran. Tujuan penelitian ini adalah merancang media pembelajaran untuk mata kuliah tipografi dengan memanfaatkan teknologi informasi yaitu media audio visual. Metode yang digunakan dalam penelitian ini adalah Research and Development dengan pendekatan model ADDIE (Analysis, Design, Development, Implementation, Evaluation. Dengan diciptakannya media pembelajaran audio visual ini diharapkan proses pembelajaran mata kuliah Tipografi dapat lebih efektif dan materi kuliah lebih mudah dipahami oleh mahasiswa. Kata Kunci : audio visual, media pembelajaran, tipografi Abstract Typography is one of the subjects in the field of visual communication design that prioritizes the visual aspect. However, based on the observation note that the media has been used less effective because the lack of use information technology, so students can't understand the course material that explained by lecturers. Today, the development of information technology is being positive impact for the advancement of education which can be used to support the media in the learning process. The purpose of this research is to design learning media for the course of typography by utilizing information technology, called audio-visual media.  The method that used in this research is Research and Development with ADDIE model (Analysis, Design, Development, Implementation, Evaluation. With the creation of audio-visual learning media is expected

The newest digital signal processing

International Nuclear Information System (INIS)

Lee, Chae Uk

2002-08-01

This book deal with the newest digital signal processing, which contains introduction on conception of digital signal processing, constitution and purpose, signal and system such as signal, continuos signal, discrete signal and discrete system, I/O expression on impress response, convolution, mutual connection of system and frequency character,z transform of definition, range, application of z transform and relationship with laplace transform, Discrete fourier, Fast fourier transform on IDFT algorithm and FFT application, foundation of digital filter of notion, expression, types, frequency characteristic of digital filter and design order of filter, Design order of filter, Design of FIR digital filter, Design of IIR digital filter, Adaptive signal processing, Audio signal processing, video signal processing and application of digital signal processing.

Communicating Art through Interactive Technology: New Approaches for Interaction Design in Art Museums

DEFF Research Database (Denmark)

Kortbek, Karen Johanne; Grønbæk, Kaj

2008-01-01

This paper discusses new approaches to interaction design for communication of art in the physical museum space. In contrast to the widespread utilization of interactive tech­nologies in cultural heritage and natural science museums it is generally a challenge to introduce technology in art museums...... without disturbing the domain of the art works. To explore the possibilities of communicating art through the use of technology, and to minimize disturbance of the artworks, we apply four main approaches in the communication: 1) gentle audio augmentation of art works; 2) conceptual affinity of art works...... and remote interactive installations; 3) using the body as an interaction device; 4) consistent audio-visual cues for interaction opportunities. The paper describes the application of these approaches for communication of inspira­tional material for a Mariko Mori exhibition. The installations are described...

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

Audio-Tactile Integration in Congenitally and Late Deaf Cochlear Implant Users

Science.gov (United States)

Nava, Elena; Bottari, Davide; Villwock, Agnes; Fengler, Ineke; Büchner, Andreas; Lenarz, Thomas; Röder, Brigitte

2014-01-01

Several studies conducted in mammals and humans have shown that multisensory processing may be impaired following congenital sensory loss and in particular if no experience is achieved within specific early developmental time windows known as sensitive periods. In this study we investigated whether basic multisensory abilities are impaired in hearing-restored individuals with deafness acquired at different stages of development. To this aim, we tested congenitally and late deaf cochlear implant (CI) recipients, age-matched with two groups of hearing controls, on an audio-tactile redundancy paradigm, in which reaction times to unimodal and crossmodal redundant signals were measured. Our results showed that both congenitally and late deaf CI recipients were able to integrate audio-tactile stimuli, suggesting that congenital and acquired deafness does not prevent the development and recovery of basic multisensory processing. However, we found that congenitally deaf CI recipients had a lower multisensory gain compared to their matched controls, which may be explained by their faster responses to tactile stimuli. We discuss this finding in the context of reorganisation of the sensory systems following sensory loss and the possibility that these changes cannot be “rewired” through auditory reafferentation. PMID:24918766

Audio-tactile integration in congenitally and late deaf cochlear implant users.

Directory of Open Access Journals (Sweden)

Elena Nava

Full Text Available Several studies conducted in mammals and humans have shown that multisensory processing may be impaired following congenital sensory loss and in particular if no experience is achieved within specific early developmental time windows known as sensitive periods. In this study we investigated whether basic multisensory abilities are impaired in hearing-restored individuals with deafness acquired at different stages of development. To this aim, we tested congenitally and late deaf cochlear implant (CI recipients, age-matched with two groups of hearing controls, on an audio-tactile redundancy paradigm, in which reaction times to unimodal and crossmodal redundant signals were measured. Our results showed that both congenitally and late deaf CI recipients were able to integrate audio-tactile stimuli, suggesting that congenital and acquired deafness does not prevent the development and recovery of basic multisensory processing. However, we found that congenitally deaf CI recipients had a lower multisensory gain compared to their matched controls, which may be explained by their faster responses to tactile stimuli. We discuss this finding in the context of reorganisation of the sensory systems following sensory loss and the possibility that these changes cannot be "rewired" through auditory reafferentation.

Illustration of decimation in digital signal processing (DSP) systems ...

African Journals Online (AJOL)

... and engineering, especially in the areas of communication and medicine. ... This multirate DSP had been found useful in application like digital audio, video and even GSM technology. The work is implemented using MATLABTM software.

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.

Studying physician-adolescent patient communication in community-based practices: recruitment challenges and solutions.

Science.gov (United States)

Bodner, Michael E; Bilheimer, Alicia; Gao, Xiaomei; Lyna, Pauline; Alexander, Stewart C; Dolor, Rowena J; Østbye, Truls; Bravender, Terrill; Tulsky, James A; Graves, Sidney; Irons, Alexis; Pollak, Kathryn I

2015-11-13

Practice-based studies are needed to assess how physicians communicate health messages about weight to overweight/obese adolescent patients, but successful recruitment to such studies is challenging. This paper describes challenges, solutions, and lessons learned to recruit physicians and adolescents to the Teen Communicating Health Analyzing Talk (CHAT) study, a randomized controlled trial of a communication skills intervention for primary care physicians to enhance communication about weight with overweight/obese adolescents. A "peer-to-peer" approach was used to recruit physicians, including the use of "clinic champions" who liaised between study leaders and physicians. Consistent rapport and cooperative working relationships with physicians and clinic staff were developed and maintained. Adolescent clinic files were reviewed (HIPAA waiver) to assess eligibility. Parents could elect to opt-out for their children. To encourage enrollment, confidentiality of audio recordings was emphasized, and financial incentives were offered to all participants. We recruited 49 physicians and audio-recorded 391 of their overweight/obese adolescents' visits. Recruitment challenges included 1) physician reticence to participate; 2) variability in clinic operating procedures; 3) variability in adolescent accrual rates; 4) clinic open access scheduling; and 5) establishing communication with parents and adolescents. Key solutions included the use of a "clinic champion" to help recruit physicians, pro-active, consistent communication with clinic staff, and adapting calling times to reach parents and adolescents. Recruiting physicians and adolescents to audio-recorded, practice-based health communication studies can be successful. Anticipated challenges to recruiting can be met with advanced planning; however, optimal solutions to challenges evolve as recruitment progresses.

Computational Analysis and Simulation of Empathic Behaviors: a Survey of Empathy Modeling with Behavioral Signal Processing Framework.

Science.gov (United States)

Xiao, Bo; Imel, Zac E; Georgiou, Panayiotis; Atkins, David C; Narayanan, Shrikanth S

2016-05-01

Empathy is an important psychological process that facilitates human communication and interaction. Enhancement of empathy has profound significance in a range of applications. In this paper, we review emerging directions of research on computational analysis of empathy expression and perception as well as empathic interactions, including their simulation. We summarize the work on empathic expression analysis by the targeted signal modalities (e.g., text, audio, and facial expressions). We categorize empathy simulation studies into theory-based emotion space modeling or application-driven user and context modeling. We summarize challenges in computational study of empathy including conceptual framing and understanding of empathy, data availability, appropriate use and validation of machine learning techniques, and behavior signal processing. Finally, we propose a unified view of empathy computation and offer a series of open problems for future research.

Computationally Efficient Amplitude Modulated Sinusoidal Audio Coding using Frequency-Domain Linear Prediction

DEFF Research Database (Denmark)

Christensen, M. G.; Jensen, Søren Holdt

2006-01-01

A method for amplitude modulated sinusoidal audio coding is presented that has low complexity and low delay. This is based on a subband processing system, where, in each subband, the signal is modeled as an amplitude modulated sum of sinusoids. The envelopes are estimated using frequency......-domain linear prediction and the prediction coefficients are quantized. As a proof of concept, we evaluate different configurations in a subjective listening test, and this shows that the proposed method offers significant improvements in sinusoidal coding. Furthermore, the properties of the frequency...

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

Dynamically-Loaded Hardware Libraries (HLL) Technology for Audio Applications

DEFF Research Database (Denmark)

Esposito, A.; Lomuscio, A.; Nunzio, L. Di

2016-01-01

In this work, we apply hardware acceleration to embedded systems running audio applications. We present a new framework, Dynamically-Loaded Hardware Libraries or HLL, to dynamically load hardware libraries on reconfigurable platforms (FPGAs). Provided a library of application-specific processors......, we load on-the-fly the specific processor in the FPGA, and we transfer the execution from the CPU to the FPGA-based accelerator. The proposed architecture provides excellent flexibility with respect to the different audio applications implemented, high quality audio, and an energy efficient solution....

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.

SC-CNNs for chaotic signal applications in secure communication systems.

Science.gov (United States)

Caponetto, Riccardo; Fortuna, Luigi; Occhipinti, Luigi; Xibilia, Maria Gabriella

2003-12-01

In this paper a CNNs based circuit for the generation of hyperchaotic signals is proposed. The circuit has been developed for applications in secure communication systems. An Saito oscillator has been designed by using a suitable configuration of a four-cells State-Controlled CNNs. A cryptography system based on the Saito oscillator has been implemented by using inverse system synchronization. The proposed circuit implementation and experimental results are given.

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.

Power, Avionics and Software Communication Network Architecture

Science.gov (United States)

Ivancic, William D.; Sands, Obed S.; Bakula, Casey J.; Oldham, Daniel R.; Wright, Ted; Bradish, Martin A.; Klebau, Joseph M.

2014-01-01

This document describes the communication architecture for the Power, Avionics and Software (PAS) 2.0 subsystem for the Advanced Extravehicular Mobile Unit (AEMU). The following systems are described in detail: Caution Warn- ing and Control System, Informatics, Storage, Video, Audio, Communication, and Monitoring Test and Validation. This document also provides some background as well as the purpose and goals of the PAS project at Glenn Research Center (GRC).

Effect of audio in-vehicle red light-running warning message on driving behavior based on a driving simulator experiment.

Science.gov (United States)

Yan, Xuedong; Liu, Yang; Xu, Yongcun

2015-01-01

Drivers' incorrect decisions of crossing signalized intersections at the onset of the yellow change may lead to red light running (RLR), and RLR crashes result in substantial numbers of severe injuries and property damage. In recent years, some Intelligent Transport System (ITS) concepts have focused on reducing RLR by alerting drivers that they are about to violate the signal. The objective of this study is to conduct an experimental investigation on the effectiveness of the red light violation warning system using a voice message. In this study, the prototype concept of the RLR audio warning system was modeled and tested in a high-fidelity driving simulator. According to the concept, when a vehicle is approaching an intersection at the onset of yellow and the time to the intersection is longer than the yellow interval, the in-vehicle warning system can activate the following audio message "The red light is impending. Please decelerate!" The intent of the warning design is to encourage drivers who cannot clear an intersection during the yellow change interval to stop at the intersection. The experimental results showed that the warning message could decrease red light running violations by 84.3 percent. Based on the logistic regression analyses, drivers without a warning were about 86 times more likely to make go decisions at the onset of yellow and about 15 times more likely to run red lights than those with a warning. Additionally, it was found that the audio warning message could significantly reduce RLR severity because the RLR drivers' red-entry times without a warning were longer than those with a warning. This driving simulator study showed a promising effect of the audio in-vehicle warning message on reducing RLR violations and crashes. It is worthwhile to further develop the proposed technology in field applications.

Mood expression by seniors in digital communication : Evaluative comparison of four mood-reporting instruments with elderly users

NARCIS (Netherlands)

Alberts, J.W.; Vastenburg, M.H.; Desmet, P.M.A.

2013-01-01

Elderly users have widely adopted digital communication. Digital communication is often text-only, e.g. instant messaging (IM) and e-mail. Text-only communication has been found less effective than communication that uses richer channels such as audio and video. Mood expression instruments, such as

An Efficient Method for Image and Audio Steganography using Least Significant Bit (LSB) Substitution

Science.gov (United States)

Chadha, Ankit; Satam, Neha; Sood, Rakshak; Bade, Dattatray

2013-09-01

In order to improve the data hiding in all types of multimedia data formats such as image and audio and to make hidden message imperceptible, a novel method for steganography is introduced in this paper. It is based on Least Significant Bit (LSB) manipulation and inclusion of redundant noise as secret key in the message. This method is applied to data hiding in images. For data hiding in audio, Discrete Cosine Transform (DCT) and Discrete Wavelet Transform (DWT) both are used. All the results displayed prove to be time-efficient and effective. Also the algorithm is tested for various numbers of bits. For those values of bits, Mean Square Error (MSE) and Peak-Signal-to-Noise-Ratio (PSNR) are calculated and plotted. Experimental results show that the stego-image is visually indistinguishable from the original cover-image when nsteganography process does not reveal presence of any hidden message, thus qualifying the criteria of imperceptible message.

Audio Source Separation in Reverberant Environments Using β-Divergence-Based Nonnegative Factorization

DEFF Research Database (Denmark)

Fakhry, Mahmoud; Svaizer, Piergiorgio; Omologo, Maurizio

2017-01-01

-maximization algorithm and used to separate the signals by means of multichannel Wiener filtering. We propose to estimate these parameters by applying nonnegative factorization based on prior information on source variances. In the nonnegative factorization, spectral basis matrices can be defined as the prior...... information. The matrices can be either extracted or indirectly made available through a redundant library that is trained in advance. In a separate step, applying nonnegative tensor factorization, two algorithms are proposed in order to either extract or detect the basis matrices that best represent......In Gaussian model-based multichannel audio source separation, the likelihood of observed mixtures of source signals is parametrized by source spectral variances and by associated spatial covariance matrices. These parameters are estimated by maximizing the likelihood through an expectation...

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...

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.

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

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.

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

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...

Signaling of noncomprehension in communication breakdowns in fragile X syndrome, Down syndrome, and autism spectrum disorder.

Science.gov (United States)

Martin, Gary E; Barstein, Jamie; Hornickel, Jane; Matherly, Sara; Durante, Genna; Losh, Molly

The ability to indicate a failure to understand a message is a critical pragmatic (social) language skill for managing communication breakdowns and supporting successful communicative exchanges. The current study examined the ability to signal noncomprehension across different types of confusing message conditions in children and adolescents with fragile X syndrome (FXS), Down syndrome (DS), autism spectrum disorder (ASD), and typical development (TD). Controlling for nonverbal mental age and receptive vocabulary skills, youth with comorbid FXS and ASD and those with DS were less likely than TD controls to signal noncomprehension of confusing messages. Youth with FXS without ASD and those with idiopathic ASD did not differ from controls. No sex differences were detected in any group. Findings contribute to current knowledge of pragmatic profiles in different forms of genetically-based neurodevelopmental disorders associated with intellectual disability, and the role of sex in the expression of such profiles. Upon completion of this article, readers will have learned about: (1) the social-communicative profiles of youth with FXS, DS, and ASD, (2) the importance of signaling noncomprehension in response to a confusing message, and (3) the similarities and differences in noncomprehension signaling in youth with FXS (with and without ASD), DS, idiopathic ASD, and TD. Copyright © 2017 Elsevier Inc. All rights reserved.

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...

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.

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...

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. 

OFDM systems for wireless communications

CERN Document Server

Narasimhamurthy, Adarsh

2010-01-01

Orthogonal Frequency Division Multiplexing (OFDM) systems are widely used in the standards for digital audio/video broadcasting, WiFi and WiMax. Being a frequency-domain approach to communications, OFDM has important advantages in dealing with the frequency-selective nature of high data rate wireless communication channels. As the needs for operating with higher data rates become more pressing, OFDM systems have emerged as an effective physical-layer solution.This short monograph is intended as a tutorial which highlights the deleterious aspects of the wireless channel and presents why OFDM is

Intracellular Redox Compartmentation and ROS-Related Communication in Regulation and Signaling.

Science.gov (United States)

Noctor, Graham; Foyer, Christine H

2016-07-01

Recent years have witnessed enormous progress in understanding redox signaling related to reactive oxygen species (ROS) in plants. The consensus view is that such signaling is intrinsic to many developmental processes and responses to the environment. ROS-related redox signaling is tightly wedded to compartmentation. Because membranes function as barriers, highly redox-active powerhouses such as chloroplasts, peroxisomes, and mitochondria may elicit specific signaling responses. However, transporter functions allow membranes also to act as bridges between compartments, and so regulated capacity to transmit redox changes across membranes influences the outcome of triggers produced at different locations. As well as ROS and other oxidizing species, antioxidants are key players that determine the extent of ROS accumulation at different sites and that may themselves act as signal transmitters. Like ROS, antioxidants can be transported across membranes. In addition, the intracellular distribution of antioxidative enzymes may be modulated to regulate or facilitate redox signaling appropriate to the conditions. Finally, there is substantial plasticity in organellar shape, with extensions such as stromules, peroxules, and matrixules playing potentially crucial roles in organelle-organelle communication. We provide an overview of the advances in subcellular compartmentation, identifying the gaps in our knowledge and discussing future developments in the area. © 2016 American Society of Plant Biologists. All Rights Reserved.

Signal Processing for Wireless Communication MIMO System with Nano- Scaled CSDG MOSFET based DP4T RF Switch.

Science.gov (United States)

Srivastava, Viranjay M

2015-01-01

In the present technological expansion, the radio frequency integrated circuits in the wireless communication technologies became useful because of the replacement of increasing number of functions, traditional hardware components by modern digital signal processing. The carrier frequencies used for communication systems, now a day, shifted toward the microwave regime. The signal processing for the multiple inputs multiple output wireless communication system using the Metal- Oxide-Semiconductor Field-Effect-Transistor (MOSFET) has been done a lot. In this research the signal processing with help of nano-scaled Cylindrical Surrounding Double Gate (CSDG) MOSFET by means of Double- Pole Four-Throw Radio-Frequency (DP4T RF) switch, in terms of Insertion loss, Isolation, Reverse isolation and Inter modulation have been analyzed. In addition to this a channel model has been presented. Here, we also discussed some patents relevant to the topic.

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...

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…

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...

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.

Auditory cross-modal reorganization in cochlear implant users indicates audio-visual integration.

Science.gov (United States)

Stropahl, Maren; Debener, Stefan

2017-01-01

There is clear evidence for cross-modal cortical reorganization in the auditory system of post-lingually deafened cochlear implant (CI) users. A recent report suggests that moderate sensori-neural hearing loss is already sufficient to initiate corresponding cortical changes. To what extend these changes are deprivation-induced or related to sensory recovery is still debated. Moreover, the influence of cross-modal reorganization on CI benefit is also still unclear. While reorganization during deafness may impede speech recovery, reorganization also has beneficial influences on face recognition and lip-reading. As CI users were observed to show differences in multisensory integration, the question arises if cross-modal reorganization is related to audio-visual integration skills. The current electroencephalography study investigated cortical reorganization in experienced post-lingually deafened CI users ( n  = 18), untreated mild to moderately hearing impaired individuals (n = 18) and normal hearing controls ( n  = 17). Cross-modal activation of the auditory cortex by means of EEG source localization in response to human faces and audio-visual integration, quantified with the McGurk illusion, were measured. CI users revealed stronger cross-modal activations compared to age-matched normal hearing individuals. Furthermore, CI users showed a relationship between cross-modal activation and audio-visual integration strength. This may further support a beneficial relationship between cross-modal activation and daily-life communication skills that may not be fully captured by laboratory-based speech perception tests. Interestingly, hearing impaired individuals showed behavioral and neurophysiological results that were numerically between the other two groups, and they showed a moderate relationship between cross-modal activation and the degree of hearing loss. This further supports the notion that auditory deprivation evokes a reorganization of the auditory system

Auditory cross-modal reorganization in cochlear implant users indicates audio-visual integration

Directory of Open Access Journals (Sweden)

Maren Stropahl

2017-01-01

Full Text Available There is clear evidence for cross-modal cortical reorganization in the auditory system of post-lingually deafened cochlear implant (CI users. A recent report suggests that moderate sensori-neural hearing loss is already sufficient to initiate corresponding cortical changes. To what extend these changes are deprivation-induced or related to sensory recovery is still debated. Moreover, the influence of cross-modal reorganization on CI benefit is also still unclear. While reorganization during deafness may impede speech recovery, reorganization also has beneficial influences on face recognition and lip-reading. As CI users were observed to show differences in multisensory integration, the question arises if cross-modal reorganization is related to audio-visual integration skills. The current electroencephalography study investigated cortical reorganization in experienced post-lingually deafened CI users (n = 18, untreated mild to moderately hearing impaired individuals (n = 18 and normal hearing controls (n = 17. Cross-modal activation of the auditory cortex by means of EEG source localization in response to human faces and audio-visual integration, quantified with the McGurk illusion, were measured. CI users revealed stronger cross-modal activations compared to age-matched normal hearing individuals. Furthermore, CI users showed a relationship between cross-modal activation and audio-visual integration strength. This may further support a beneficial relationship between cross-modal activation and daily-life communication skills that may not be fully captured by laboratory-based speech perception tests. Interestingly, hearing impaired individuals showed behavioral and neurophysiological results that were numerically between the other two groups, and they showed a moderate relationship between cross-modal activation and the degree of hearing loss. This further supports the notion that auditory deprivation evokes a reorganization of the

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.

Real Time Recognition Of Speakers From Internet Audio Stream

Directory of Open Access Journals (Sweden)

Weychan Radoslaw

2015-09-01

Full Text Available In this paper we present an automatic speaker recognition technique with the use of the Internet radio lossy (encoded speech signal streams. We show an influence of the audio encoder (e.g., bitrate on the speaker model quality. The model of each speaker was calculated with the use of the Gaussian mixture model (GMM approach. Both the speaker recognition and the further analysis were realized with the use of short utterances to facilitate real time processing. The neighborhoods of the speaker models were analyzed with the use of the ISOMAP algorithm. The experiments were based on four 1-hour public debates with 7–8 speakers (including the moderator, acquired from the Polish radio Internet services. The presented software was developed with the MATLAB environment.

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.

The nature of alarm communication in Constrictotermes cyphergaster (Blattodea: Termitoidea: Termitidae: the integration of chemical and vibroacoustic signals

Directory of Open Access Journals (Sweden)

Paulo F. Cristaldo

2015-12-01

Full Text Available Alarm signalling is of paramount importance to communication in all social insects. In termites, vibroacoustic and chemical alarm signalling are bound to operate synergistically but have never been studied simultaneously in a single species. Here, we inspected the functional significance of both communication channels in Constrictotermes cyphergaster (Termitidae: Nasutitermitinae, confirming the hypothesis that these are not exclusive, but rather complementary processes. In natural situations, the alarm predominantly attracts soldiers, which actively search for the source of a disturbance. Laboratory testing revealed that the frontal gland of soldiers produces a rich mixture of terpenoid compounds including an alarm pheromone. Extensive testing led to identification of the alarm pheromone being composed of abundant monoterpene hydrocarbons (1S-α-pinene and myrcene, along with a minor component, (E-β-ocimene. The vibratory alarm signalling consists of vibratory movements evidenced as bursts; a series of beats produced predominantly by soldiers. Exposing termite groups to various mixtures containing the alarm pheromone (crushed soldier heads, frontal gland extracts, mixture of all monoterpenes, and the alarm pheromone mixture made of standards resulted in significantly higher activity in the tested groups and also increased intensity of the vibratory alarm communication, with the responses clearly dose-dependent. Lower doses of the pheromone provoked higher numbers of vibratory signals compared to higher doses. Higher doses induced long-term running of all termites without stops necessary to perform vibratory behaviour. Surprisingly, even crushed worker heads led to low (but significant increases in the alarm responses, suggesting that other unknown compound in the worker's head is perceived and answered by termites. Our results demonstrate the existence of different alarm levels in termites, with lower levels being communicated through

Audiovisual Speech Synchrony Measure: Application to Biometrics

Directory of Open Access Journals (Sweden)

Gérard Chollet

2007-01-01

Full Text Available Speech is a means of communication which is intrinsically bimodal: the audio signal originates from the dynamics of the articulators. This paper reviews recent works in the field of audiovisual speech, and more specifically techniques developed to measure the level of correspondence between audio and visual speech. It overviews the most common audio and visual speech front-end processing, transformations performed on audio, visual, or joint audiovisual feature spaces, and the actual measure of correspondence between audio and visual speech. Finally, the use of synchrony measure for biometric identity verification based on talking faces is experimented on the BANCA database.

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.

Multimodalité et expression en langue étrangère dans une plate-forme audio-synchrone Multimodality and foreign language ouput

Directory of Open Access Journals (Sweden)

Thierry Chanier

2006-06-01

Full Text Available Le développement actuel d'environnements audio-synchrones pose un nouveau défi, non seulement aux concepteurs et aux tuteurs de formations en langues à distance, mais encore aux chercheurs en analyse du discours. Il devient en effet indispensable de comprendre comment s'organise la communication à finalité pédagogique dans ce type d'environnement. L'article aborde ce champ encore peu exploré sous deux angles : l'un méthodologique et l'autre analytique. Nous fournissons, d'une part, un cadre méthodologique pour l'analyse conversationnelle multimodale à partir des notions de média, mode et modalité et, d'autre part, nous définissons les composantes d'une plate-forme synchrone. La partie analytique synthétise d'abord les premiers résultats quantitatifs de l'expérimentation Copéas sur l'usage combiné des modalités audio et clavardage et son impact sur la participation des apprenants. L'analyse indique ensuite, en s'appuyant sur des séquences transcrites, comment s'organise la communication pédagogique lorsque les différents modes (verbaux et non verbaux s'associent pour structurer et soutenir des conversations en L2. Elle montre que le libre choix des modes et des modalités soutient la production verbale des apprenants. Dans ces conversations, le mode parole occupe tantôt une place prépondérante, tantôt est totalement absent des échanges au profit des modes texte, graphique et iconique. Enfin, nous élargissons le champ de description du discours multimodal en considérant le mode spatial qui s'impose comme le contexte dans lequel chaque transaction doit être située. Ces différents exemples constituent l'occasion de mettre en évidence deux questions connexes : celle de la conception et de l'animation pédagogique dans ces environnements de communication multimodaux à travers le phénomène de polyfocalisation ; et celle de la transcription du discours multimodal.The current development of audio

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...

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.

Scalable and Anonymous Group Communication with MTor

Directory of Open Access Journals (Sweden)

Lin Dong

2016-04-01

Full Text Available This paper presents MTor, a low-latency anonymous group communication system. We construct MTor as an extension to Tor, allowing the construction of multi-source multicast trees on top of the existing Tor infrastructure. MTor does not depend on an external service to broker the group communication, and avoids central points of failure and trust. MTor’s substantial bandwidth savings and graceful scalability enable new classes of anonymous applications that are currently too bandwidth-intensive to be viable through traditional unicast Tor communication-e.g., group file transfer, collaborative editing, streaming video, and real-time audio conferencing.

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...

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)

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

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

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…

A Communication Architecture for an Advanced Extravehicular Mobile Unit

Science.gov (United States)

Ivancic, William D.; Sands, Obed S.; Bakula, Casey J.; Oldham, Daniel R.; Wright, Ted; Bradish, Martin A.; Klebau, Joseph M.

2014-01-01

This document describes the communication architecture for the Power, Avionics and Software (PAS) 1.0 subsystem for the Advanced Extravehicular Mobility Unit (AEMU). The following systems are described in detail: Caution Warning and Control System, Informatics, Storage, Video, Audio, Communication, and Monitoring Test and Validation. This document also provides some background as well as the purpose and goals of the PAS subsystem being developed at Glenn Research Center (GRC).

Detecting Parkinson's disease from sustained phonation and speech signals.

Directory of Open Access Journals (Sweden)

Evaldas Vaiciukynas

Full Text Available This study investigates signals from sustained phonation and text-dependent speech modalities for Parkinson's disease screening. Phonation corresponds to the vowel /a/ voicing task and speech to the pronunciation of a short sentence in Lithuanian language. Signals were recorded through two channels simultaneously, namely, acoustic cardioid (AC and smart phone (SP microphones. Additional modalities were obtained by splitting speech recording into voiced and unvoiced parts. Information in each modality is summarized by 18 well-known audio feature sets. Random forest (RF is used as a machine learning algorithm, both for individual feature sets and for decision-level fusion. Detection performance is measured by the out-of-bag equal error rate (EER and the cost of log-likelihood-ratio. Essentia audio feature set was the best using the AC speech modality and YAAFE audio feature set was the best using the SP unvoiced modality, achieving EER of 20.30% and 25.57%, respectively. Fusion of all feature sets and modalities resulted in EER of 19.27% for the AC and 23.00% for the SP channel. Non-linear projection of a RF-based proximity matrix into the 2D space enriched medical decision support by visualization.

[Media for 21st century--towards human communication media].

Science.gov (United States)

Harashima, H

2000-05-01

Today, with the approach of the 21st century, attention is focused on multi-media communications combining computer, visual and audio technologies. This article discusses the communication media target and the technological problems constituting the nucleus of multi-media. The communication media is becoming an environment from which no one can escape. Since the media has such a great power, what is needed now is not to predict the future technologies, but to estimate the future world and take to responsibility for future environments.

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...

Distortion Analysis Toolkit—A Software Tool for Easy Analysis of Nonlinear Audio Systems

Directory of Open Access Journals (Sweden)

Jyri Pakarinen

2010-01-01

Full Text Available Several audio effects devices deliberately add nonlinear distortion to the processed signal in order to create a desired sound. When creating virtual analog models of nonlinearly distorting devices, it would be very useful to carefully analyze the type of distortion, so that the model could be made as realistic as possible. While traditional system analysis tools such as the frequency response give detailed information on the operation of linear and time-invariant systems, they are less useful for analyzing nonlinear devices. Furthermore, although there do exist separate algorithms for nonlinear distortion analysis, there is currently no unified, easy-to-use tool for rapid analysis of distorting audio systems. This paper offers a remedy by introducing a new software tool for easy analysis of distorting effects. A comparison between a well-known guitar tube amplifier and two commercial software simulations is presented as a case study. This freely available software is written in Matlab language, but the analysis tool can also run as a standalone program, so the user does not need to have Matlab installed in order to perform the analysis.

EEG Recording and Online Signal Processing on Android: A Multiapp Framework for Brain-Computer Interfaces on Smartphone.

Science.gov (United States)

Blum, Sarah; Debener, Stefan; Emkes, Reiner; Volkening, Nils; Fudickar, Sebastian; Bleichner, Martin G

2017-01-01

Our aim was the development and validation of a modular signal processing and classification application enabling online electroencephalography (EEG) signal processing on off-the-shelf mobile Android devices. The software application SCALA (Signal ProCessing and CLassification on Android) supports a standardized communication interface to exchange information with external software and hardware. In order to implement a closed-loop brain-computer interface (BCI) on the smartphone, we used a multiapp framework, which integrates applications for stimulus presentation, data acquisition, data processing, classification, and delivery of feedback to the user. We have implemented the open source signal processing application SCALA. We present timing test results supporting sufficient temporal precision of audio events. We also validate SCALA with a well-established auditory selective attention paradigm and report above chance level classification results for all participants. Regarding the 24-channel EEG signal quality, evaluation results confirm typical sound onset auditory evoked potentials as well as cognitive event-related potentials that differentiate between correct and incorrect task performance feedback. We present a fully smartphone-operated, modular closed-loop BCI system that can be combined with different EEG amplifiers and can easily implement other paradigms.

EEG Recording and Online Signal Processing on Android: A Multiapp Framework for Brain-Computer Interfaces on Smartphone

Directory of Open Access Journals (Sweden)

Sarah Blum

2017-01-01

Full Text Available Objective. Our aim was the development and validation of a modular signal processing and classification application enabling online electroencephalography (EEG signal processing on off-the-shelf mobile Android devices. The software application SCALA (Signal ProCessing and CLassification on Android supports a standardized communication interface to exchange information with external software and hardware. Approach. In order to implement a closed-loop brain-computer interface (BCI on the smartphone, we used a multiapp framework, which integrates applications for stimulus presentation, data acquisition, data processing, classification, and delivery of feedback to the user. Main Results. We have implemented the open source signal processing application SCALA. We present timing test results supporting sufficient temporal precision of audio events. We also validate SCALA with a well-established auditory selective attention paradigm and report above chance level classification results for all participants. Regarding the 24-channel EEG signal quality, evaluation results confirm typical sound onset auditory evoked potentials as well as cognitive event-related potentials that differentiate between correct and incorrect task performance feedback. Significance. We present a fully smartphone-operated, modular closed-loop BCI system that can be combined with different EEG amplifiers and can easily implement other paradigms.

EEG Recording and Online Signal Processing on Android: A Multiapp Framework for Brain-Computer Interfaces on Smartphone

Science.gov (United States)

Debener, Stefan; Emkes, Reiner; Volkening, Nils; Fudickar, Sebastian; Bleichner, Martin G.

2017-01-01

Objective Our aim was the development and validation of a modular signal processing and classification application enabling online electroencephalography (EEG) signal processing on off-the-shelf mobile Android devices. The software application SCALA (Signal ProCessing and CLassification on Android) supports a standardized communication interface to exchange information with external software and hardware. Approach In order to implement a closed-loop brain-computer interface (BCI) on the smartphone, we used a multiapp framework, which integrates applications for stimulus presentation, data acquisition, data processing, classification, and delivery of feedback to the user. Main Results We have implemented the open source signal processing application SCALA. We present timing test results supporting sufficient temporal precision of audio events. We also validate SCALA with a well-established auditory selective attention paradigm and report above chance level classification results for all participants. Regarding the 24-channel EEG signal quality, evaluation results confirm typical sound onset auditory evoked potentials as well as cognitive event-related potentials that differentiate between correct and incorrect task performance feedback. Significance We present a fully smartphone-operated, modular closed-loop BCI system that can be combined with different EEG amplifiers and can easily implement other paradigms. PMID:29349070

Signal Quality Outage Analysis for Ultra-Reliable Communications in Cellular Networks

DEFF Research Database (Denmark)

Gerardino, Guillermo Andrés Pocovi; Alvarez, Beatriz Soret; Lauridsen, Mads

2015-01-01

Ultra-reliable communications over wireless will open the possibility for a wide range of novel use cases and applications. In cellular networks, achieving reliable communication is challenging due to many factors, particularly the fading of the desired signal and the interference. In this regard......, we investigate the potential of several techniques to combat these main threats. The analysis shows that traditional microscopic multiple-input multiple-output schemes with 2x2 or 4x4 antenna configurations are not enough to fulfil stringent reliability requirements. It is revealed how such antenna...... schemes must be complemented with macroscopic diversity as well as interference management techniques in order to ensure the necessary SINR outage performance. Based on the obtained performance results, it is discussed which of the feasible options fulfilling the ultra-reliable criteria are most promising...

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...

Phenotypic integration and the evolution of signal repertoires: A case study of treefrog acoustic communication.

Science.gov (United States)

Reichert, Michael S; Höbel, Gerlinde

2018-03-01

Animal signals are inherently complex phenotypes with many interacting parts combining to elicit responses from receivers. The pattern of interrelationships between signal components reflects the extent to which each component is expressed, and responds to selection, either in concert with or independently of others. Furthermore, many species have complex repertoires consisting of multiple signal types used in different contexts, and common morphological and physiological constraints may result in interrelationships extending across the multiple signals in species' repertoires. The evolutionary significance of interrelationships between signal traits can be explored within the framework of phenotypic integration, which offers a suite of quantitative techniques to characterize complex phenotypes. In particular, these techniques allow for the assessment of modularity and integration, which describe, respectively, the extent to which sets of traits covary either independently or jointly. Although signal and repertoire complexity are thought to be major drivers of diversification and social evolution, few studies have explicitly measured the phenotypic integration of signals to investigate the evolution of diverse communication systems. We applied methods from phenotypic integration studies to quantify integration in the two primary vocalization types (advertisement and aggressive calls) in the treefrogs Hyla versicolor , Hyla cinerea, and Dendropsophus ebraccatus . We recorded male calls and calculated standardized phenotypic variance-covariance ( P ) matrices for characteristics within and across call types. We found significant integration across call types, but the strength of integration varied by species and corresponded with the acoustic similarity of the call types within each species. H. versicolor had the most modular advertisement and aggressive calls and the least acoustically similar call types. Additionally, P was robust to changing social competition

Contribution of Prosody in Audio-Visual Integration to Emotional Perception of Virtual Characters

Directory of Open Access Journals (Sweden)

Ekaterina Volkova

2011-10-01

Full Text Available Recent technology provides us with realistic looking virtual characters. Motion capture and elaborate mathematical models supply data for natural looking, controllable facial and bodily animations. With the help of computational linguistics and artificial intelligence, we can automatically assign emotional categories to appropriate stretches of text for a simulation of those social scenarios where verbal communication is important. All this makes virtual characters a valuable tool for creation of versatile stimuli for research on the integration of emotion information from different modalities. We conducted an audio-visual experiment to investigate the differential contributions of emotional speech and facial expressions on emotion identification. We used recorded and synthesized speech as well as dynamic virtual faces, all enhanced for seven emotional categories. The participants were asked to recognize the prevalent emotion of paired faces and audio. Results showed that when the voice was recorded, the vocalized emotion influenced participants' emotion identification more than the facial expression. However, when the voice was synthesized, facial expression influenced participants' emotion identification more than vocalized emotion. Additionally, individuals did worse on identifying either the facial expression or vocalized emotion when the voice was synthesized. Our experimental method can help to determine how to improve synthesized emotional speech.

A combined model of sensory and cognitive representations underlying tonal expectations in music: from audio signals to behavior.

Science.gov (United States)

Collins, Tom; Tillmann, Barbara; Barrett, Frederick S; Delbé, Charles; Janata, Petr

2014-01-01

Listeners' expectations for melodies and harmonies in tonal music are perhaps the most studied aspect of music cognition. Long debated has been whether faster response times (RTs) to more strongly primed events (in a music theoretic sense) are driven by sensory or cognitive mechanisms, such as repetition of sensory information or activation of cognitive schemata that reflect learned tonal knowledge, respectively. We analyzed over 300 stimuli from 7 priming experiments comprising a broad range of musical material, using a model that transforms raw audio signals through a series of plausible physiological and psychological representations spanning a sensory-cognitive continuum. We show that RTs are modeled, in part, by information in periodicity pitch distributions, chroma vectors, and activations of tonal space--a representation on a toroidal surface of the major/minor key relationships in Western tonal music. We show that in tonal space, melodies are grouped by their tonal rather than timbral properties, whereas the reverse is true for the periodicity pitch representation. While tonal space variables explained more of the variation in RTs than did periodicity pitch variables, suggesting a greater contribution of cognitive influences to tonal expectation, a stepwise selection model contained variables from both representations and successfully explained the pattern of RTs across stimulus categories in 4 of the 7 experiments. The addition of closure--a cognitive representation of a specific syntactic relationship--succeeded in explaining results from all 7 experiments. We conclude that multiple representational stages along a sensory-cognitive continuum combine to shape tonal expectations in music. (PsycINFO Database Record (c) 2014 APA, all rights reserved).

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…

Signaling in a polluted world: oxidative stress as an overlooked mechanism linking contaminants to animal communication

OpenAIRE

Valeria Marasco; David Costantini; David Costantini

2016-01-01

The capacity to communicate effectively with other individuals plays a critical role in the daily life of an individual and can have important fitness consequences. Animals rely on a number of visual and non-visual signals, whose production brings costs to the individual. The theory of honest signaling states that these costs are higher for low than for high-quality individuals, which prevents cheating and makes signals, such as skin and plumage colouration, indicators of individual’s quality...

Signaling in a Polluted World: Oxidative Stress as an Overlooked Mechanism Linking Contaminants to Animal Communication

OpenAIRE

Marasco, Valeria; Costantini, David

2016-01-01

The capacity to communicate effectively with other individuals plays a critical role in the daily life of an individual and can have important fitness consequences. Animals rely on a number of visual and non-visual signals, whose production brings costs to the individual. The theory of honest signaling states that these costs are higher for low than for high-quality individuals, which prevents cheating and makes signals, such as skin and plumage coloration, indicators of individual's quality ...

Aerospace Communications Security Technologies Demonstrated

Science.gov (United States)

Griner, James H.; Martzaklis, Konstantinos S.

2003-01-01

In light of the events of September 11, 2001, NASA senior management requested an investigation of technologies and concepts to enhance aviation security. The investigation was to focus on near-term technologies that could be demonstrated within 90 days and implemented in less than 2 years. In response to this request, an internal NASA Glenn Research Center Communications, Navigation, and Surveillance Aviation Security Tiger Team was assembled. The 2-year plan developed by the team included an investigation of multiple aviation security concepts, multiple aircraft platforms, and extensively leveraged datalink communications technologies. It incorporated industry partners from NASA's Graphical Weather-in-the-Cockpit research, which is within NASA's Aviation Safety Program. Two concepts from the plan were selected for demonstration: remote "black box," and cockpit/cabin surveillance. The remote "black box" concept involves real-time downlinking of aircraft parameters for remote monitoring and archiving of aircraft data, which would assure access to the data following the loss or inaccessibility of an aircraft. The cockpit/cabin surveillance concept involves remote audio and/or visual surveillance of cockpit and cabin activity, which would allow immediate response to any security breach and would serve as a possible deterrent to such breaches. The datalink selected for the demonstrations was VDL Mode 2 (VHF digital link), the first digital datalink for air-ground communications designed for aircraft use. VDL Mode 2 is beginning to be implemented through the deployment of ground stations and aircraft avionics installations, with the goal of being operational in 2 years. The first demonstration was performed December 3, 2001, onboard the LearJet 25 at Glenn. NASA worked with Honeywell, Inc., for the broadcast VDL Mode 2 datalink capability and with actual Boeing 757 aircraft data. This demonstration used a cockpitmounted camera for video surveillance and a coupling to

Quorum sensing communication between bacteria and human cells: signals, targets and functions

Directory of Open Access Journals (Sweden)

Angelika eHolm

2014-06-01

Full Text Available Both direct and long-range interactions between pathogenic Pseudomonas aeruginosa bacteria and their eukaryotic hosts are important in the outcome of infections. For cell-to-cell communication, these bacteria employ the quorum sensing (QS system to pass on information of the density of the bacterial population and collectively switch on virulence factor production, biofilm formation and resistance development. Thus, QS allows bacteria to behave as a community to perform tasks which would be impossible for individual cells, e.g. to overcome defense and immune systems and establish infections in higher organisms. This review highlights these aspects of QS and our own recent research on how P.aeruginosa communicates with human cells using the small QS signal molecules N-acyl homoserine lactones (AHL. We focus on how this conversation changes the behavior and function of neutrophils, macrophages and epithelial cells and on how the signaling machinery in human cells responsible for the recognition of AHL. Understanding the bacteria-host relationships at both cellular and molecular levels is essential for the identification of new targets and for the development of novel strategies to fight bacterial infections in the future.

Learning sparse generative models of audiovisual signals

OpenAIRE

Monaci, Gianluca; Sommer, Friedrich T.; Vandergheynst, Pierre

2008-01-01

This paper presents a novel framework to learn sparse represen- tations for audiovisual signals. An audiovisual signal is modeled as a sparse sum of audiovisual kernels. The kernels are bimodal functions made of synchronous audio and video components that can be positioned independently and arbitrarily in space and time. We design an algorithm capable of learning sets of such audiovi- sual, synchronous, shift-invariant functions by alternatingly solving a coding and a learning pr...

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...

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…

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...

Utterance independent bimodal emotion recognition in spontaneous communication

Science.gov (United States)

Tao, Jianhua; Pan, Shifeng; Yang, Minghao; Li, Ya; Mu, Kaihui; Che, Jianfeng

2011-12-01

Emotion expressions sometimes are mixed with the utterance expression in spontaneous face-to-face communication, which makes difficulties for emotion recognition. This article introduces the methods of reducing the utterance influences in visual parameters for the audio-visual-based emotion recognition. The audio and visual channels are first combined under a Multistream Hidden Markov Model (MHMM). Then, the utterance reduction is finished by finding the residual between the real visual parameters and the outputs of the utterance related visual parameters. This article introduces the Fused Hidden Markov Model Inversion method which is trained in the neutral expressed audio-visual corpus to solve the problem. To reduce the computing complexity the inversion model is further simplified to a Gaussian Mixture Model (GMM) mapping. Compared with traditional bimodal emotion recognition methods (e.g., SVM, CART, Boosting), the utterance reduction method can give better results of emotion recognition. The experiments also show the effectiveness of our emotion recognition system when it was used in a live environment.

Propagation Engineering in Wireless Communications

CERN Document Server

Ghasemi, Abdollah; Ghasemi, Farshid

2012-01-01

Wireless communications has seen explosive growth in recent decades, in a realm that is both broad and rapidly expanding to include satellite services, navigational aids, remote sensing, telemetering, audio and video broadcasting, high-speed data communications, mobile radio systems and much more. Propagation Engineering in Wireless Communications deals with the basic principles of radiowaves propagation for frequency bands used in radio-communications, offering descriptions of new achievements and newly developed propagation models. The book bridges the gap between theoretical calculations and approaches, and applied procedures needed for advanced radio links design. The primary objective of this two-volume set is to demonstrate the fundamentals, and to introduce propagation phenomena and mechanisms that engineers are likely to encounter in the design and evaluation of radio links of a given type and operating frequency. Volume one covers basic principles, along with tropospheric and ionospheric propagation,...

Methods for communicating technical information as public information

International Nuclear Information System (INIS)

Zara, S.A.

1987-01-01

Many challenges face the nuclear industry, especially in the waste management area. One of the biggest challenges is effective communication with the general public. Technical complexity, combined with the public's lack of knowledge and negative emotional response, complicate clear communication of radioactive waste management issues. The purpose of this session is to present and discuss methods for overcoming these obstacles and effectively transmitting technical information as public information. The methods presented encompass audio, visual, and print approaches to message transmission. To support these methods, the author also discusses techniques, based on current research, for improving the communication process

Discrimination of acoustic communication signals by grasshoppers (Chorthippus biguttulus): temporal resolution, temporal integration, and the impact of intrinsic noise.

Science.gov (United States)

Ronacher, Bernhard; Wohlgemuth, Sandra; Vogel, Astrid; Krahe, Rüdiger

2008-08-01

A characteristic feature of hearing systems is their ability to resolve both fast and subtle amplitude modulations of acoustic signals. This applies also to grasshoppers, which for mate identification rely mainly on the characteristic temporal patterns of their communication signals. Usually the signals arriving at a receiver are contaminated by various kinds of noise. In addition to extrinsic noise, intrinsic noise caused by stochastic processes within the nervous system contributes to making signal recognition a difficult task. The authors asked to what degree intrinsic noise affects temporal resolution and, particularly, the discrimination of similar acoustic signals. This study aims at exploring the neuronal basis for sexual selection, which depends on exploiting subtle differences between basically similar signals. Applying a metric, by which the similarities of spike trains can be assessed, the authors investigated how well the communication signals of different individuals of the same species could be discriminated and correctly classified based on the responses of auditory neurons. This spike train metric yields clues to the optimal temporal resolution with which spike trains should be evaluated. (c) 2008 APA, all rights reserved

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…