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Sample records for ac-3 audio system

  1. 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...

  2. 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.

  3. Nonlinear dynamic macromodeling techniques for audio systems

    Science.gov (United States)

    Ogrodzki, Jan; Bieńkowski, Piotr

    2015-09-01

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

  4. 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...

  5. 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

  6. 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.

  7. Aging, Counterfeiting Configuration Control (AC3)

    Science.gov (United States)

    2010-01-31

    Systems Intergrated Into AC3 CABS - Common As-Built System PRISM - Process Re-inventing Integration Systems for Manufacturing PDM - Product Data...looks forward to deploying the completed tool at Raytheon in a true production environment, for as much as we like the challenge associated with...performance of DoD systems. DoD systems are particularly susceptible to intrusion of counterfeit parts, especially during surge and extended production

  8. 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...

  9. 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...

  10. 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...

  11. 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

  12. 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.

  13. Music Identification System Using MPEG-7 Audio Signature Descriptors

    Science.gov (United States)

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

    2013-01-01

    This paper describes a multiresolution system based on MPEG-7 audio signature descriptors for music identification. Such an identification system may be used to detect illegally copied music circulated over the Internet. In the proposed system, low-resolution descriptors are used to search likely candidates, and then full-resolution descriptors are used to identify the unknown (query) audio. With this arrangement, the proposed system achieves both high speed and high accuracy. To deal with the problem that a piece of query audio may not be inside the system's database, we suggest two different methods to find the decision threshold. Simulation results show that the proposed method II can achieve an accuracy of 99.4% for query inputs both inside and outside the database. Overall, it is highly possible to use the proposed system for copyright control. PMID:23533359

  14. Music Identification System Using MPEG-7 Audio Signature Descriptors

    Directory of Open Access Journals (Sweden)

    Shingchern D. You

    2013-01-01

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

  15. Robustness evaluation of transactional audio watermarking systems

    Science.gov (United States)

    Neubauer, Christian; Steinebach, Martin; Siebenhaar, Frank; Pickel, Joerg

    2003-06-01

    Distribution via Internet is of increasing importance. Easy access, transmission and consumption of digitally represented music is very attractive to the consumer but led also directly to an increasing problem of illegal copying. To cope with this problem watermarking is a promising concept since it provides a useful mechanism to track illicit copies by persistently attaching property rights information to the material. Especially for online music distribution the use of so-called transaction watermarking, also denoted with the term bitstream watermarking, is beneficial since it offers the opportunity to embed watermarks directly into perceptually encoded material without the need of full decompression/compression. Besides the concept of bitstream watermarking, former publications presented the complexity, the audio quality and the detection performance. These results are now extended by an assessment of the robustness of such schemes. The detection performance before and after applying selected attacks is presented for MPEG-1/2 Layer 3 (MP3) and MPEG-2/4 AAC bitstream watermarking, contrasted to the performance of PCM spread spectrum watermarking.

  16. Audio-Visual Perception System for a Humanoid Robotic Head

    Directory of Open Access Journals (Sweden)

    Raquel Viciana-Abad

    2014-05-01

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

  17. A listening test system for automative audio

    DEFF Research Database (Denmark)

    Bech, Søren; Gulbol, Mehmet-Ali; Martin, Geoff

    2005-01-01

    This paper describes two listening tests that were performed to provide initial validation of an auralisation system (see Part 1) to mimic the acoustics of a car interior. The validation is based on a comparison of results from an in-car listening test and another test using the auralisation system...... and recordings of the stimuli used for the in-car test. The music samples for the test were chosen from a database of various CODEC examples from a previous extensive ITU test to validate the ITU-R BS.1387-1 standard....

  18. A listening test system for automotive audio - listeners

    DEFF Research Database (Denmark)

    Choisel, Sylvain; Hegarty, Patrick; Christensen, Flemming

    2007-01-01

    A series of experiments was conducted in order to validate an experimental procedure to perform listening tests on car audio systems in a simulation of the car environment in a laboratory, using binaural synthesis with head-tracking. Seven experts and 40 non-expert listeners rated a range...... of stimuli for 15 sound-quality attributes developed by the experts. This paper presents a comparison between the attribute ratings from the two groups of participants. Overall preference of the non-experts was also measured using direct ratings as well as indirect scaling based on paired comparisons...

  19. 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.

  20. A Smart Audio on Demand Application on Android Systems

    Directory of Open Access Journals (Sweden)

    Ing-Jr Ding

    2015-05-01

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

  1. 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.

  2. 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 ...

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

    DEFF Research Database (Denmark)

    Dimitrov, Smilen; Serafin, Stefania

    2011-01-01

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

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

    Energy Technology Data Exchange (ETDEWEB)

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

    2001-11-01

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

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

    International Nuclear Information System (INIS)

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

    2001-01-01

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

  6. System-Level Optimization of a DAC for Hearing-Aid Audio Class D Output Stage

    DEFF Research Database (Denmark)

    Pracný, Peter; Jørgensen, Ivan Harald Holger; Bruun, Erik

    2013-01-01

    This paper deals with system-level optimization of a digital-to-analog converter (DAC) for hearing-aid audio Class D output stage. We discuss the ΣΔ modulator system-level design parameters – the order, the oversampling ratio (OSR) and the number of bits in the quantizer. We show that combining...... by comparing two ΣΔ modulator designs. The proposed optimization has impact on the whole hearing-aid audio back-end system including less hardware in the interpolation filter and half the switching rate in the digital-pulse-width-modulation (DPWM) block and Class D output stage...... a reduction of the OSR with an increase of the order results in considerable power savings while the audio quality is kept. For further savings in the ΣΔ modulator, overdesign and subsequent coarse coefficient quantization are used. A figure of merit (FOM) is introduced to confirm this optimization approach...

  7. 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.

  8. A method for Perceptual Assessment of Automotive Audio Systems and Cabin Acoustics

    DEFF Research Database (Denmark)

    Kaplanis, Neofytos; Bech, Søren; Sakari, Tervo

    2016-01-01

    This paper reports the design and implementation of a method to perceptually assess the acoustical prop- erties of a car cabin and the subsequent sound reproduction properties of automotive audio systems. Here, we combine Spatial Decomposition Method and Rapid Sensory Analysis techniques. The for......This paper reports the design and implementation of a method to perceptually assess the acoustical prop- erties of a car cabin and the subsequent sound reproduction properties of automotive audio systems. Here, we combine Spatial Decomposition Method and Rapid Sensory Analysis techniques...

  9. 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

  10. The perceptual influence of the cabin acoustics on the reproduced sound of a car audio system

    DEFF Research Database (Denmark)

    Kaplanis, Neofytos; Bech, Søren; Sakari, Tervo

    2015-01-01

    -end car audio system was performed for different physical settings of the car's cabin. A novel spatial auralization methodology was then used, and participants were asked to describe verbally the perceived acoustical characteristics of the stimuli. The elicited attributes were then analyzed following...... a previous review [Kaplanis et al., in 55th Int. Conf. Aud. Eng. Soc. (2014)] and possible links to the acoustical properties of the car cabin are discussed. [This study is a part of Marie Curie Network on Dereverberation and Reverberation of Audio, Music, and Speech. EU-FP7 under agreement ITN-GA-2012-316969.]...

  11. 76 FR 79755 - First Meeting: RTCA Special Committee 226 Audio Systems and Equipment

    Science.gov (United States)

    2011-12-22

    ... Administrative Remarks Introductions RTCA Overview Audio Systems and Equipment--Background and History Agenda..., Discussion, Recommendations and Assignment of Responsibilities Other Business Establish Agenda for Next..., Manager, Business Operations Branch, Federal Aviation Administration. [FR Doc. 2011-32863 Filed 12-21-11...

  12. 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...

  13. 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.

  14. Audio Key Finding: Considerations in System Design and Case Studies on Chopin's 24 Preludes

    Directory of Open Access Journals (Sweden)

    Elaine Chew

    2007-01-01

    Full Text Available We systematically analyze audio key finding to determine factors important to system design, and the selection and evaluation of solutions. First, we present a basic system, fuzzy analysis spiral array center of effect generator algorithm, with three key determination policies: nearest-neighbor (NN, relative distance (RD, and average distance (AD. AD achieved a 79% accuracy rate in an evaluation on 410 classical pieces, more than 8% higher RD and NN. We show why audio key finding sometimes outperforms symbolic key finding. We next propose three extensions to the basic key finding system—the modified spiral array (mSA, fundamental frequency identification (F0, and post-weight balancing (PWB—to improve performance, with evaluations using Chopin's Preludes (Romantic repertoire was the most challenging. F0 provided the greatest improvement in the first 8 seconds, while mSA gave the best performance after 8 seconds. Case studies examine when all systems were correct, or all incorrect.

  15. Audio system using binaural synthesis for multimodal telepresence applications

    DEFF Research Database (Denmark)

    Madsen, Esben; Markovic, Milos; Olesen, Søren Krarup

    2013-01-01

    are implemented in a distributed manner. Body-tracking of all participants is provided through the system for the purpose of using binaural synthesis for directional sound. Head-worn microphones are used to capture sound, and the visitor is provided with directional sound through headphones. The visitor...

  16. Automatic Speech Acquisition and Recognition for Spacesuit Audio Systems

    Science.gov (United States)

    Ye, Sherry

    2015-01-01

    NASA has a widely recognized but unmet need for novel human-machine interface technologies that can facilitate communication during astronaut extravehicular activities (EVAs), when loud noises and strong reverberations inside spacesuits make communication challenging. WeVoice, Inc., has developed a multichannel signal-processing method for speech acquisition in noisy and reverberant environments that enables automatic speech recognition (ASR) technology inside spacesuits. The technology reduces noise by exploiting differences between the statistical nature of signals (i.e., speech) and noise that exists in the spatial and temporal domains. As a result, ASR accuracy can be improved to the level at which crewmembers will find the speech interface useful. System components and features include beam forming/multichannel noise reduction, single-channel noise reduction, speech feature extraction, feature transformation and normalization, feature compression, and ASR decoding. Arithmetic complexity models were developed and will help designers of real-time ASR systems select proper tasks when confronted with constraints in computational resources. In Phase I of the project, WeVoice validated the technology. The company further refined the technology in Phase II and developed a prototype for testing and use by suited astronauts.

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

    Science.gov (United States)

    2004-11-01

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

  18. 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.

  19. Extraction Of Audio Features For Emotion Recognition System Based On Music

    Directory of Open Access Journals (Sweden)

    Kee Moe Han

    2015-08-01

    Full Text Available Music is the combination of melody linguistic information and the vocalists emotion. Since music is a work of art analyzing emotion in music by computer is a difficult task. Many approaches have been developed to detect the emotions included in music but the results are not satisfactory because emotion is very complex. In this paper the evaluations of audio features from the music files are presented. The extracted features are used to classify the different emotion classes of the vocalists. Musical features extraction is done by using Music Information Retrieval MIR tool box in this paper. The database of 100 music clips are used to classify the emotions perceived in music clips. Music may contain many emotions according to the vocalists mood such as happy sad nervous bored peace etc. In this paper the audio features related to the emotions of the vocalists are extracted to use in emotion recognition system based on music.

  20. 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.

  1. 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)

  2. Blind speech separation system for humanoid robot with FastICA for audio filtering and separation

    Science.gov (United States)

    Budiharto, Widodo; Santoso Gunawan, Alexander Agung

    2016-07-01

    Nowadays, there are many developments in building intelligent humanoid robot, mainly in order to handle voice and image. In this research, we propose blind speech separation system using FastICA for audio filtering and separation that can be used in education or entertainment. Our main problem is to separate the multi speech sources and also to filter irrelevant noises. After speech separation step, the results will be integrated with our previous speech and face recognition system which is based on Bioloid GP robot and Raspberry Pi 2 as controller. The experimental results show the accuracy of our blind speech separation system is about 88% in command and query recognition cases.

  3. Exploration of a digital audio processing platform using a compositional system level performance estimation framework

    DEFF Research Database (Denmark)

    Tranberg-Hansen, Anders Sejer; Madsen, Jan

    2009-01-01

    This paper presents the application of a compositional simulation based system-level performance estimation framework on a non-trivial industrial case study. The case study is provided by the Danish company Bang & Olufsen ICEpower a/s and focuses on the exploration of a digital mobile audio...... processing platform. A short overview of the compositional performance estimation framework used is given followed by a presentation of how it is used for performance estimation using an iterative refinement process towards the final implementation. Finally, an evaluation in terms of accuracy and speed...

  4. System Level Power Optimization of Digital Audio Back End for Hearing Aids

    DEFF Research Database (Denmark)

    Pracny, Peter; Jørgensen, Ivan Harald Holger; Bruun, Erik

    2017-01-01

    This work deals with power optimization of the audio processing back end for hearing aids - the interpolation filter (IF), the sigma-delta (SD modulator and the Class D power amplifier (PA) as a whole. Specifications are derived and insight into the tradeoffs involved is used to optimize...... the interpolation filter and the SD modulator on the system level so that the switching frequency of the Class D PA - the main power consumer in the back end - is minimized. A figure-of-merit (FOM) which allows judging the power consumption of the digital part of the back end early in the design process is used...

  5. 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....

  6. 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.

  7. Safety of the HyperSound® Audio System in subjects with normal hearing

    Directory of Open Access Journals (Sweden)

    Ritvik P. Mehta

    2015-11-01

    Full Text Available The objective of the study was to assess the safety of the HyperSound® Audio System (HSS, a novel audio system using ultrasound technology, in normal hearing subjects under normal use conditions; we considered preexposure and post-exposure test design. We investigated primary and secondary outcome measures: i temporary threshold shift (TTS, defined as >10 dB shift in pure tone air conduction thresholds and/or a decrement in distortion product otoacoustic emissions (DPOAEs >10 dB at two or more frequencies; ii presence of new-onset otologic symptoms after exposure. Twenty adult subjects with normal hearing underwent a pre-exposure assessment (pure tone air conduction audiometry, tympanometry, DPOAEs and otologic symptoms questionnaire followed by exposure to a 2-h movie with sound delivered through the HSS emitter followed by a post-exposure assessment. No TTS or new-onset otological symptoms were identified. HSS demonstrates excellent safety in normal hearing subjects under normal use conditions.

  8. Safety of the HyperSound® Audio System in Subjects with Normal Hearing.

    Science.gov (United States)

    Mehta, Ritvik P; Mattson, Sara L; Kappus, Brian A; Seitzman, Robin L

    2015-06-11

    The objective of the study was to assess the safety of the HyperSound® Audio System (HSS), a novel audio system using ultrasound technology, in normal hearing subjects under normal use conditions; we considered pre-exposure and post-exposure test design. We investigated primary and secondary outcome measures: i) temporary threshold shift (TTS), defined as >10 dB shift in pure tone air conduction thresholds and/or a decrement in distortion product otoacoustic emissions (DPOAEs) >10 dB at two or more frequencies; ii) presence of new-onset otologic symptoms after exposure. Twenty adult subjects with normal hearing underwent a pre-exposure assessment (pure tone air conduction audiometry, tympanometry, DPOAEs and otologic symptoms questionnaire) followed by exposure to a 2-h movie with sound delivered through the HSS emitter followed by a post-exposure assessment. No TTS or new-onset otological symptoms were identified. HSS demonstrates excellent safety in normal hearing subjects under normal use conditions.

  9. 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.

  10. 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.

  11. Synthetic Modeling of A Geothermal System Using Audio-magnetotelluric (AMT) and Magnetotelluric (MT)

    Science.gov (United States)

    Mega Saputra, Rifki; Widodo

    2017-04-01

    Indonesia has 40% of the world’s potential geothermal resources with estimated capacity of 28,910 MW. Generally, the characteristic of the geothermal system in Indonesia is liquid-dominated systems, which driven by volcanic activities. In geothermal exploration, electromagnetic methods are used to map structures that could host potential reservoirs and source rocks. We want to know the responses of a geothermal system using synthetic data of Audio-magnetotelluric (AMT) and Magnetotelluric (MT). Due to frequency range, AMT and MT data can resolve the shallow and deeper structure, respectively. 1-D models have been performed using AMT and MT data. The results indicate that AMT and MT data give detailed conductivity distribution of geothermal structure.

  12. 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.

  13. A Low-Cost Audio Prescription Labeling System Using RFID for Thai Visually-Impaired People.

    Science.gov (United States)

    Lertwiriyaprapa, Titipong; Fakkheow, Pirapong

    2015-01-01

    This research aims to develop a low-cost audio prescription labeling (APL) system for visually-impaired people by using the RFID system. The developed APL system includes the APL machine and APL software. The APL machine is for visually-impaired people while APL software allows caregivers to record all important information into the APL machine. The main objective of the development of the APL machine is to reduce costs and size by designing all of the electronic devices to fit into one print circuit board. Also, it is designed so that it is easy to use and can become an electronic aid for daily living. The developed APL software is based on Java and MySQL, both of which can operate on various operating platforms and are easy to develop as commercial software. The developed APL system was first evaluated by 5 experts. The APL system was also evaluated by 50 actual visually-impaired people (30 elders and 20 blind individuals) and 20 caregivers, pharmacists and nurses. After using the APL system, evaluations were carried out, and it can be concluded from the evaluation results that this proposed APL system can be effectively used for helping visually-impaired people in terms of self-medication.

  14. Audio gunshot detection and localization systems: History, basic design, and future possibilities

    Science.gov (United States)

    Graves, Jordan R.

    For decades, law enforcement organizations have increasingly utilized audio detection and localization systems to identify potential gunshot incidents and to respond accordingly. These systems have grown from simple microphone configurations used to estimate location into complex arrays that seem to pinpoint gunfire to within mere feet of its actual occurrence. Such technology comes from a long and dynamic history of developing equipment dating back to the First World War. Additionally, though basic designs require little in terms of programming or engineering experience, the mere presence of this tool invokes a firestorm of debate amongst economists, law enforcement groups, and the general public, which leads to questions about future possibilities for its use. The following pages will retell the history of these systems from theoretical conception to current capabilities. This work will also dissect these systems to reveal fundamental elements of their inner workings, in order to build a basic demonstrative system. Finally, this work will discuss some legal and moral points of dissension, and will explore these systems’ roles in society now and in the future, in additional applications as well.

  15. A technology prototype system for rating therapist empathy from audio recordings in addiction counseling.

    Science.gov (United States)

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

    2016-04-01

    Scaling up psychotherapy services such as for addiction counseling is a critical societal need. One challenge is ensuring quality of therapy, due to the heavy cost of manual observational assessment. This work proposes a speech technology-based system to automate the assessment of therapist empathy-a key therapy quality index-from audio recordings of the psychotherapy interactions. We designed a speech processing system that includes voice activity detection and diarization modules, and an automatic speech recognizer plus a speaker role matching module to extract the therapist's language cues. We employed Maximum Entropy models, Maximum Likelihood language models, and a Lattice Rescoring method to characterize high vs. low empathic language. We estimated therapy-session level empathy codes using utterance level evidence obtained from these models. Our experiments showed that the fully automated system achieved a correlation of 0.643 between expert annotated empathy codes and machine-derived estimations, and an accuracy of 81% in classifying high vs. low empathy, in comparison to a 0.721 correlation and 86% accuracy in the oracle setting using manual transcripts. The results show that the system provides useful information that can contribute to automatic quality insurance and therapist training.

  16. A technology prototype system for rating therapist empathy from audio recordings in addiction counseling

    Directory of Open Access Journals (Sweden)

    Bo Xiao

    2016-04-01

    Full Text Available Scaling up psychotherapy services such as for addiction counseling is a critical societal need. One challenge is ensuring quality of therapy, due to the heavy cost of manual observational assessment. This work proposes a speech technology-based system to automate the assessment of therapist empathy—a key therapy quality index—from audio recordings of the psychotherapy interactions. We designed a speech processing system that includes voice activity detection and diarization modules, and an automatic speech recognizer plus a speaker role matching module to extract the therapist’s language cues. We employed Maximum Entropy models, Maximum Likelihood language models, and a Lattice Rescoring method to characterize high vs. low empathic language. We estimated therapy-session level empathy codes using utterance level evidence obtained from these models. Our experiments showed that the fully automated system achieved a correlation of 0.643 between expert annotated empathy codes and machine-derived estimations, and an accuracy of 81% in classifying high vs. low empathy, in comparison to a 0.721 correlation and 86% accuracy in the oracle setting using manual transcripts. The results show that the system provides useful information that can contribute to automatic quality insurance and therapist training.

  17. MRI-compatible audio/visual system: impact on pediatric sedation

    International Nuclear Information System (INIS)

    Harned, R.K. II; Strain, J.D.

    2001-01-01

    Background. While sedation is necessary for much pediatric imaging, there are new alternatives that may help patients hold still without medication. Objective. We examined the effect of an audio/visual system consisting of video goggles and earphones on the need for sedation during magnetic resonance imaging (MRI). Materials and methods. All MRI examinations from May 1999 to October 1999 performed after installation of the MRVision 2000 (Resonance Technology, Inc.) were compared to the same 6-month period in 1998. Imaging and sedation protocols remained constant. Data collected included: patient age, type of examination, use of intravenous contrast enhancement, and need for sedation. The average supply charge and nursing cost per sedated patient were calculated. Results. The 955 patients from 1998 and 1,112 patients from 1999 were similar in demographics and examination distribution. There was an overall reduction in the percent of patients requiring sedation in the group using the video goggle system from 49 to 40 % (P < 0.001). There was no significant change for 0-2 years (P = 0.805), but there was a reduction from 53 to 40 % for age 3-10 years (P < 0.001) and 16 to 8 % for those older than 10 years (P < 0.001). There was a 17 % decrease in MRI room time for those patients whose examinations could be performed without sedation. Sedation costs per patient were $80 for nursing and $29 for supplies. Conclusion. The use of this video system reduced the number of children requiring sedation for MRI examination by 18 %. In addition to reducing patient risk, this can potentially reduce cost. (orig.)

  18. 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...

  19. Frequency dependent loss analysis and minimization of system losses in switchmode audio power amplifiers

    DEFF Research Database (Denmark)

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

    2014-01-01

    In this paper, frequency dependent losses in switch-mode audio power amplifiers are analyzed and a loss model is improved by taking the voltage dependence of the parasitic capacitance of MOSFETs into account. The estimated power losses are compared to the measurement and great accuracy is achieved...

  20. Sharing the 620-790 MHz band allocated to terrestrial television with an audio-bandwidth social service satellite system

    Science.gov (United States)

    Smith, E. K.; Reinhart, E. E.

    1977-01-01

    A study was carried out to identify the optimum uplink and downlink frequencies for audio-bandwidth channels for use by a satellite system distributing social services. The study considered functional-user-need models for five types of social services and identified a general baseline system that is appropriate for most of them. Technical aspects and costs of this system and of the frequency bands that it might use were reviewed, leading to the identification of the 620-790 MHz band as a perferred candidate for both uplink and downlink transmissions for nonmobile applications. The study also led to some ideas as to how to configure the satellite system.

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

    DEFF Research Database (Denmark)

    Pracný, Peter; Bruun, Erik

    2012-01-01

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

  2. Audio Restoration

    Science.gov (United States)

    Esquef, Paulo A. A.

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

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

    Science.gov (United States)

    Aleman-Centeno, Josefina R.

    1983-01-01

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

  4. Two-dimensional block-based reception for differentially encoded OFDM systems : a study on improved reception techniques for digital audio broadcasting systems

    NARCIS (Netherlands)

    Houtum, van W.J.

    2012-01-01

    Digital audio broadcast (DAB), DAB+ and Terrestrial-Digital Multimedia Broadcasting (T-DMB) systems use multi-carrier modulation (MCM). The principle of MCM in the DAB-family is based on orthogonal frequency division multiplexing (OFDM), for which every subcarrier is modulated by p 4 differentially

  5. 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.

  6. 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...

  7. pyAudioAnalysis: An Open-Source Python Library for Audio Signal Analysis.

    Science.gov (United States)

    Giannakopoulos, Theodoros

    2015-01-01

    Audio information plays a rather important role in the increasing digital content that is available today, resulting in a need for methodologies that automatically analyze such content: audio event recognition for home automations and surveillance systems, speech recognition, music information retrieval, multimodal analysis (e.g. audio-visual analysis of online videos for content-based recommendation), etc. This paper presents pyAudioAnalysis, an open-source Python library that provides a wide range of audio analysis procedures including: feature extraction, classification of audio signals, supervised and unsupervised segmentation and content visualization. pyAudioAnalysis is licensed under the Apache License and is available at GitHub (https://github.com/tyiannak/pyAudioAnalysis/). Here we present the theoretical background behind the wide range of the implemented methodologies, along with evaluation metrics for some of the methods. pyAudioAnalysis has been already used in several audio analysis research applications: smart-home functionalities through audio event detection, speech emotion recognition, depression classification based on audio-visual features, music segmentation, multimodal content-based movie recommendation and health applications (e.g. monitoring eating habits). The feedback provided from all these particular audio applications has led to practical enhancement of the library.

  8. 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.

  9. 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 ...

  10. A System for the Semantic Multimodal Analysis of News Audio-Visual Content

    Directory of Open Access Journals (Sweden)

    Michael G. Strintzis

    2010-01-01

    Full Text Available News-related content is nowadays among the most popular types of content for users in everyday applications. Although the generation and distribution of news content has become commonplace, due to the availability of inexpensive media capturing devices and the development of media sharing services targeting both professional and user-generated news content, the automatic analysis and annotation that is required for supporting intelligent search and delivery of this content remains an open issue. In this paper, a complete architecture for knowledge-assisted multimodal analysis of news-related multimedia content is presented, along with its constituent components. The proposed analysis architecture employs state-of-the-art methods for the analysis of each individual modality (visual, audio, text separately and proposes a novel fusion technique based on the particular characteristics of news-related content for the combination of the individual modality analysis results. Experimental results on news broadcast video illustrate the usefulness of the proposed techniques in the automatic generation of semantic annotations.

  11. 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...

  12. 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

  13. 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.

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

    Energy Technology Data Exchange (ETDEWEB)

    NONE

    2000-03-01

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

  15. 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.

  16. Breathe with the Ocean: a System for Relaxation using Audio, Haptic and Visual Stimuli

    NARCIS (Netherlands)

    Dijk, E.O.; Weffers, A.

    2011-01-01

    In this paper we present the “Breathe with the Ocean” system concept, which is a breathing guidance system that aims to help a user relax. It provides an immersive experience where the user is virtually present at an ocean shore. We describe the design and implementation of three embodiments of this

  17. 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...

  18. Design and fuel fabrication processes for the AC-3 mixed-carbide irradiation test

    International Nuclear Information System (INIS)

    Latimer, T.W.; Chidester, K.M.; Stratton, R.W.; Ledergerber, G.; Ingold, F.

    1992-01-01

    The AC-3 test was a cooperative U.S./Swiss irradiation test of 91 wire-wrapped helium-bonded U-20% Pu carbide fuel pins irradiated to 8.3 at % peak burnup in the Fast Flux Test Facility. The test consisted of 25 pins that contained spherepac fuel fabricated by the Paul Scherrer Institute (PSI) and 66 pins that contained pelletized fuel fabricated by the Los Alamos National Laboratory. Design of AC-3 by LANL and PSI was begun in 1981, the fuel pins were fabricated from 1983 to 1985, and the test was irradiated from 1986 to 1988. The principal objective of the AC-3 test was to compare the irradiation performance of mixed-carbide fuel pins that contained either pelletized or sphere-pac fuel at prototypic fluence and burnup levels for a fast breeder reactor

  19. Identification the geothermal system using 1-D audio-magnetotelluric inversion in Lamongan volcano field, East Java, Indonesia

    Science.gov (United States)

    Ilham, N.; Niasari, S. W.

    2018-04-01

    Tiris village, Probolinggo, East Java, is one of geothermal potential areas in Indonesia. This area is located in a valley flank of Mount Lamongan and Argopuro volcanic complex. This research aimed to identify a geothermal system at Tiris area, particularly the fluid pathways. The geothermal potential can be seen from the presence of warm springs with temperature ranging 35-45°C. The warm spring locations are aligned in the same orientation with major fault structure in the area. The fault structure shows dominant northwest-southeast orientation. We used audio-magnetotelluric data in the frequency range of 10 Hz until 92 kHz. The total magnetotelluric sites are 6. From the data analysis, most of the data orientation were 2-D with geo-electrical direction north-south. We used 1-D inversion using Newton algorithm. The 1-D inversion resulted in low resistive anomaly that corresponds to Lamongan lavas. Additionally, the depth of the resistor are different between the area to the west (i.e. 75 m) and to the east (i.e. 25 m). This indicates that there is a fault around the aligned maar (e.g. Ranu Air).

  20. A Real-Time Semiautonomous Audio Panning System for Music Mixing

    Directory of Open Access Journals (Sweden)

    Perez_Gonzalez Enrique

    2010-01-01

    Full Text Available A real-time semiautonomous stereo panning system for music mixing has been implemented. The system uses spectral decomposition, constraint rules, and cross-adaptive algorithms to perform real-time placement of sources in a stereo mix. A subjective evaluation test was devised to evaluate its quality against human panning. It was shown that the automatic panning technique performed better than a nonexpert and showed no significant statistical difference to the performance of a professional mixing engineer.

  1. Irradiation and examination results of the AC-3 mixed-carbide test

    International Nuclear Information System (INIS)

    Mason, R.E.; Hoth, C.W.; Stratton, R.W.; Botta, F.

    1992-01-01

    The AC-3 test was a cooperative Swiss/US irradiation test of mixed-carbide, (U,Pr)C, fuel pins in the Fast Flux Test Facility. The test included 25 Swiss-fabricated sphere-pac-type fuel pins and 66 U.S. fabricated pellet-type fuel pins. The test was designed to operate at prototypical fast reactor conditions to provide a direct comparison of the irradiation performance of the two fuel types. The test design and fuel fabrication processes used for the AC-3 test are presented

  2. Detecting regional lung properties using audio transfer functions of the respiratory system.

    Science.gov (United States)

    Mulligan, K; Adler, A; Goubran, R

    2009-01-01

    In this study, a novel instrument has been developed for measuring changes in the distribution of lung fluid the respiratory system. The instrument consists of a speaker that inputs a 0-4kHz White Gaussian Noise (WGN) signal into a patient's mouth and an array of 4 electronic stethoscopes, linked via a fully adjustable harness, used to recover signals on the chest surface. The software system for processing the data utilizes the principles of adaptive filtering in order to obtain a transfer function that represents the input-output relationship for the signal as the volume of fluid in the lungs is varied. A chest phantom model was constructed to simulate the behavior of fluid related diseases within the lungs through the injection of varying volumes of water. Tests from the phantom model were compared to healthy subjects. Results show the instrument can obtain similar transfer functions and sound propagation delays between both human and phantom chests.

  3. Speech Acquisition and Automatic Speech Recognition for Integrated Spacesuit Audio Systems

    Science.gov (United States)

    Huang, Yiteng; Chen, Jingdong; Chen, Shaoyan

    2010-01-01

    A voice-command human-machine interface system has been developed for spacesuit extravehicular activity (EVA) missions. A multichannel acoustic signal processing method has been created for distant speech acquisition in noisy and reverberant environments. This technology reduces noise by exploiting differences in the statistical nature of signal (i.e., speech) and noise that exists in the spatial and temporal domains. As a result, the automatic speech recognition (ASR) accuracy can be improved to the level at which crewmembers would find the speech interface useful. The developed speech human/machine interface will enable both crewmember usability and operational efficiency. It can enjoy a fast rate of data/text entry, small overall size, and can be lightweight. In addition, this design will free the hands and eyes of a suited crewmember. The system components and steps include beam forming/multi-channel noise reduction, single-channel noise reduction, speech feature extraction, feature transformation and normalization, feature compression, model adaption, ASR HMM (Hidden Markov Model) training, and ASR decoding. A state-of-the-art phoneme recognizer can obtain an accuracy rate of 65 percent when the training and testing data are free of noise. When it is used in spacesuits, the rate drops to about 33 percent. With the developed microphone array speech-processing technologies, the performance is improved and the phoneme recognition accuracy rate rises to 44 percent. The recognizer can be further improved by combining the microphone array and HMM model adaptation techniques and using speech samples collected from inside spacesuits. In addition, arithmetic complexity models for the major HMMbased ASR components were developed. They can help real-time ASR system designers select proper tasks when in the face of constraints in computational resources.

  4. 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.

  5. 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....

  6. 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...

  7. 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

  8. Service provider perceptions of transitioning from audio to video capability in a telehealth system: a qualitative evaluation

    OpenAIRE

    Clay-Williams, Robyn; Baysari, Melissa; Taylor, Natalie; Zalitis, Dianne; Georgiou, Andrew; Robinson, Maureen; Braithwaite, Jeffrey; Westbrook, Johanna

    2017-01-01

    Background Telephone consultation and triage services are increasingly being used to deliver health advice. Availability of high speed internet services in remote areas allows healthcare providers to move from telephone to video telehealth services. Current approaches for assessing video services have limitations. This study aimed to identify the challenges for service providers associated with transitioning from audio to video technology. Methods Using a mixed-method, qualitative approach, w...

  9. 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...

  10. 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...

  11. 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.

  12. Tomato leaf curl Kerala virus (ToLCKeV AC3 protein forms a higher order oligomer and enhances ATPase activity of replication initiator protein (Rep/AC1

    Directory of Open Access Journals (Sweden)

    Mukherjee Sunil K

    2010-06-01

    Full Text Available Abstract Background Geminiviruses are emerging plant viruses that infect a wide variety of vegetable crops, ornamental plants and cereal crops. They undergo recombination during co-infections by different species of geminiviruses and give rise to more virulent species. Antiviral strategies targeting a broad range of viruses necessitate a detailed understanding of the basic biology of the viruses. ToLCKeV, a virus prevalent in the tomato crop of Kerala state of India and a member of genus Begomovirus has been used as a model system in this study. Results AC3 is a geminiviral protein conserved across all the begomoviral species and is postulated to enhance viral DNA replication. In this work we have successfully expressed and purified the AC3 fusion proteins from E. coli. We demonstrated the higher order oligomerization of AC3 using sucrose gradient ultra-centrifugation and gel-filtration experiments. In addition we also established that ToLCKeV AC3 protein interacted with cognate AC1 protein and enhanced the AC1-mediated ATPase activity in vitro. Conclusions Highly hydrophobic viral protein AC3 can be purified as a fusion protein with either MBP or GST. The purification method of AC3 protein improves scope for the biochemical characterization of the viral protein. The enhancement of AC1-mediated ATPase activity might lead to increased viral DNA replication.

  13. 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.

  14. 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.

  15. 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.

  16. 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.

  17. 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.

  18. 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.

  19. DAFX Digital Audio Effects

    CERN Document Server

    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

  20. Effect of In-Vehicle Audio Warning System on Driver’s Speed Control Performance in Transition Zones from Rural Areas to Urban Areas

    Directory of Open Access Journals (Sweden)

    Xuedong Yan

    2016-06-01

    Full Text Available Speeding is a major contributing factor to traffic crashes and frequently happens in areas where there is a mutation in speed limits, such as the transition zones that connect urban areas from rural areas. The purpose of this study is to investigate the effects of an in-vehicle audio warning system and lit speed limit sign on preventing drivers’ speeding behavior in transition zones. A high-fidelity driving simulator was used to establish a roadway network with the transition zone. A total of 41 participants were recruited for this experiment, and the driving speed performance data were collected from the simulator. The experimental results display that the implementation of the audio warning system could significantly reduce drivers’ operating speed before they entered the urban area, while the lit speed limit sign had a minimal effect on improving the drivers’ speed control performance. Without consideration of different types of speed limit signs, it is found that male drivers generally had a higher operating speed both upstream and in the transition zones and have a larger maximum deceleration for speed reduction than female drivers. Moreover, the drivers who had medium-level driving experience had the higher operating speed and were more likely to have speeding behaviors in the transition zones than those who had low-level and high-level driving experience in the transition zones.

  1. 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...

  2. 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…

  3. 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.

  4. 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

  5. Wavelet-based audio embedding and audio/video compression

    Science.gov (United States)

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

    2001-12-01

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

  6. 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.

  7. 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.

  8. 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

  9. 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...

  10. 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

  11. Assessing the allelotypic effect of two aminocyclopropane carboxylic acid synthase-encoding genes MdACS1 and MdACS3a on fruit ethylene production and softening in Malus

    Science.gov (United States)

    Dougherty, Laura; Zhu, Yuandi; Xu, Kenong

    2016-01-01

    Phytohormone ethylene largely determines apple fruit shelf life and storability. Previous studies demonstrated that MdACS1 and MdACS3a, which encode 1-aminocyclopropane-1-carboxylic acid synthases (ACS), are crucial in apple fruit ethylene production. MdACS1 is well-known to be intimately involved in the climacteric ethylene burst in fruit ripening, while MdACS3a has been regarded a main regulator for ethylene production transition from system 1 (during fruit development) to system 2 (during fruit ripening). However, MdACS3a was also shown to have limited roles in initiating the ripening process lately. To better assess their roles, fruit ethylene production and softening were evaluated at five time points during a 20-day post-harvest period in 97 Malus accessions and in 34 progeny from 2 controlled crosses. Allelotyping was accomplished using an existing marker (ACS1) for MdACS1 and two markers (CAPS866 and CAPS870) developed here to specifically detect the two null alleles (ACS3a-G289V and Mdacs3a) of MdACS3a. In total, 952 Malus accessions were allelotyped with the three markers. The major findings included: The effect of MdACS1 was significant on fruit ethylene production and softening while that of MdACS3a was less detectable; allele MdACS1–2 was significantly associated with low ethylene and slow softening; under the same background of the MdACS1 allelotypes, null allele Mdacs3a (not ACS3a-G289V) could confer a significant delay of ethylene peak; alleles MdACS1–2 and Mdacs3a (excluding ACS3a-G289V) were highly enriched in M. domestica and M. hybrid when compared with those in M. sieversii. These findings are of practical implications in developing apples of low and delayed ethylene profiles by utilizing the beneficial alleles MdACS1-2 and Mdacs3a. PMID:27231553

  12. 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.

  13. 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.

  14. 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.

  15. 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...

  16. 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)

  17. 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...

  18. 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...

  19. 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.

  20. Superfund TIO videos: Set C. Ground water: Ground water containment and removal systems. Part 7. Audio-Visual

    International Nuclear Information System (INIS)

    1990-01-01

    The videotape analyzes containment and control systems that are used to obtain hydraulic control and discusses selection of preferred control measures that are based on site-specific criteria and general performance information. Advantages and disadvantages of slurry walls, subsurface drains, well systems, sheet pilings, and grout curtains are also covered

  1. The Evaluation of Science Learning Program, Technology and Society Application of Audio Bio Harmonic System with Solar Energy to Improve Crop Productivity

    Directory of Open Access Journals (Sweden)

    D. Rosana

    2017-04-01

    Full Text Available One of the greatest challenges in science learning is how to integrate a wide range of basic scientific concepts of physics, chemistry, and biology into an integrated learning material. Research-based teaching material in this area is still very poor and does not much involve students of science education in its implementation as part of the learning program science technology and society (STS. The purpose of this study is to get the result of evaluation of the teaching and learning of STS in the form of public service in Kulon Progo, Yogyakarta. The program to improve crop productivity through the application of Audio Bio Harmonic System (ABHS with solar energy have been selected for utilizing the natural animal sounds to open stomata of the leaves conducted during foliar fertilization, making it suitable for integrated science lessons. Component of evaluation model used is Stufflebeam model evaluation (CIPP. CIPP evaluation in these activities resulted in two aspects: The first aspect was improving the skills of students and farmers in using ABHS, and these two aspects, namely food crop productivity; (1 cayenne increased 76.4%, (2 increased red onions (56.3% and (3 of maize increased by 67.8%. Besides, it was also the effect of the application of ABHS on the rate of plant growth. The outcome of this study is the STS teaching materials and appropriate technology of ABHS with solar energy.

  2. 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. 

  3. Spatial Analysis and Synthesis of Car Audio System and Car Cabin Acoustics with a Compact Microphone Array

    DEFF Research Database (Denmark)

    Sakari, Tervo; Pätynen, Jukka; Kaplanis, Neofytos

    2015-01-01

    This research proposes a spatial sound analysis and synthesis approach for automobile sound systems, where the acquisition of the measurement data is much faster than with the Binaural Car Scanning method. This approach avoids the problems that are typically found with binaural reproduction...

  4. 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

  5. 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)

  6. 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.

  7. 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...

  8. 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.

  9. 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....

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

    Science.gov (United States)

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

    1996-05-01

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

  11. 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....

  12. 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...

  13. 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,

  14. 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

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

    CERN Document Server

    Lerch, Alexander

    2012-01-01

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

  16. Sharing Annotated Audio Recordings of Clinic Visits With Patients-Development of the Open Recording Automated Logging System (ORALS): Study Protocol.

    Science.gov (United States)

    Barr, Paul J; Dannenberg, Michelle D; Ganoe, Craig H; Haslett, William; Faill, Rebecca; Hassanpour, Saeed; Das, Amar; Arend, Roger; Masel, Meredith C; Piper, Sheryl; Reicher, Haley; Ryan, James; Elwyn, Glyn

    2017-07-06

    Providing patients with recordings of their clinic visits enhances patient and family engagement, yet few organizations routinely offer recordings. Challenges exist for organizations and patients, including data safety and navigating lengthy recordings. A secure system that allows patients to easily navigate recordings may be a solution. The aim of this project is to develop and test an interoperable system to facilitate routine recording, the Open Recording Automated Logging System (ORALS), with the aim of increasing patient and family engagement. ORALS will consist of (1) technically proficient software using automated machine learning technology to enable accurate and automatic tagging of in-clinic audio recordings (tagging involves identifying elements of the clinic visit most important to patients [eg, treatment plan] on the recording) and (2) a secure, easy-to-use Web interface enabling the upload and accurate linkage of recordings to patients, which can be accessed at home. We will use a mixed methods approach to develop and formatively test ORALS in 4 iterative stages: case study of pioneer clinics where recordings are currently offered to patients, ORALS design and user experience testing, ORALS software and user interface development, and rapid cycle testing of ORALS in a primary care clinic, assessing impact on patient and family engagement. Dartmouth's Informatics Collaboratory for Design, Development and Dissemination team, patients, patient partners, caregivers, and clinicians will assist in developing ORALS. We will implement a publication plan that includes a final project report and articles for peer-reviewed journals. In addition to this work, we will regularly report on our progress using popular relevant Tweet chats and online using our website, www.openrecordings.org. We will disseminate our work at relevant conferences (eg, Academy Health, Health Datapalooza, and the Institute for Healthcare Improvement Quality Forums). Finally, Iora Health, a

  17. 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.

  18. Design guidelines for audio presentation of graphs and tables

    OpenAIRE

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

    2003-01-01

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

  19. 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

  20. 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....

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

    Science.gov (United States)

    George, Rohini

    function could be approximated to a normal distribution function. A statistical analysis was also performed to investigate if a patient's physical, tumor or general characteristics played a role in identifying whether he/she responded positively to the coaching type---signified by a reduction in the variability of respiratory motion. The analysis demonstrated that, although there were some characteristics like disease type and dose per fraction that were significant with respect to time-independent analysis, there were no significant time trends observed for the inter-session or intra-session analysis. Based on patient feedback with the existing audio-visual biofeedback system used for the study and research performed on other feedback systems, an improved audio-visual biofeedback system was designed. It is hoped the widespread clinical implementation of audio-visual biofeedback for radiotherapy will improve the accuracy of lung cancer radiotherapy.

  2. 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...

  3. Hydrothermal system beneath the crater of Tarumai volcano, Japan : 3-D resistivity structure revealed using audio-magnetotellurics and induction vector

    OpenAIRE

    Yamaya, Yusuke; Mogi, Toru; Hashimoto, Takeshi; Ichihara, Hiroshi

    2009-01-01

    Audio-magnetotelluric (AMT) measurements were recorded in the crater area of Tarumai volcano, northeastern Japan. This survey brought the specific structures beneath the lava dome of Tarumai volcano, enabling us to interpret the relationship between the subsurface structure and fumarolic activity in the vicinity of a lava dome. Three-dimensional resistivity modeling was performed to achieve this purpose. The measured induction vectors pointed toward the center of the dome, implying the topogr...

  4. 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

  5. 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.

  6. Engaging Students with Audio Feedback

    Science.gov (United States)

    Cann, Alan

    2014-01-01

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

  7. 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...

  8. Radioactive Decay: Audio Data Collection

    Science.gov (United States)

    Struthers, Allan

    2009-01-01

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

  9. 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

  10. 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…

  11. 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.

  12. 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.

  13. 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.

  14. Design of a WAV audio player based on K20

    Directory of Open Access Journals (Sweden)

    Xu Yu

    2016-01-01

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

  15. Minimizing Crosstalk in Self Oscillating Switch Mode Audio Power Amplifiers

    DEFF Research Database (Denmark)

    Knott, Arnold; Ploug, Rasmus Overgaard

    2012-01-01

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

  16. 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.

  17. 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.

  18. 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...

  19. Modified BTC Algorithm for Audio Signal Coding

    Directory of Open Access Journals (Sweden)

    TOMIC, S.

    2016-11-01

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

  20. 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...

  1. Audio Arduino - an ALSA (Advanced Linux Sound Architecture) audio driver for FTDI-based Arduinos

    DEFF Research Database (Denmark)

    Dimitrov, Smilen; Serafin, Stefania

    2011-01-01

    be considered to be a system, that encompasses design decisions on both hardware and software levels - that also demand a certain understanding of the architecture of the target PC operating system. This project outlines how an Arduino Duemillanove board (containing a USB interface chip, manufactured by Future...... Technology Devices International Ltd [FTDI] company) can be demonstrated to behave as a full-duplex, mono, 8-bit 44.1 kHz soundcard, through an implementation of: a PC audio driver for ALSA (Advanced Linux Sound Architecture); a matching program for the Arduino's ATmega microcontroller - and nothing more...

  2. 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)

  3. 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

  4. Design And Construction Of 300W Audio Power Amplifier For Classroom

    Directory of Open Access Journals (Sweden)

    Shune Lei Aung

    2015-07-01

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

  5. Modular Sensor Environment : Audio Visual Industry Monitoring Applications

    OpenAIRE

    Guillot, Calvin

    2017-01-01

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

  6. 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

  7. AKTIVITAS SEKUNDER AUDIO UNTUK MENJAGA KEWASPADAAN PENGEMUDI MOBIL INDONESIA

    Directory of Open Access Journals (Sweden)

    Iftikar Zahedi Sutalaksana

    2013-03-01

    the awake, alert, and able to process all the stimulus well. The results of this study generate some form of audio response test that is integrated with the system drive in the car. Sound source is played with constant intensity between 80-85 dB. The sound will stop if the driver to respond to the sound stimulus. Response test is designed to be capable of monitoring the driver's level of alertness while driving. Its application is expected to help reduce the rate of traffic accidents in Indonesia. Keywords: driving, secondary activities, audio, alertness, response test

  8. 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.

  9. 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.

  10. 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

  11. 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...

  12. Turkish Music Genre Classification using Audio and Lyrics Features

    Directory of Open Access Journals (Sweden)

    Önder ÇOBAN

    2017-05-01

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

  13. The relationship between basic audio quality and overall listening experience.

    Science.gov (United States)

    Schoeffler, Michael; Herre, Jürgen

    2016-09-01

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

  14. Detecting double compression of audio signal

    Science.gov (United States)

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

    2010-01-01

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

  15. 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.

  16. 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.

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

    Directory of Open Access Journals (Sweden)

    Md. Rabiul Islam

    2014-01-01

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

  18. The Complement C3a-C3aR Axis Promotes Development of Thoracic Aortic Dissection via Regulation of MMP2 Expression.

    Science.gov (United States)

    Ren, Weihong; Liu, Yan; Wang, Xuerui; Piao, Chunmei; Ma, Youcai; Qiu, Shulan; Jia, Lixin; Chen, Boya; Wang, Yuan; Jiang, Wenjian; Zheng, Shuai; Liu, Chang; Dai, Nan; Lan, Feng; Zhang, Hongjia; Song, Wen-Chao; Du, Jie

    2018-03-01

    Thoracic aortic dissection (TAD), once ruptured, is devastating to patients, and no effective pharmaceutical therapy is available. Anaphylatoxins released by complement activation are involved in a variety of diseases. However, the role of the complement system in TAD is unknown. We found that plasma levels of C3a, C4a, and C5a were significantly increased in patients with TAD. Elevated circulating C3a levels were also detected in the developmental process of mouse TAD, which was induced by β-aminopropionitrile monofumarate (BAPN) treatment, with enhanced expression of C1q and properdin in mouse dissected aortas. These findings indicated activation of classical and alternative complement pathways. Further, expression of C3aR was obviously increased in smooth muscle cells of human and mouse dissected aortas, and knockout of C3aR notably inhibited BAPN-induced formation and rupture of TAD in mice. C3aR antagonist administered pre- and post-BAPN treatment attenuated the development of TAD. We found that C3aR knockout decreased matrix metalloproteinase 2 (MMP2) expression in BAPN-treated mice. Additionally, recombinant C3a stimulation enhanced MMP2 expression and activation in smooth muscle cells that were subjected to mechanical stretch. Finally, we generated MMP2-knockdown mice by in vivo MMP2 short hairpin RNA delivery using recombinant adeno-associated virus and found that MMP2 deficiency significantly reduced the formation of TAD. Therefore, our study suggests that the C3a - C3aR axis contributes to the development of TAD via regulation of MMP2 expression. Targeting the C3a-C3aR axis may represent a strategy for inhibiting the formation of TAD. Copyright © 2018 by The American Association of Immunologists, Inc.

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

    DEFF Research Database (Denmark)

    Turchet, Luca; Serafin, Stefania; Dimitrov, Smilen

    2010-01-01

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

  20. 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...

  1. 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...

  2. 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.

  3. 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.

  4. An Audio Architecture Integrating Sound and Live Voice for Virtual Environments

    National Research Council Canada - National Science Library

    Krebs, Eric

    2002-01-01

    The purpose behind this thesis was to design and implement audio system architecture, both in hardware and in software, for use in virtual environments The hardware and software design requirements...

  5. 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...

  6. 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.

  7. 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...

  8. 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.

  9. 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%..

  10. Audio production principles practical studio applications

    CERN Document Server

    Elmosnino, Stephane

    2018-01-01

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

  11. Adaptive DCTNet for Audio Signal Classification

    OpenAIRE

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

    2016-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Rita Kumalasari

    2017-05-01

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

  13. 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

  14. 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.

  15. Active Learning for Automatic Audio Processing of Unwritten Languages (ALAPUL)

    Science.gov (United States)

    2016-07-01

    AFRL-RH-WP-TR-2016-0074 ACTIVE LEARNING FOR AUTOMATIC AUDIO PROCESSING OF UNWRITTEN LANGUAGES (ALAPUL) Dimitra Vergyri Andreas Kathol Wen Wang...FA8650-15-C-9101 5b. GRANT NUMBER 5c. PROGRAM ELEMENT NUMBER 6. AUTHOR(S) *Dimitra Vergyri; Andreas Kathol; Wen Wang; Chris Bartels; Julian VanHout...feature transform through deep auto-encoders for better phone recognition performance. We target iterative learning to improve the system through

  16. Virtual environment display for a 3D audio room simulation

    Science.gov (United States)

    Chapin, William L.; Foster, Scott

    1992-06-01

    Recent developments in virtual 3D audio and synthetic aural environments have produced a complex acoustical room simulation. The acoustical simulation models a room with walls, ceiling, and floor of selected sound reflecting/absorbing characteristics and unlimited independent localizable sound sources. This non-visual acoustic simulation, implemented with 4 audio ConvolvotronsTM by Crystal River Engineering and coupled to the listener with a Poihemus IsotrakTM, tracking the listener's head position and orientation, and stereo headphones returning binaural sound, is quite compelling to most listeners with eyes closed. This immersive effect should be reinforced when properly integrated into a full, multi-sensory virtual environment presentation. This paper discusses the design of an interactive, visual virtual environment, complementing the acoustic model and specified to: 1) allow the listener to freely move about the space, a room of manipulable size, shape, and audio character, while interactively relocating the sound sources; 2) reinforce the listener's feeling of telepresence into the acoustical environment with visual and proprioceptive sensations; 3) enhance the audio with the graphic and interactive components, rather than overwhelm or reduce it; and 4) serve as a research testbed and technology transfer demonstration. The hardware/software design of two demonstration systems, one installed and one portable, are discussed through the development of four iterative configurations. The installed system implements a head-coupled, wide-angle, stereo-optic tracker/viewer and multi-computer simulation control. The portable demonstration system implements a head-mounted wide-angle, stereo-optic display, separate head and pointer electro-magnetic position trackers, a heterogeneous parallel graphics processing system, and object oriented C++ program code.

  17. 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.

  18. Video equipment of tele dosimetry and audio

    International Nuclear Information System (INIS)

    Ojeda R, M.A.; Padilla C, I.

    2007-01-01

    To develop a work in an area with high radiation, it requires of a detailed knowledge of the surroundings work, a communication and effective vision, a near dosimetric control. In a work where the spaces variables and reduced accesses exist, noise that hinders the communication, defendant operative condition, radiation field and taking of decision, it is necessary to have tools that allow a total control of the environment to make opportune and effective decisions, there where the task is developed. Under this elementary concept, it was developed in the Laguna Verde Central a project that it allowed a mechanism, interactive of control in spaces complex; to see, to hear, to speak, to measure. This concept takes to the creation of an equipped system with closed circuit of television, wireless communication systems, tele dosimetry wireless systems, VHS and DVD recording equipment, uninterrupted energy units. The system requires of an electric power socket, and the installation of two cables by CCTV camera. The system is mobilized by a person. He puts on in operation in 5 minutes using a verification list. The concept was developed in the project denominated VETA-1, (Video Equipment of Tele dosimetry and Audio). It is objective of this work to present before the society the development of the VETA-1 tool that conclude in their first prototype in May of the present year. The VETA-1 project arises by a necessity of optimizing dose, it is an ALARA tool, with a countless applications, like it was proven in the 12 recharge stop of the Unit 1. The VETA-1 project integrate a recording system, with the primary end of analyzing in the place where the task is developed the details for an effective and opportune decision, but the resulting information is of utility for the personnel's training and the planning of future works. The VETA-1 system is an ALARA tool of quick response control. (Author)

  19. 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.

  20. 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…

  1. Objective Audio Quality Assessment Based on Spectro-Temporal Modulation Analysis

    OpenAIRE

    Guo, Ziyuan

    2011-01-01

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

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

    OpenAIRE

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

    2016-01-01

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

  3. Lecture Hall and Learning Design: A Survey of Variables, Parameters, Criteria and Interrelationships for Audio-Visual Presentation Systems and Audience Reception.

    Science.gov (United States)

    Justin, J. Karl

    Variables and parameters affecting architectural planning and audiovisual systems selection for lecture halls and other learning spaces are surveyed. Interrelationships of factors are discussed, including--(1) design requirements for modern educational techniques as differentiated from cinema, theater or auditorium design, (2) general hall…

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

    DEFF Research Database (Denmark)

    Poulsen, Søren; Andersen, Michael Andreas E.

    2004-01-01

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

  5. 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.

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

    Science.gov (United States)

    Hage, Steffen R; Nieder, Andreas

    2015-05-06

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

  7. 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

  8. 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....

  9. Modified DCTNet for audio signals classification

    Science.gov (United States)

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

    2016-10-01

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

  10. 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.

  11. 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...

  12. audio-ultrasonic waves by argon gas discharge

    International Nuclear Information System (INIS)

    Ragheb, M.S.

    2010-01-01

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

  13. Automatic summarization of soccer highlights using audio-visual descriptors.

    Science.gov (United States)

    Raventós, A; Quijada, R; Torres, Luis; Tarrés, Francesc

    2015-01-01

    Automatic summarization generation of sports video content has been object of great interest for many years. Although semantic descriptions techniques have been proposed, many of the approaches still rely on low-level video descriptors that render quite limited results due to the complexity of the problem and to the low capability of the descriptors to represent semantic content. In this paper, a new approach for automatic highlights summarization generation of soccer videos using audio-visual descriptors is presented. The approach is based on the segmentation of the video sequence into shots that will be further analyzed to determine its relevance and interest. Of special interest in the approach is the use of the audio information that provides additional robustness to the overall performance of the summarization system. For every video shot a set of low and mid level audio-visual descriptors are computed and lately adequately combined in order to obtain different relevance measures based on empirical knowledge rules. The final summary is generated by selecting those shots with highest interest according to the specifications of the user and the results of relevance measures. A variety of results are presented with real soccer video sequences that prove the validity of the approach.

  14. 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.

  15. 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.

  16. 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.

  17. The complete guide to high-end audio

    CERN Document Server

    Harley, Robert

    2015-01-01

    An updated edition of what many consider the "bible of high-end audio"   In this newly revised and updated fifth edition, Robert Harley, editor in chief of the Absolute Sound magazine, tells you everything you need to know about buying and enjoying high-quality hi-fi. With this book, discover how to get the best sound for your money, how to identify the weak links in your system and upgrade where it will do the most good, how to set up and tweak your system for maximum performance, and how to become a more perceptive and appreciative listener. Just a few of the secrets you will learn cover hi

  18. 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.

  19. 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.

  20. Nonspeech audio in user interfaces for TV

    NARCIS (Netherlands)

    Sluis, van de Richard; Eggen, J.H.; Rypkema, J.A.

    1997-01-01

    This study explores the end-user benefits of using nonspeech audio in television user interfaces. A prototype of an Electronic Programme Guide (EPG) served as a carrier for the research. One of the features of this EPG is the possibility to search for TV programmes in a category-based way. The EPG

  1. 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.

  2. Spatial audio quality perception (part 2)

    DEFF Research Database (Denmark)

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

    2015-01-01

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

  3. Study of audio speakers containing ferrofluid

    Energy Technology Data Exchange (ETDEWEB)

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

    2008-05-21

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

  4. 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…

  5. 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.

  6. Agency Video, Audio and Imagery Library

    Science.gov (United States)

    Grubbs, Rodney

    2015-01-01

    The purpose of this presentation was to inform the ISS International Partners of the new NASA Agency Video, Audio and Imagery Library (AVAIL) website. AVAIL is a new resource for the public to search for and download NASA-related imagery, and is not intended to replace the current process by which the International Partners receive their Space Station imagery products.

  7. A listening test system for automotive audio

    DEFF Research Database (Denmark)

    Christensen, Flemming; Martin, Geoff; Minnaar, Pauli

    2005-01-01

    A selection procedure was devised in order to select listeners for experiments in which their main task will be to judge multi-channel reproduced sound. 91 participants filled in a web-based questionnaire. 78 of them took part in an assessment of their hearing thresholds, their spatial hearing......, and their verbal production abilities. The listeners displayed large individual differences in their performance. 40 subjects were selected based on the test results. The self-assessed listening habits and experience in the web questionnaire could not predict the results of the selection procedure. Further......, the hearing thresholds did not correlate with the spatial-hearing test. This leads to the conclusion that task-specific performance tests might be the preferable means of selecting a listening panel....

  8. 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...

  9. 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

  10. 47 CFR 10.520 - Common audio attention signal.

    Science.gov (United States)

    2010-10-01

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

  11. 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...

  12. 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

  13. 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...

  14. 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

  15. Using Audio-Derived Affective Offset to Enhance TV Recommendation

    DEFF Research Database (Denmark)

    Shepstone, Sven Ewan; Tan, Zheng-Hua; Jensen, Søren Holdt

    2014-01-01

    . First a user's mood profile is determined using 12-class audio-based emotion classifications . An initial TV content item is then displayed to the user based on the extracted mood profile. The user has the option to either accept the recommendation, or to critique the item once or several times......, by navigating the emotion space to request an alternative match. The final match is then compared to the initial match, in terms of the difference in the items' affective parameterization . This offset is then utilized in future recommendation sessions. The system was evaluated by eliciting three different...

  16. Multilevel tracking power supply for switch-mode audio power amplifiers

    DEFF Research Database (Denmark)

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

    2018-01-01

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

  17. New musical organology : the audio-games

    OpenAIRE

    Zénouda , Hervé

    2012-01-01

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

  18. Audio Networking in the Music Industry

    OpenAIRE

    Glebs Kuzmics; Maaruf Ali

    2018-01-01

    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...

  19. Digitisation of the CERN Audio Archives

    CERN Multimedia

    Maximilien Brice

    2006-01-01

    Since the creation of CERN in 1954 until mid 1980s, the audiovisual service has recorded hundreds of hours of moments of life at CERN on audio tapes. These moments range from inaugurations of new facilities to VIP speeches and general interest cultural seminars The preservation process started in June 2005 On these pictures, we see Waltraud Hug working on an open-reel tape.

  20. Securing Digital Audio using Complex Quadratic Map

    Science.gov (United States)

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

    2018-03-01

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

  1. Theory and Application of Audio-Based Assessment of Cough

    Directory of Open Access Journals (Sweden)

    Yan Shi

    2018-01-01

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

  2. 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.

  3. 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.

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

    Directory of Open Access Journals (Sweden)

    Butko Taras

    2011-01-01

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

  5. 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.

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

    NARCIS (Netherlands)

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

    2008-01-01

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

  7. Music information retrieval in compressed audio files: a survey

    Science.gov (United States)

    Zampoglou, Markos; Malamos, Athanasios G.

    2014-07-01

    In this paper, we present an organized survey of the existing literature on music information retrieval systems in which descriptor features are extracted directly from the compressed audio files, without prior decompression to pulse-code modulation format. Avoiding the decompression step and utilizing the readily available compressed-domain information can significantly lighten the computational cost of a music information retrieval system, allowing application to large-scale music databases. We identify a number of systems relying on compressed-domain information and form a systematic classification of the features they extract, the retrieval tasks they tackle and the degree in which they achieve an actual increase in the overall speed-as well as any resulting loss in accuracy. Finally, we discuss recent developments in the field, and the potential research directions they open toward ultra-fast, scalable systems.

  8. 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...

  9. 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...

  10. 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.

  11. 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.

  12. 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.

  13. [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.

  14. 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.

  15. Automatic processing of CERN video, audio and photo archives

    Energy Technology Data Exchange (ETDEWEB)

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

    2008-07-15

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

  16. Extraction of Information of Audio-Visual Contents

    Directory of Open Access Journals (Sweden)

    Carlos Aguilar

    2011-10-01

    Full Text Available In this article we show how it is possible to use Channel Theory (Barwise and Seligman, 1997 for modeling the process of information extraction realized by audiences of audio-visual contents. To do this, we rely on the concepts pro- posed by Channel Theory and, especially, its treatment of representational systems. We then show how the information that an agent is capable of extracting from the content depends on the number of channels he is able to establish between the content and the set of classifications he is able to discriminate. The agent can endeavor the extraction of information through these channels from the totality of content; however, we discuss the advantages of extracting from its constituents in order to obtain a greater number of informational items that represent it. After showing how the extraction process is endeavored for each channel, we propose a method of representation of all the informative values an agent can obtain from a content using a matrix constituted by the channels the agent is able to establish on the content (source classifications, and the ones he can understand as individual (destination classifications. We finally show how this representation allows reflecting the evolution of the informative items through the evolution of audio-visual content.

  17. Audio-tactile integration and the influence of musical training.

    Directory of Open Access Journals (Sweden)

    Anja Kuchenbuch

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

  18. Automatic processing of CERN video, audio and photo archives

    International Nuclear Information System (INIS)

    Kwiatek, M

    2008-01-01

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

  19. Audio-tactile integration and the influence of musical training.

    Science.gov (United States)

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

    2014-01-01

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

  20. 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....

  1. 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.

  2. 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.

  3. Predistortion of a Bidirectional Cuk Audio Amplifier

    DEFF Research Database (Denmark)

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

    2014-01-01

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

  4. Mixing audio concepts, practices and tools

    CERN Document Server

    Izhaki, Roey

    2013-01-01

    Your mix can make or break a record, and mixing is an essential catalyst for a record deal. Professional engineers with exceptional mixing skills can earn vast amounts of money and find that they are in demand by the biggest acts. To develop such skills, you need to master both the art and science of mixing. The new edition of this bestselling book offers all you need to know and put into practice in order to improve your mixes. Covering the entire process --from fundamental concepts to advanced techniques -- and offering a multitude of audio samples, tips and tricks, this boo

  5. Calibration of an audio frequency noise generator

    DEFF Research Database (Denmark)

    Diamond, Joseph M.

    1966-01-01

    a noise bandwidth Bn = π/2 × (3dB bandwidth). To apply this method to low audio frequencies, the noise bandwidth of the low Q parallel resonant circuit has been found, including the effects of both series and parallel damping. The method has been used to calibrate a General Radio 1390-B noise generator...... it is used for measurement purposes. The spectral density of a noise source may be found by measuring its rms output over a known noise bandwidth. Such a bandwidth may be provided by a passive filter using accurately known elements. For example, the parallel resonant circuit with purely parallel damping has...

  6. 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...

  7. Cortical Integration of Audio-Visual Information

    Science.gov (United States)

    Vander Wyk, Brent C.; Ramsay, Gordon J.; Hudac, Caitlin M.; Jones, Warren; Lin, David; Klin, Ami; Lee, Su Mei; Pelphrey, Kevin A.

    2013-01-01

    We investigated the neural basis of audio-visual processing in speech and non-speech stimuli. Physically identical auditory stimuli (speech and sinusoidal tones) and visual stimuli (animated circles and ellipses) were used in this fMRI experiment. Relative to unimodal stimuli, each of the multimodal conjunctions showed increased activation in largely non-overlapping areas. The conjunction of Ellipse and Speech, which most resembles naturalistic audiovisual speech, showed higher activation in the right inferior frontal gyrus, fusiform gyri, left posterior superior temporal sulcus, and lateral occipital cortex. The conjunction of Circle and Tone, an arbitrary audio-visual pairing with no speech association, activated middle temporal gyri and lateral occipital cortex. The conjunction of Circle and Speech showed activation in lateral occipital cortex, and the conjunction of Ellipse and Tone did not show increased activation relative to unimodal stimuli. Further analysis revealed that middle temporal regions, although identified as multimodal only in the Circle-Tone condition, were more strongly active to Ellipse-Speech or Circle-Speech, but regions that were identified as multimodal for Ellipse-Speech were always strongest for Ellipse-Speech. Our results suggest that combinations of auditory and visual stimuli may together be processed by different cortical networks, depending on the extent to which speech or non-speech percepts are evoked. PMID:20709442

  8. 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.

  9. 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.

  10. 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....

  11. 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.

  12. 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...

  13. Creating Accessible Science Museums with User-Activated Environmental Audio Beacons (Ping!)

    Science.gov (United States)

    Landau, Steven; Wiener, William; Naghshineh, Koorosh; Giusti, Ellen

    2005-01-01

    In 2003, Touch Graphics Company carried out research on a new invention that promises to improve accessibility to science museums for visitors who are visually impaired. The system, nicknamed Ping!, allows users to navigate an exhibit area, listen to audio descriptions, and interact with exhibits using a cell phone-based interface. The system…

  14. Online Dissection Audio-Visual Resources for Human Anatomy: Undergraduate Medical Students' Usage and Learning Outcomes

    Science.gov (United States)

    Choi-Lundberg, Derek L.; Cuellar, William A.; Williams, Anne-Marie M.

    2016-01-01

    In an attempt to improve undergraduate medical student preparation for and learning from dissection sessions, dissection audio-visual resources (DAVR) were developed. Data from e-learning management systems indicated DAVR were accessed by 28% ± 10 (mean ± SD for nine DAVR across three years) of students prior to the corresponding dissection…

  15. 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…

  16. 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

  17. acceleration observed in an audio air gas discharge

    International Nuclear Information System (INIS)

    Ragheb, M.S.

    2010-01-01

    an audio air gas discharge enclosed in a pyrex glass of 34 mm diameter and 25 cm long , lead to trace the occurrence of an unusual phenomenon. injected relative huge light spots of intense brightness, distributed regularly on the contour and in the center of one of the discharge electrodes, are observed. very high heat is pronounced on both electrodes, while, one of them is higher than the other it attains 660 degree C in 3-4 minutes. series of photographs and registered video films define and clarify the sequence of events that describe the observed phenomenon. the plasma is created by applying an audio power through the electrodes of an air gas discharge of 10 khz and up to 500 watts power supply. the discharge voltage is up to 900 volts: the discharge current flowing through the plasma attains 360 mA. it is found that the discharge system must attain its optimal working conditions in order to produce the amazing phenomena. the obtained plasma is classified as the maximum conditions borders of a γ-discharge type. at these conditions, the corresponding maximum electron temperature and density are 16 eV and 10 15 cm -3 respectively . the observation system succeeded to reveal and to clarify the sequence of the phenomenon events. in addition, by means of the scanning electron microscope and the energy dispersive x- ray systems, the effects on the electrodes surface are investigated and analyzed. the optical observations, in conjunction with the micrograph and surface microanalysis,demonstrate the collision occurrence, of powered agglomerations groups, to the electrode surface. detailed interpretation of that phenomenon suggests a molecular acceleration gaining their energy from the formed plasma due to optimal discharge working conditions. as a consequence, due to the ions agglomerates size this procedure could be considered as a mesoscopic acceleration technique.

  18. Automatic Detection and Classification of Audio Events for Road Surveillance Applications

    Directory of Open Access Journals (Sweden)

    Noor Almaadeed

    2018-06-01

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

  19. Feature Selection for Audio Surveillance in Urban Environment

    Directory of Open Access Journals (Sweden)

    KIKTOVA Eva

    2014-05-01

    Full Text Available This paper presents the work leading to the acoustic event detection system, which is designed to recognize two types of acoustic events (shot and breaking glass in urban environment. For this purpose, a huge front-end processing was performed for the effective parametric representation of an input sound. MFCC features and features computed during their extraction (MELSPEC and FBANK, then MPEG-7 audio descriptors and other temporal and spectral characteristics were extracted. High dimensional feature sets were created and in the next phase reduced by the mutual information based selection algorithms. Hidden Markov Model based classifier was applied and evaluated by the Viterbi decoding algorithm. Thus very effective feature sets were identified and also the less important features were found.

  20. 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.)

  1. Time-Scale Invariant Audio Data Embedding

    Directory of Open Access Journals (Sweden)

    Mansour Mohamed F

    2003-01-01

    Full Text Available We propose a novel algorithm for high-quality data embedding in audio. The algorithm is based on changing the relative length of the middle segment between two successive maximum and minimum peaks to embed data. Spline interpolation is used to change the lengths. To ensure smooth monotonic behavior between peaks, a hybrid orthogonal and nonorthogonal wavelet decomposition is used prior to data embedding. The possible data embedding rates are between 20 and 30 bps. However, for practical purposes, we use repetition codes, and the effective embedding data rate is around 5 bps. The algorithm is invariant after time-scale modification, time shift, and time cropping. It gives high-quality output and is robust to mp3 compression.

  2. Audio visual information materials for risk communication

    International Nuclear Information System (INIS)

    Gunji, Ikuko; Tabata, Rimiko; Ohuchi, Naomi

    2005-07-01

    Japan Nuclear Cycle Development Institute (JNC), Tokai Works set up the Risk Communication Study Team in January, 2001 to promote mutual understanding between the local residents and JNC. The Team has studied risk communication from various viewpoints and developed new methods of public relations which are useful for the local residents' risk perception toward nuclear issues. We aim to develop more effective risk communication which promotes a better mutual understanding of the local residents, by providing the risk information of the nuclear fuel facilities such a Reprocessing Plant and other research and development facilities. We explain the development process of audio visual information materials which describe our actual activities and devices for the risk management in nuclear fuel facilities, and our discussion through the effectiveness measurement. (author)

  3. 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...

  4. 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)

  5. 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...

  6. 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.

  7. 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.…

  8. 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…

  9. 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...

  10. 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…

  11. 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

  12. 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

  13. 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

  14. 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

  15. 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

  16. 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

  17. 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

  18. 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...

  19. 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...

  20. 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…

  1. 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

  2. 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.

  3. 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…

  4. 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.

  5. 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...

  6. 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.

  7. 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

  8. 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…

  9. ANALYSIS OF MULTIMODAL FUSION TECHNIQUES FOR AUDIO-VISUAL SPEECH RECOGNITION

    Directory of Open Access Journals (Sweden)

    D.V. Ivanko

    2016-05-01

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

  10. 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.

  11. 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.

  12. 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.

  13. 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 %....

  14. 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...

  15. 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.

  16. Integration of top-down and bottom-up information for audio organization and retrieval

    DEFF Research Database (Denmark)

    Jensen, Bjørn Sand

    The increasing availability of digital audio and music calls for methods and systems to analyse and organize these digital objects. This thesis investigates three elements related to such systems focusing on the ability to represent and elicit the user's view on the multimedia object and the system...... output. The aim is to provide organization and processing, which aligns with the understanding and needs of the users. Audio and music is often characterized by the large amount of heterogenous information. The rst aspect investigated is the integration of such multi-variate and multi-modal information...... (indirect scaling). Inference is performed by analytical and simulation based methods, including the Laplace approximation and expectation propagation. In order to minimize the cost of the often expensive and lengthly experimentation, sequential experiment design or active learning is supported. The setup...

  17. 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...

  18. Proper Use of Audio-Visual Aids: Essential for Educators.

    Science.gov (United States)

    Dejardin, Conrad

    1989-01-01

    Criticizes educators as the worst users of audio-visual aids and among the worst public speakers. Offers guidelines for the proper use of an overhead projector and the development of transparencies. (DMM)

  19. Ferrite bead effect on Class-D amplifier audio quality

    OpenAIRE

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

    2014-01-01

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

  20. Precision Scaling of Neural Networks for Efficient Audio Processing

    OpenAIRE

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

    2017-01-01

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

  1. 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?

  2. 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.

  3. 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.

  4. Audio-based Age and Gender Identification to Enhance the Recommendation of TV Content

    DEFF Research Database (Denmark)

    Shepstone, Sven Ewan; Tan, Zheng-Hua; Jensen, Søren Holdt

    2013-01-01

    Recommending TV content to groups of viewers is best carried out when relevant information such as the demographics of the group is available. However, it can be difficult and time consuming to extract information for every user in the group. This paper shows how an audio analysis of the age...... and gender of a group of users watching the TV can be used for recommending a sequence of N short TV content items for the group. First, a state of the art audio-based classifier determines the age and gender of each user in an M-user group and creates a group profile. A genetic recommender algorithm...... profile, thus ensuring that items are proportionally allocated to users with respect to their demographic categorization. The proposed system is compared to an ideal system where the group demographics are provided explicitly. Results using real speaker utterances show that, in spite of the inaccuracies...

  5. 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...

  6. Virtual Acoustic Displays for Teleconferencing: Intelligibility Advantage for "Telephone Grade" Audio

    Science.gov (United States)

    Begault, Durand R.; Null, Cynthia H. (Technical Monitor)

    1994-01-01

    Speech intelligibility was evaluated using a virtual acoustic ("3-D audio") display using the method specified by ANSI. Ten subjects were evaluated with stimuli either unfiltered or low-pass filtered at 4 kHz. Results show virtual acoustic techniques are advantageous for both full-bandwidth (44.1 kHz srate) and low (8 kHz srate) bandwidth "telephone-grade" teleconferencing systems.

  7. 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.

  8. AUTOMATIC SEGMENTATION OF BROADCAST AUDIO SIGNALS USING AUTO ASSOCIATIVE NEURAL NETWORKS

    Directory of Open Access Journals (Sweden)

    P. Dhanalakshmi

    2010-12-01

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

  9. 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...

  10. Mobile Message Services Using Text, Audio or Video for Improving the Learning Infrastructure in Higher Education

    Directory of Open Access Journals (Sweden)

    Björn Olof Hedin

    2006-06-01

    Full Text Available This study examines how media files sent to mobile phones can be used to improve education at universities, and describes a prototype implement of such a system using standard components. To accomplish this, university students were equipped with mobile phones and software that allowed teachers to send text-based, audio-based and video-based messages to the students. Data was collected using questionnaires, focus groups and log files. The conclusions were that students preferred to have information and learning content sent as text, rather than audio or video. Text messages sent to phones should be no longer than 2000 characters. The most appreciated services were notifications of changes in course schedules, short lecture introductions and reminders. The prototype showed that this functionality is easy to implement using standard components.

  11. Neuromorphic Audio-Visual Sensor Fusion on a Sound-Localising Robot

    Directory of Open Access Journals (Sweden)

    Vincent Yue-Sek Chan

    2012-02-01

    Full Text Available This paper presents the first robotic system featuring audio-visual sensor fusion with neuromorphic sensors. We combine a pair of silicon cochleae and a silicon retina on a robotic platform to allow the robot to learn sound localisation through self-motion and visual feedback, using an adaptive ITD-based sound localisation algorithm. After training, the robot can localise sound sources (white or pink noise in a reverberant environment with an RMS error of 4 to 5 degrees in azimuth. In the second part of the paper, we investigate the source binding problem. An experiment is conducted to test the effectiveness of matching an audio event with a corresponding visual event based on their onset time. The results show that this technique can be quite effective, despite its simplicity.

  12. Can audio coached 4D CT emulate free breathing during the treatment course?

    DEFF Research Database (Denmark)

    Persson, Gitte F; Nygaard, Ditte E; Olsen, Mikael

    2008-01-01

    BACKGROUND: The image quality of 4DCT depends on breathing regularity. Respiratory audio coaching may improve regularity and reduce motion artefacts. We question the safety of coached planning 4DCT without coaching during treatment. We investigated the possibility of coaching to a more stable...... breathing without changing the breathing amplitude. The interfraction variation of the breathing cycle amplitude in free and coached breathing was studied as well as the possible impact of fatigue on longer coaching sessions. METHODS: Thirteen volunteers completed respiratory audio coaching on 3 days within...... a 2 week period. An external marker system monitoring the motion of the thoraco-abdominal wall was used to track the respiration. On all days, free breathing and two coached breathing curves were recorded. We assumed that free versus coached breathing from day 1 (reference session) simulated breathing...

  13. EXPERIMENTAL STUDIES FOR DEVELOPMENT HIGH-POWER AUDIO SPEAKER DEVICES PERFORMANCE USING PERMANENT NdFeB MAGNETS SPECIAL TECHNOLOGY

    Directory of Open Access Journals (Sweden)

    Constantin D. STĂNESCU

    2013-05-01

    Full Text Available In this paper the authors shows the research made for improving high-power audio speaker devices performance using permanent NdFeB magnets special technology. Magnetic losses inside these audio devices are due to mechanical system frictions and to thermal effect of Joules eddy currents. In this regard, by special technology, were made conical surfaces at top plate and center pin. Analysing results obtained by modelling the magnetic circuit finite element method using electronic software package,was measured increase efficiency by over 10 %, from 1,136T to13T.

  14. Audio-visual assistance in co-creating transition knowledge

    Science.gov (United States)

    Hezel, Bernd; Broschkowski, Ephraim; Kropp, Jürgen P.

    2013-04-01

    Earth system and climate impact research results point to the tremendous ecologic, economic and societal implications of climate change. Specifically people will have to adopt lifestyles that are very different from those they currently strive for in order to mitigate severe changes of our known environment. It will most likely not suffice to transfer the scientific findings into international agreements and appropriate legislation. A transition is rather reliant on pioneers that define new role models, on change agents that mainstream the concept of sufficiency and on narratives that make different futures appealing. In order for the research community to be able to provide sustainable transition pathways that are viable, an integration of the physical constraints and the societal dynamics is needed. Hence the necessary transition knowledge is to be co-created by social and natural science and society. To this end, the Climate Media Factory - in itself a massively transdisciplinary venture - strives to provide an audio-visual connection between the different scientific cultures and a bi-directional link to stake holders and society. Since methodology, particular language and knowledge level of the involved is not the same, we develop new entertaining formats on the basis of a "complexity on demand" approach. They present scientific information in an integrated and entertaining way with different levels of detail that provide entry points to users with different requirements. Two examples shall illustrate the advantages and restrictions of the approach.

  15. 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. 

  16. Imagination and Modern Audio Visual Form

    Directory of Open Access Journals (Sweden)

    Ana Đurković

    2017-09-01

    Full Text Available Through three episodes Archetype of modern fairy tales, the mysterious world of fantasy and reality,tell as a serious story about archetypes, symbols, knowledge of good and evil. Rts editor: Natasa Neskovic Written and directed by: Suncica Jergovic Editing: Ana Djurkovic How to illuminate concept of phantasy and affective factors in our imagination a priori something so imaginary, by their genetic provenance, such as a movie scene, or digital picture and sound. You can not always avoid the association to a valid phrase of arnhajm’s truth: mass age -massage: the medium is the message. In elementary and tersely definition of „the shot“ from Plaževsky film language there is term for „le cadre“, however these are selected bits of reality, immanent frame that contains the individual act of images divided of the continent’s view of reality, handling the specific code of semantic value, when its’s imaginative, of course, by aesthetic categories and evaluations. In this type of positive simulacrum, it can not be better segment for the current thinking about the limits of imagination and truth in contemporary media, and contemporary global environment, than the original audio-visual forms through whose prism we search throught a fairy tale in a same time myth and imagination as well as exploring its overall impact on the personality. Everything can be a fairy tale, even false, amoral platitudes politicized by political lobbies in a contemporary existing power sistems, but this is no fairy tale authenticity in it, or creative act, nor humanity and artificial and historical entity of a man that is always present in the ethical effort of a true artist. So, we are investigating the conditions of creative images, modalities of audiovisual media in film language,and it is the archetype of the fairy tale, which, with its psychodynamics still exists and which is removed when the modern man is tired of lies and simulations during his global

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

    Science.gov (United States)

    Pimmer, Christoph; Mbvundula, Francis

    2018-01-01

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

  18. 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....

  19. 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)

  20. 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...

  1. 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.

  2. 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.

  3. Distortion-Free 1-Bit PWM Coding for Digital Audio Signals

    Directory of Open Access Journals (Sweden)

    John Mourjopoulos

    2007-01-01

    Full Text Available Although uniformly sampled pulse width modulation (UPWM represents a very efficient digital audio coding scheme for digital-to-analog conversion and full-digital amplification, it suffers from strong harmonic distortions, as opposed to benign non-harmonic artifacts present in analog PWM (naturally sampled PWM, NPWM. Complete elimination of these distortions usually requires excessive oversampling of the source PCM audio signal, which results to impractical realizations of digital PWM systems. In this paper, a description of digital PWM distortion generation mechanism is given and a novel principle for their minimization is proposed, based on a process having some similarity to the dithering principle employed in multibit signal quantization. This conditioning signal is termed “jither” and it can be applied either in the PCM amplitude or the PWM time domain. It is shown that the proposed method achieves significant decrement of the harmonic distortions, rendering digital PWM performance equivalent to that of source PCM audio, for mild oversampling (e.g., ×4 resulting to typical PWM clock rates of 90 MHz.

  4. Distortion-Free 1-Bit PWM Coding for Digital Audio Signals

    Directory of Open Access Journals (Sweden)

    Mourjopoulos John

    2007-01-01

    Full Text Available Although uniformly sampled pulse width modulation (UPWM represents a very efficient digital audio coding scheme for digital-to-analog conversion and full-digital amplification, it suffers from strong harmonic distortions, as opposed to benign non-harmonic artifacts present in analog PWM (naturally sampled PWM, NPWM. Complete elimination of these distortions usually requires excessive oversampling of the source PCM audio signal, which results to impractical realizations of digital PWM systems. In this paper, a description of digital PWM distortion generation mechanism is given and a novel principle for their minimization is proposed, based on a process having some similarity to the dithering principle employed in multibit signal quantization. This conditioning signal is termed "jither" and it can be applied either in the PCM amplitude or the PWM time domain. It is shown that the proposed method achieves significant decrement of the harmonic distortions, rendering digital PWM performance equivalent to that of source PCM audio, for mild oversampling (e.g., resulting to typical PWM clock rates of 90 MHz.

  5. 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.

  6. 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.

  7. 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...

  8. 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...

  9. Cambridge English First 2 audio CDs : authentic examination papers

    CERN Document Server

    2016-01-01

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

  10. Audio engineering 101 a beginner's guide to music production

    CERN Document Server

    Dittmar, Tim

    2013-01-01

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

  11. 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

  12. Can audio recording of outpatient consultations improve patient outcome?

    DEFF Research Database (Denmark)

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

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

  13. 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...

  14. The Single- and Multichannel Audio Recordings Database (SMARD)

    DEFF Research Database (Denmark)

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

    2014-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Mark Guttenbrunner

    2011-03-01

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

  16. PHYSIOLOGICAL MONITORING OPERATORS ACS IN AUDIO-VISUAL SIMULATION OF AN EMERGENCY

    Directory of Open Access Journals (Sweden)

    S. S. Aleksanin

    2010-01-01

    Full Text Available In terms of ship simulator automated control systems we have investigated the information content of physiological monitoring cardiac rhythm to assess the reliability and noise immunity of operators of various specializations with audio-visual simulation of an emergency. In parallel, studied the effectiveness of protection against the adverse effects of electromagnetic fields. Monitoring of cardiac rhythm in a virtual crash it is possible to differentiate the degree of voltage regulation systems of body functions of operators on specialization and note the positive effect of the use of means of protection from exposure of electromagnetic fields.

  17. Evaluation of an Audio Cassette Tape Lecture Course

    Science.gov (United States)

    Blank, Jerome W.

    1975-01-01

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

  18. Audio-visual materials usage preference among agricultural ...

    African Journals Online (AJOL)

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

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

    Science.gov (United States)

    Spadaccini, Jim

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

  20. A Power Efficient Audio Amplifier Combining Switching and Linear Techniques

    NARCIS (Netherlands)

    van der Zee, Ronan A.R.; van Tuijl, Adrianus Johannes Maria

    1998-01-01

    Integrated Class D audio amplifiers are very power efficient, but require an external filter which prevents further integration. Also due to this filter, large feedback factors are hard to realise, so that the load influences the distortion- and transfer characteristics. The amplifier presented in

  1. 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…

  2. Haptic and Visual feedback in 3D Audio Mixing Interfaces

    DEFF Research Database (Denmark)

    Gelineck, Steven; Overholt, Daniel

    2015-01-01

    This paper describes the implementation and informal evaluation of a user interface that explores haptic feedback for 3D audio mixing. The implementation compares different approaches using either the LEAP Motion for mid-air hand gesture control, or the Novint Falcon for active haptic feed- back...

  3. Audio-Visual Aid in Teaching "Fatty Liver"

    Science.gov (United States)

    Dash, Sambit; Kamath, Ullas; Rao, Guruprasad; Prakash, Jay; Mishra, Snigdha

    2016-01-01

    Use of audio visual tools to aid in medical education is ever on a rise. Our study intends to find the efficacy of a video prepared on "fatty liver," a topic that is often a challenge for pre-clinical teachers, in enhancing cognitive processing and ultimately learning. We prepared a video presentation of 11:36 min, incorporating various…

  4. Studies on a Spatialized Audio Interface for Sonar

    Science.gov (United States)

    2011-10-03

    addition of spatialized audio to visual displays for sonar is much akin to the development of talking movies in the early days of cinema and can be...than using the brute-force approach. PCA is one among several techniques that share similarities with the computational architecture of a

  5. 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…

  6. Market potential for interactive audio-visual media

    NARCIS (Netherlands)

    Leurdijk, A.; Limonard, S.

    2005-01-01

    NM2 (New Media for a New Millennium) develops tools for interactive, personalised and non-linear audio-visual content that will be tested in seven pilot productions. This paper looks at the market potential for these productions from a technological, a business and a users' perspective. It shows

  7. Towards a universal representation for audio information retrieval and analysis

    DEFF Research Database (Denmark)

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

    2013-01-01

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

  8. Computationally efficient clustering of audio-visual meeting data

    NARCIS (Netherlands)

    Hung, H.; Friedland, G.; Yeo, C.; Shao, L.; Shan, C.; Luo, J.; Etoh, M.

    2010-01-01

    This chapter presents novel computationally efficient algorithms to extract semantically meaningful acoustic and visual events related to each of the participants in a group discussion using the example of business meeting recordings. The recording setup involves relatively few audio-visual sensors,

  9. Multi Carrier Modulator for Switch-Mode Audio Power Amplifiers

    DEFF Research Database (Denmark)

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

    2008-01-01

    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, in particular radio receivers. Lowering the EMI of swit...

  10. Audio Quality Assurance : An Application of Cross Correlation

    DEFF Research Database (Denmark)

    Jurik, Bolette Ammitzbøll; Nielsen, Jesper Asbjørn Sindahl

    2012-01-01

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

  11. 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...

  12. Audio effects on haptics perception during drilling simulation

    Directory of Open Access Journals (Sweden)

    Yair Valbuena

    2017-06-01

    Full Text Available Virtual reality has provided immersion and interactions through computer generated environments attempting to reproduce real life experiences through sensorial stimuli. Realism can be achieved through multimodal interactions which can enhance the user’s presence within the computer generated world. The most notorious advances in virtual reality can be seen in computer graphics visuals, where photorealism is the norm thriving to overcome the uncanny valley. Other advances have followed related to sound, haptics, and in a lesser manner smell and taste feedback. Currently, virtual reality systems (multimodal immersion and interactions through visual-haptic-sound are being massively used in entertainment (e.g., cinema, video games, art, and in non-entertainment scenarios (e.g., social inclusion, educational, training, therapy, and tourism. Moreover, the cost reduction of virtual reality technologies has resulted in the availability at a consumer-level of various haptic, headsets, and motion tracking devices. Current consumer-level devices offer low-fidelity experiences due to the properties of the sensors, displays, and other electro-mechanical devices, that may not be suitable for high-precision or realistic experiences requiring dexterity. However, research has been conducted on how toovercome or compensate the lack of high fidelity to provide an engaging user experience using storytelling, multimodal interactions and gaming elements. Our work focuses on analyzing the possible effects of auditory perception on haptic feedback within a drilling scenario. Drilling involves multimodal interactions and it is a task with multiple applications in medicine, crafting, and construction. We compare two drilling scenarios were two groups of participants had to drill through wood while listening to contextual and non-contextual audios. We gathered their perception using a survey after the task completion. From the results, we believe that sound does

  13. 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.

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

    NARCIS (Netherlands)

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

    2015-01-01

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

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

    Science.gov (United States)

    Anderson, Gerald D.; Olson, David B.

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

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

    Science.gov (United States)

    Dickinson Public Schools, ND. Instructional Media Center.

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

  17. 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...

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

    OpenAIRE

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

    2013-01-01

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

  19. On the definition of adapted audio/video profiles for high-quality video calling services over LTE/4G

    Science.gov (United States)

    Ndiaye, Maty; Quinquis, Catherine; Larabi, Mohamed Chaker; Le Lay, Gwenael; Saadane, Hakim; Perrine, Clency

    2014-01-01

    During the last decade, the important advances and widespread availability of mobile technology (operating systems, GPUs, terminal resolution and so on) have encouraged a fast development of voice and video services like video-calling. While multimedia services have largely grown on mobile devices, the generated increase of data consumption is leading to the saturation of mobile networks. In order to provide data with high bit-rates and maintain performance as close as possible to traditional networks, the 3GPP (The 3rd Generation Partnership Project) worked on a high performance standard for mobile called Long Term Evolution (LTE). In this paper, we aim at expressing recommendations related to audio and video media profiles (selection of audio and video codecs, bit-rates, frame-rates, audio and video formats) for a typical video-calling services held over LTE/4G mobile networks. These profiles are defined according to targeted devices (smartphones, tablets), so as to ensure the best possible quality of experience (QoE). Obtained results indicate that for a CIF format (352 x 288 pixels) which is usually used for smartphones, the VP8 codec provides a better image quality than the H.264 codec for low bitrates (from 128 to 384 kbps). However sequences with high motion, H.264 in slow mode is preferred. Regarding audio, better results are globally achieved using wideband codecs offering good quality except for opus codec (at 12.2 kbps).

  20. 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.

  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…

  2. 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...

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

    Science.gov (United States)

    2016-03-01

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

  4. 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…

  5. 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...

  6. 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...

  7. Análisis de viabilidad y desarrollo de una app de audio

    OpenAIRE

    Vega Díaz, Jorge

    2016-01-01

    Els terminals de comunicació mòbil incorporen sistemes d'enregistrament i reproducció de senyal d'àudio. El projecte preten analitzar diferents funcions de tractament de senyal en temps real, incorporant filtratge, anàlisi espectral i edició. This project consists on the design and implementation of an audio editing tool for Android devices, with similar features that we can find in operating systems like Linux, Windows or Mac. This app should have the basic editing operations, such as aud...

  8. An interactive audio-visual installation using ubiquitous hardware and web-based software deployment

    Directory of Open Access Journals (Sweden)

    Tiago Fernandes Tavares

    2015-05-01

    Full Text Available This paper describes an interactive audio-visual musical installation, namely MOTUS, that aims at being deployed using low-cost hardware and software. This was achieved by writing the software as a web application and using only hardware pieces that are built-in most modern personal computers. This scenario implies in specific technical restrictions, which leads to solutions combining both technical and artistic aspects of the installation. The resulting system is versatile and can be freely used from any computer with Internet access. Spontaneous feedback from the audience has shown that the provided experience is interesting and engaging, regardless of the use of minimal hardware.

  9. Surround by Sound: A Review of Spatial Audio Recording and Reproduction

    Directory of Open Access Journals (Sweden)

    Wen Zhang

    2017-05-01

    Full Text Available In this article, a systematic overview of various recording and reproduction techniques for spatial audio is presented. While binaural recording and rendering is designed to resemble the human two-ear auditory system and reproduce sounds specifically for a listener’s two ears, soundfield recording and reproduction using a large number of microphones and loudspeakers replicate an acoustic scene within a region. These two fundamentally different types of techniques are discussed in the paper. A recent popular area, multi-zone reproduction, is also briefly reviewed in the paper. The paper is concluded with a discussion of the current state of the field and open problems.

  10. Investigation of multiple visualisation techniques and dynamic queries in conjunction with direct sonification to support the browsing of audio resources

    OpenAIRE

    Brazil, Eoin

    2003-01-01

    non-peer-reviewed In this thesis, a prototype system for the browsing of audio resources was developed and an initial evaluation of this system was performed. The main contributions of this thesis are dynamic queries and multiple visualisation techniques in conjunction with direct sonification. Dynamic queries are queries that provide immediate feedback while maintaining consistency between the queries themselves and the graphical/auditory display. The multiple visualisation techniques are...

  11. 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

  12. 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

  13. Audio-Biofeedback training for posture and balance in Patients with Parkinson's disease

    Directory of Open Access Journals (Sweden)

    Zijlstra Wiebren

    2011-06-01

    Full Text Available Abstract Background Patients with Parkinson's disease (PD suffer from dysrhythmic and disturbed gait, impaired balance, and decreased postural responses. These alterations lead to falls, especially as the disease progresses. Based on the observation that postural control improved in patients with vestibular dysfunction after audio-biofeedback training, we tested the feasibility and effects of this training modality in patients with PD. Methods Seven patients with PD were included in a pilot study comprised of a six weeks intervention program. The training was individualized to each patient's needs and was delivered using an audio-biofeedback (ABF system with headphones. The training was focused on improving posture, sit-to-stand abilities, and dynamic balance in various positions. Non-parametric statistics were used to evaluate training effects. Results The ABF system was well accepted by all participants with no adverse events reported. Patients declared high satisfaction with the training. A significant improvement of balance, as assessed by the Berg Balance Scale, was observed (improvement of 3% p = 0.032, and a trend in the Timed up and go test (improvement of 11%; p = 0.07 was also seen. In addition, the training appeared to have a positive influence on psychosocial aspects of the disease as assessed by the Parkinson's disease quality of life questionnaire (PDQ-39 and the level of depression as assessed by the Geriatric Depression Scale. Conclusions This is, to our knowledge, the first report demonstrating that audio-biofeedback training for patients with PD is feasible and is associated with improvements of balance and several psychosocial aspects.

  14. 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.

  15. 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.

  16. “Wrapping” X3DOM around Web Audio API

    Directory of Open Access Journals (Sweden)

    Andreas Stamoulias

    2015-12-01

    Full Text Available Spatial sound has a conceptual role in the Web3D environments, due to highly realism scenes that can provide. Lately the efforts are concentrated on the extension of the X3D/ X3DOM through spatial sound attributes. This paper presents a novel method for the introduction of spatial sound components in the X3DOM framework, based on X3D specification and Web Audio API. The proposed method incorporates the introduction of enhanced sound nodes for X3DOM which are derived by the implementation of the X3D standard components, enriched with accessional features of Web Audio API. Moreover, several examples-scenarios developed for the evaluation of our approach. The implemented examples established the achievability of new registered nodes in X3DOM, for spatial sound characteristics in Web3D virtual worlds.

  17. Audio teleconferencing: creative use of a forgotten innovation.

    Science.gov (United States)

    Mather, Carey; Marlow, Annette

    2012-06-01

    As part of a regional School of Nursing and Midwifery's commitment to addressing recruitment and retention issues, approximately 90% of second year undergraduate student nurses undertake clinical placements at: multipurpose centres; regional or district hospitals; aged care; or community centres based in rural and remote regions within the State. The remaining 10% undertake professional experience placement in urban areas only. This placement of a large cohort of students, in low numbers in a variety of clinical settings, initiated the need to provide consistent support to both students and staff at these facilities. Subsequently the development of an audio teleconferencing model of clinical facilitation to guide student teaching and learning and to provide support to registered nurse preceptors in clinical practice was developed. This paper draws on Weimer's 'Personal Accounts of Change' approach to describe, discuss and evaluate the modifications that have occurred since the inception of this audio teleconferencing model (Weimer, 2006).

  18. Audio Visual Media Components in Educational Game for Elementary Students

    Directory of Open Access Journals (Sweden)

    Meilani Hartono

    2016-12-01

    Full Text Available The purpose of this research was to review and implement interactive audio visual media used in an educational game to improve elementary students’ interest in learning mathematics. The game was developed for desktop platform. The art of the game was set as 2D cartoon art with animation and audio in order to make students more interest. There were four mini games developed based on the researches on mathematics study. Development method used was Multimedia Development Life Cycle (MDLC that consists of requirement, design, development, testing, and implementation phase. Data collection methods used are questionnaire, literature study, and interview. The conclusion is elementary students interest with educational game that has fun and active (moving objects, with fast tempo of music, and carefree color like blue. This educational game is hoped to be an alternative teaching tool combined with conventional teaching method.

  19. Computationally Efficient Clustering of Audio-Visual Meeting Data

    Science.gov (United States)

    Hung, Hayley; Friedland, Gerald; Yeo, Chuohao

    This chapter presents novel computationally efficient algorithms to extract semantically meaningful acoustic and visual events related to each of the participants in a group discussion using the example of business meeting recordings. The recording setup involves relatively few audio-visual sensors, comprising a limited number of cameras and microphones. We first demonstrate computationally efficient algorithms that can identify who spoke and when, a problem in speech processing known as speaker diarization. We also extract visual activity features efficiently from MPEG4 video by taking advantage of the processing that was already done for video compression. Then, we present a method of associating the audio-visual data together so that the content of each participant can be managed individually. The methods presented in this article can be used as a principal component that enables many higher-level semantic analysis tasks needed in search, retrieval, and navigation.

  20. 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....

  1. Pitch range variations improve cognitive processing of audio messages

    OpenAIRE

    Rodero Antón, Emma; Potter, Rob F.; Prieto Vives, Pilar, 1965-

    2017-01-01

    This study explores the effect of different speaker intonation strategies in audio messages on attention, autonomic arousal, and memory. An experiment was conducted in which participants listened to 16 radio commercials produced to vary in pitch range across sentences. Dependent variables were self-reported effectiveness and adequacy, psychophysiological arousal and attention, immediate word recall and recognition of information. Results showed that messages conveyed with pitch variations ach...

  2. Comparison of Linear Prediction Models for Audio Signals

    Directory of Open Access Journals (Sweden)

    2009-03-01

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

  3. 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.

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

    Science.gov (United States)

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

    2014-12-30

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

  5. Formal usability evaluation of audio track widget graphical representation for two-dimensional stage audio mixing interface

    OpenAIRE

    Dewey, Christopher; Wakefield, Jonathan P.

    2017-01-01

    The two-dimensional stage paradigm (2DSP) has been suggested as an alternative audio mixing interface (AMI). This study seeks to refine the 2DSP by formally evaluating graphical track visualisation styles. Track visualisations considered were text only, circles containing text, individually coloured circles containing text, circles colour coded by instrument type with text, icons with text superimposed, circles with RMS related dynamic opacity and a traditional AMI. The usability evaluation f...

  6. Review: Music analysis and retrieval systems for audio signals

    NARCIS (Netherlands)

    van den Broek, Egon

    2005-01-01

    This paper provides an overview of four years of the authors’ research on music information retrieval (MIR), omitting technical details. An overview of terminology, music analysis techniques, and research done in the last five years is also provided. Some of the results achieved by the authors are

  7. 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.

  8. 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

  9. 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.

  10. APPLICATION OF CONTROLLED SOURCE AUDIO MAGNETOTELLURIC (CSAMT AT GEOTHERMAL

    Directory of Open Access Journals (Sweden)

    Susilawati S.

    2017-04-01

    Full Text Available CSAMT or Controlled Source Audio-Magnetotelluric is one of the Geophysics methods to determine the resistivity of rock under earth surface. CSAMT method utilizes artificial stream and injected into the ground, the frequency of artificial sources ranging from 0.1 Hz to 10 kHz, CSAMT data source effect correction is inverted. From the inversion results showed that there is a layer having resistivity values ranged between 2.5 Ω.m – 15 Ω.m, which is interpreted that the layer is clay.

  11. Digital audio recordings improve the outcomes of patient consultations

    DEFF Research Database (Denmark)

    Wolderslund, Maiken; Kofoed, Poul-Erik; Holst, René

    2017-01-01

    OBJECTIVES: To investigate the effects on patients' outcome of the consultations when provided with: a Digital Audio Recording (DAR) of the consultation and a Question Prompt List (QPL). METHODS: This is a three-armed randomised controlled cluster trial. One group of patients received standard care......, while the other two groups received either the QPL in combination with a recording of their consultation or only the recording. Patients from four outpatient clinics participated: Paediatric, Orthopaedic, Internal Medicine, and Urology. The effects were evaluated by patient-administered questionnaires...

  12. Audio-haptic interaction in simulated walking experiences

    DEFF Research Database (Denmark)

    Serafin, Stefania

    2011-01-01

    and interchangeable use of the haptic and auditory modality in floor interfaces, and for the synergy of perception and action in capturing and guiding human walking. We describe the technology developed in the context of this project, together with some experiments performed to evaluate the role of auditory......In this paper an overview of the work conducted on audio-haptic physically based simulation and evaluation of walking is provided. This work has been performed in the context of the Natural Interactive Walking (NIW) project, whose goal is to investigate possibilities for the integrated...... and haptic feedback in walking tasks....

  13. An assessment of individualized technical ear training for audio production.

    Science.gov (United States)

    Kim, Sungyoung

    2015-07-01

    An individualized technical ear training method is compared to a non-individualized method. The efficacy of the individualized method is assessed using a standardized test conducted before and after the training period. Participants who received individualized training improved better than the control group on the test. Results indicate the importance of individualized training for acquisition of spectrum-identification and spectrum-matching skills. Individualized training, therefore, should be implemented by default into technical ear training programs used in audio production industry and education.

  14. 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.

  15. Digital video and audio broadcasting technology a practical engineering guide

    CERN Document Server

    Fischer, Walter

    2010-01-01

    Digital Video and Audio Broadcasting Technology - A Practical Engineering Guide' deals with all the most important digital television, sound radio and multimedia standards such as MPEG, DVB, DVD, DAB, ATSC, T-DMB, DMB-T, DRM and ISDB-T. The book provides an in-depth look at these subjects in terms of practical experience. In addition it contains chapters on the basics of technologies such as analog television, digital modulation, COFDM or mathematical transformations between time and frequency domains. The attention in the respective field under discussion is focussed on aspects of measuring t

  16. 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.

  17. [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.

  18. Amplificador de audio en clase A para auriculares

    OpenAIRE

    Martín Ruiz, Manuel

    2012-01-01

    El presente proyecto muestra el desarrollo, la simulación y la implantación de un amplificador de audio de altas prestaciones, empleando para ello transistores discretos y amplificadores operacionales sobre una PCB diseñada previamente con un programa software. La aplicación de este amplificador será como amplificador de potencia para auriculares de alta impedancia. El circuito empleará una técnica de realimentación directa sobre los auriculares conectados a 4 hilos. El amplificador incorpora...

  19. 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)

  20. 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.

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

    DEFF Research Database (Denmark)

    Ljusev, Petar; Andersen, Michael Andreas E.

    2005-01-01

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

  2. 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; ·     �...

  3. 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

  4. On the relative importance of audio and video in the presence of packet losses

    DEFF Research Database (Denmark)

    Korhonen, Jari; Reiter, Ulrich; Myakotnykh, Eugene

    2010-01-01

    In streaming applications, unequal protection of audio and video tracks may be necessary to maintain the optimal perceived overall quality. For this purpose, the application should be aware of the relative importance of audio and video in an audiovisual sequence. In this paper, we propose...... a subjective test arrangement for finding the optimal tradeoff between subjective audio and video qualities in situations when it is not possible to have perfect quality for both modalities concurrently. Our results show that content poses a significant impact on the preferred compromise between audio...... and video quality, but also that the currently used classification criteria for content are not sufficient to predict the users’ preference...

  5. 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.

  6. 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.

  7. A Model of Distraction in an Audio-on-Audio Interference Situation with Music Program Material

    DEFF Research Database (Denmark)

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

    2015-01-01

    listener can be viewed as having a personal sound zone system. In order to evaluate and optimize such situations in a perceptually relevant manner, the authors created a predictive model using the features that contribute to the distraction from unwanted sounds. Feature extraction was motivated...

  8. 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)

  9. 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.

  10. Pengembangan Trainer Equalizer Grafis dan Parametris sebagai Media Pembelajaran Mata Kuliah Praktik Sistem Audio

    Directory of Open Access Journals (Sweden)

    Bekti Wulandari

    2016-01-01

    Full Text Available The objectives of this study were to determine the development process of graphic and parametric equalizer trainer media in the course of Audio System and to examine the quality of the media. The development process of the media referred to the model offered by Lee and Owens. The research procedures consisted of assessment/analysis, designing, implementation and evaluation. The results of the study showed that the percentages for the quality aspect of the media and the companion module were 4.31 and 4.42 respectively. Based on these two aspects, it was obtained the overall percentage that was equal to 4.36. Whereas from the process of the trial by the students, the percentages of the media and the companion module were 4.47 and 4.36 respectively. Thus the level of validity and feasibility of the media was categorized as very good.

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

    Energy Technology Data Exchange (ETDEWEB)

    Kierkegaard, P.

    1998-10-01

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

  12. Audio Logo Recognition, Reduced Articulation and Coding Orientation

    DEFF Research Database (Denmark)

    Bonde, Anders; Hansen, Allan Grutt

    2013-01-01

    In this paper we explore an interdisciplinary theoretical framework for the analysis of corporate audio logos and their effectiveness regarding recognisability and identification. This is done by combining three different academic disciplines: 1) social semiotics, 2) branding theory and 3) music...... on musicological descriptors. We consider as a starting point Kress and Van Leeuwen’s (1996, 2006) conceptualisation of ‘modality’, which is central to their ‘visual grammar’ theory and subsequently extended to auditory expressions such as spoken language, music and sound effects (Van Leeuwen, 1999). While...... connected to notions of brand recognisability and brand identification, thus resulting in the concept of ‘Reduced Articulation Form’ (RAF). The concept has been tested empirically through a survey of 137 upper secondary school students. On the basis of a conditioning experiment, manipulating five existing...

  13. Audio collection in the SASA Institute of Musicology

    Directory of Open Access Journals (Sweden)

    Lajić-Mihajlović Danka

    2010-01-01

    Full Text Available The paper is relating to audio collection of the Institute of Musicology SASA as extremely important part of this institution’s fund. The collection comprises of valuable sound materials, especially significant collections of fieldwork recordings of traditional folk and church music, as also recordings of pieces of the 19th and 20th century Serbian composers. Information on sound carriers, methodologies and circumstances in which the recordings have been made, their preservation and further treatment with modern technologies, are a part of ethnomusicological and musicological histories in Serbia. According to number of sound recordings, diachronical dimensions that encompass, geographical areas and genre diversity, this collection is one of the most important sound collections of scientific profile in Serbia.

  14. 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.

  15. 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.

  16. 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...

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

    DEFF Research Database (Denmark)

    Ljusev, Petar; Andersen, Michael Andreas E.

    2004-01-01

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

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

    Science.gov (United States)

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

    2010-01-01

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

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

    Science.gov (United States)

    Oringel, Robert S.

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

  20. A Perceptual Model for Sinusoidal Audio Coding Based on Spectral Integration

    NARCIS (Netherlands)

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

    2005-01-01

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

  1. A perceptual model for sinusoidal audio coding based on spectral integration

    NARCIS (Netherlands)

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

    2005-01-01

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

  2. Changes of the Prefrontal EEG (Electroencephalogram) Activities According to the Repetition of Audio-Visual Learning.

    Science.gov (United States)

    Kim, Yong-Jin; Chang, Nam-Kee

    2001-01-01

    Investigates the changes of neuronal response according to a four time repetition of audio-visual learning. Obtains EEG data from the prefrontal (Fp1, Fp2) lobe from 20 subjects at the 8th grade level. Concludes that the habituation of neuronal response shows up in repetitive audio-visual learning and brain hemisphericity can be changed by…

  3. 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.

  4. 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

  5. Toward Personal and Emotional Connectivity in Mobile Higher Education through Asynchronous Formative Audio Feedback

    Science.gov (United States)

    Rasi, Päivi; Vuojärvi, Hanna

    2018-01-01

    This study aims to develop asynchronous formative audio feedback practices for mobile learning in higher education settings. The development was conducted in keeping with the principles of design-based research. The research activities focused on an inter-university online course, within which the use of instructor audio feedback was tested,…

  6. 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

  7. 16 CFR 307.8 - Requirements for disclosure in audiovisual and audio advertising.

    Science.gov (United States)

    2010-01-01

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

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

    Science.gov (United States)

    Vigil, Kathleen Murray

    2017-01-01

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

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

    NARCIS (Netherlands)

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

    2010-01-01

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

  10. An Exploratory Evaluation of User Interfaces for 3D Audio Mixing

    DEFF Research Database (Denmark)

    Gelineck, Steven; Korsgaard, Dannie Michael

    2015-01-01

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

  11. A Preliminary Investigation into the Search Behaviour of Users in a Collection of Digitized Broadcast Audio

    DEFF Research Database (Denmark)

    Lund, Haakon; Skov, Mette; Larsen, Birger

    2014-01-01

    An increasing number of large digitized audio-visual collections within digital humanities have recently been made available for users. Often access to digitized audio-visual collections is hampered by little and inconsistent metadata. This paper presents the preliminary findings from a study of ...

  12. Vertigo with sudden hearing loss: audio-vestibular characteristics.

    Science.gov (United States)

    Pogson, Jacob M; Taylor, Rachael L; Young, Allison S; McGarvie, Leigh A; Flanagan, Sean; Halmagyi, G Michael; Welgampola, Miriam S

    2016-10-01

    Acute vertigo with sudden sensorineural hearing loss (SSNHL) is a rare clinical emergency. Here, we report the audio-vestibular test profiles of 27 subjects who presented with these symptoms. The vestibular test battery consisted of a three-dimensional video head impulse test (vHIT) of semicircular canal function and recording ocular and cervical vestibular-evoked myogenic potentials (oVEMP, cVEMP) to test otolith dysfunction. Unlike vestibular neuritis, where the horizontal and anterior canals with utricular function are more frequently impaired, 74 % of subjects with vertigo and SSNHL demonstrated impairment of the posterior canal gain (0.45 ± 0.20). Only 41 % showed impairment of the horizontal canal gains (0.78 ± 0.27) and 30 % of the anterior canal gains (0.79 ± 0.26), while 38 % of oVEMPs [asymmetry ratio (AR) = 41.0 ± 41.3 %] and 33 % of cVEMPs (AR = 47.3 ± 41.2 %) were significantly asymmetrical. Twenty-three subjects were diagnosed with labyrinthitis/labyrinthine infarction in the absence of evidence for an underlying pathology. Four subjects had a definitive diagnosis [Ramsay Hunt Syndrome, vestibular schwannoma, anterior inferior cerebellar artery (AICA) infarction, and traction injury]. Ischemia involving the common-cochlear or vestibulo-cochlear branches of the labyrinthine artery could be the simplest explanation for vertigo with SSNHL. Audio-vestibular tests did not provide easy separation between ischaemic and non-ischaemic causes of vertigo with SSNHL.

  13. Probabilistic Graphical Models for the Analysis and Synthesis of Musical Audio

    Science.gov (United States)

    Hoffmann, Matthew Douglas

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

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

    Directory of Open Access Journals (Sweden)

    Avanthika Meenakshi

    2015-01-01

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

  15. 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

  16. 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.

  17. Can audio coached 4D CT emulate free breathing during the treatment course?

    International Nuclear Information System (INIS)

    Persson, Gitte F.; Nygaard, Ditte E.; Olsen, Mikael; Juhler-Noettrup, Trine; Pedersen, Anders N.; Specht, Lena; Korreman, Stine S.

    2008-01-01

    Background. The image quality of 4DCT depends on breathing regularity. Respiratory audio coaching may improve regularity and reduce motion artefacts. We question the safety of coached planning 4DCT without coaching during treatment. We investigated the possibility of coaching to a more stable breathing without changing the breathing amplitude. The interfraction variation of the breathing cycle amplitude in free and coached breathing was studied as well as the possible impact of fatigue on longer coaching sessions. Methods. Thirteen volunteers completed respiratory audio coaching on 3 days within a 2 week period. An external marker system monitoring the motion of the thoraco-abdominal wall was used to track the respiration. On all days, free breathing and two coached breathing curves were recorded. We assumed that free versus coached breathing from day 1 (reference session) simulated breathing during an uncoached versus coached planning 4DCT, respectively, and compared the mean breathing cycle amplitude to the free versus coached breathing from day 2 and 3 simulating free versus coached breathing during treatment. Results. For most volunteers it was impossible to apply coaching without changes in breathing cycle amplitude. No significant decrease in standard deviation of breathing cycle amplitude distribution was seen. Generally it was not possible to predict the breathing cycle amplitude and its variation the following days based on the breathing in the reference session irrespective of coaching or free breathing. We found a significant tendency towards an increased breathing cycle amplitude variation with the duration of the coaching session. Conclusion. These results suggest that large interfraction variation is present in breathing amplitude irrespective of coaching, leading to the suggestion of daily image guidance for verification of respiratory pattern and tumour related motion. Until further investigated it is not recommendable to use coached 4DCT for

  18. Can audio coached 4D CT emulate free breathing during the treatment course?

    Energy Technology Data Exchange (ETDEWEB)

    Persson, Gitte F.; Nygaard, Ditte E.; Olsen, Mikael; Juhler-Noettrup, Trine; Pedersen, Anders N.; Specht, Lena; Korreman, Stine S. (Dept. of Radiation Oncology, Rigshospitalet, Copenhagen (Denmark))

    2008-08-15

    Background. The image quality of 4DCT depends on breathing regularity. Respiratory audio coaching may improve regularity and reduce motion artefacts. We question the safety of coached planning 4DCT without coaching during treatment. We investigated the possibility of coaching to a more stable breathing without changing the breathing amplitude. The interfraction variation of the breathing cycle amplitude in free and coached breathing was studied as well as the possible impact of fatigue on longer coaching sessions. Methods. Thirteen volunteers completed respiratory audio coaching on 3 days within a 2 week period. An external marker system monitoring the motion of the thoraco-abdominal wall was used to track the respiration. On all days, free breathing and two coached breathing curves were recorded. We assumed that free versus coached breathing from day 1 (reference session) simulated breathing during an uncoached versus coached planning 4DCT, respectively, and compared the mean breathing cycle amplitude to the free versus coached breathing from day 2 and 3 simulating free versus coached breathing during treatment. Results. For most volunteers it was impossible to apply coaching without changes in breathing cycle amplitude. No significant decrease in standard deviation of breathing cycle amplitude distribution was seen. Generally it was not possible to predict the breathing cycle amplitude and its variation the following days based on the breathing in the reference session irrespective of coaching or free breathing. We found a significant tendency towards an increased breathing cycle amplitude variation with the duration of the coaching session. Conclusion. These results suggest that large interfraction variation is present in breathing amplitude irrespective of coaching, leading to the suggestion of daily image guidance for verification of respiratory pattern and tumour related motion. Until further investigated it is not recommendable to use coached 4DCT for

  19. Transmisión de audio usando redes Zigbee

    Directory of Open Access Journals (Sweden)

    David Delgado León

    2011-03-01

    Full Text Available Normal 0 21 false false false MicrosoftInternetExplorer4 /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Tabla normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-parent:""; mso-padding-alt:0cm 5.4pt 0cm 5.4pt; mso-para-margin:0cm; mso-para-margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:10.0pt; font-family:"Times New Roman"; mso-ansi-language:#0400; mso-fareast-language:#0400; mso-bidi-language:#0400;} Normal 0 21 false false false MicrosoftInternetExplorer4 /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Tabla normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-parent:""; mso-padding-alt:0cm 5.4pt 0cm 5.4pt; mso-para-margin:0cm; mso-para-margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:10.0pt; font-family:"Times New Roman"; mso-ansi-language:#0400; mso-fareast-language:#0400; mso-bidi-language:#0400;} Zigbee es un protocolo de comunicaciones basado en el estándar para redes inalámbricas IEEE_802.15.4. Concebido para el control y la monitorización de redes de sensores tanto en entornos industriales, médicos, como domóticos, ha existido un creciente interés por evaluarlo en aplicaciones de multimedia. Aun sin garantizar QoS (Quality of service por su limitado ancho de banda existen un conjunto de aplicaciones para vigilancia, grupos de rescate y salvamento, seguridad en entornos domóticos, grupos desplegados en un área limitada con necesidad de comunicación donde un sistema de audio y video en tiempo real de bajo costo basado en tecnología Zigbee es una idea sumamente atractiva.   Se presenta el diseño de un sistema que permita la comunicación de un grupo de usuarios desplegadas en un área limitada. Utiliza Microcontroladores RISC y tecnología Zigbee. Se investiga la factibilidad de usar la tecnología Zigbee para la transmisión de audio, se analizan

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

    Energy Technology Data Exchange (ETDEWEB)

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

    2005-07-01

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

  1. 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)

  2. 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...

  3. 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...

  4. 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...

  5. Créer des ressources audio pour le cours de FLE

    Directory of Open Access Journals (Sweden)

    Florence Gérard Lojacono

    2010-01-01

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

  6. Digitální audio zesilovač

    OpenAIRE

    Tiller, Jakub

    2010-01-01

    Tématem bakalářské práce jsou výkonové audio zesilovače pracující ve třídě D. Jejich velké rozšíření je způsobeno hlavně vysokou účinností a dobrými parametry. Tato práce je zaměřena na rozbor jednotlivých částí těchto zesilovačů a na rozbor možností měření jejich parametrů. Následně je v práci uveden návrh zesilovače jako laboratorního přípravku s možností číslicového řízení zesílení a navržena automatizovaná měření parametrů tohoto zesilovače v prostředí VEE Pro. Dále je v této práci navrže...

  7. 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 %.

  8. Audio-Visual Integration Modifies Emotional Judgment in Music

    Directory of Open Access Journals (Sweden)

    Shen-Yuan Su

    2011-10-01

    Full Text Available The conventional view that perceived emotion in music is derived mainly from auditory signals has led to neglect of the contribution of visual image. In this study, we manipulated mode (major vs. minor and examined the influence of a video image on emotional judgment in music. Melodies in either major or minor mode were controlled for tempo and rhythm and played to the participants. We found that Taiwanese participants, like Westerners, judged major melodies as expressing positive, and minor melodies negative, emotions. The major or minor melodies were then paired with video images of the singers, which were either emotionally congruent or incongruent with their modes. Results showed that participants perceived stronger positive or negative emotions with congruent audio-visual stimuli. Compared to listening to music alone, stronger emotions were perceived when an emotionally congruent video image was added and weaker emotions were perceived when an incongruent image was added. We therefore demonstrate that mode is important to perceive the emotional valence in music and that treating musical art as a purely auditory event might lose the enhanced emotional strength perceived in music, since going to a concert may lead to stronger perceived emotion than listening to the CD at home.

  9. Audio-visual aid in teaching "fatty liver".

    Science.gov (United States)

    Dash, Sambit; Kamath, Ullas; Rao, Guruprasad; Prakash, Jay; Mishra, Snigdha

    2016-05-06

    Use of audio visual tools to aid in medical education is ever on a rise. Our study intends to find the efficacy of a video prepared on "fatty liver," a topic that is often a challenge for pre-clinical teachers, in enhancing cognitive processing and ultimately learning. We prepared a video presentation of 11:36 min, incorporating various concepts of the topic, while keeping in view Mayer's and Ellaway guidelines for multimedia presentation. A pre-post test study on subject knowledge was conducted for 100 students with the video shown as intervention. A retrospective pre study was conducted as a survey which inquired about students understanding of the key concepts of the topic and a feedback on our video was taken. Students performed significantly better in the post test (mean score 8.52 vs. 5.45 in pre-test), positively responded in the retrospective pre-test and gave a positive feedback for our video presentation. Well-designed multimedia tools can aid in cognitive processing and enhance working memory capacity as shown in our study. In times when "smart" device penetration is high, information and communication tools in medical education, which can act as essential aid and not as replacement for traditional curriculums, can be beneficial to the students. © 2015 by The International Union of Biochemistry and Molecular Biology, 44:241-245, 2016. © 2015 The International Union of Biochemistry and Molecular Biology.

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

    Directory of Open Access Journals (Sweden)

    Soraya Ferreira Alves

    2016-12-01

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

  11. Do gender differences in audio-visual benefit and visual influence in audio-visual speech perception emerge with age?

    Directory of Open Access Journals (Sweden)

    Magnus eAlm

    2015-07-01

    Full Text Available Gender and age have been found to affect adults’ audio-visual (AV speech perception. However, research on adult aging focuses on adults over 60 years, who have an increasing likelihood for cognitive and sensory decline, which may confound positive effects of age-related AV-experience and its interaction with gender. Observed age and gender differences in AV speech perception may also depend on measurement sensitivity and AV task difficulty. Consequently both AV benefit and visual influence were used to measure visual contribution for gender-balanced groups of young (20-30 years and middle-aged adults (50-60 years with task difficulty varied using AV syllables from different talkers in alternative auditory backgrounds. Females had better speech-reading performance than males. Whereas no gender differences in AV benefit or visual influence were observed for young adults, visually influenced responses were significantly greater for middle-aged females than middle-aged males. That speech-reading performance did not influence AV benefit may be explained by visual speech extraction and AV integration constituting independent abilities. Contrastingly, the gender difference in visually influenced responses in middle adulthood may reflect an experience-related shift in females’ general AV perceptual strategy. Although young females’ speech-reading proficiency may not readily contribute to greater visual influence, between young and middle-adulthood recurrent confirmation of the contribution of visual cues induced by speech-reading proficiency may gradually shift females AV perceptual strategy towards more visually dominated responses.

  12. 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.

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

    Directory of Open Access Journals (Sweden)

    Koji Iwano

    2007-03-01

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

  14. Relative Effectiveness of Audio Tools for Fighter Pilots in Simulated Operational Flights: A Human Factors Approach

    National Research Council Canada - National Science Library

    Hourlier, Sylvain; Meehan, James; Leger, Alain; Roumes, Corinne

    2005-01-01

    .... Increasing use of audio has been suggested as a means to reduce visual workload, to enhance situation awareness, and mitigate the manual and cognitive demands of HOTAS and existing command-and-display concepts...

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

    Science.gov (United States)

    Le, Long N; Jones, Douglas L

    2018-03-01

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

  16. 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.

  17. 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...

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

    Energy Technology Data Exchange (ETDEWEB)

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

    2005-07-01

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

  19. Correspondence between audio and visual deep models for musical instrument detection in video recordings

    OpenAIRE

    Slizovskaia, Olga; Gómez, Emilia; Haro, Gloria

    2017-01-01

    This work aims at investigating cross-modal connections between audio and video sources in the task of musical instrument recognition. We also address in this work the understanding of the representations learned by convolutional neural networks (CNNs) and we study feature correspondence between audio and visual components of a multimodal CNN architecture. For each instrument category, we select the most activated neurons and investigate exist- ing cross-correlations between neurons from the ...

  20. Audio/visual analysis for high-speed TV advertisement detection from MPEG bitstream

    OpenAIRE

    Sadlier, David A.

    2002-01-01

    Advertisement breaks dunng or between television programmes are typically flagged by senes of black-and-silent video frames, which recurrendy occur in order to audio-visually separate individual advertisement spots from one another. It is the regular prevalence of these flags that enables automatic differentiauon between what is programme content and what is advertisement break. Detection of these audio-visual depressions within broadcast television content provides a basis on which advertise...

  1. The Success of Free to Play Games and Possibilities of Audio Monetization

    OpenAIRE

    Hahl, Kalle

    2014-01-01

    Video games are a huge business – nearly four times greater than film and music business combined. Free to play is the fastest growing category in video gaming. Game audio is part of the development of every game having a direct correlation between the growth of gaming industry and the growth of gaming audio industry. Games have inherently different goals for the players and the developers. Players are consumers seeking for entertainment. Developers are content producers trying to moneti...

  2. 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.

  3. 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

  4. 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.

  5. 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.

  6. 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.

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

    Directory of Open Access Journals (Sweden)

    Niels Böttcher

    2013-01-01

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

  8. Portable audio magnetotellurics - experimental measurements and joint inversion with radiomagnetotelluric data from Gotland, Sweden

    Science.gov (United States)

    Shan, Chunling; Kalscheuer, Thomas; Pedersen, Laust B.; Erlström, Mikael; Persson, Lena

    2017-08-01

    Field setup of an audio magnetotelluric (AMT) station is a very time consuming and heavy work load. In contrast, radio magnetotelluric (RMT) equipment is more portable and faster to deploy but has shallower investigation depth owing to its higher signal frequencies. To increase the efficiency in the acquisition of AMT data from 10 to 300 Hz, we introduce a modification of the AMT method, called portable audio magnetotellurics (PAMT), that uses a lighter AMT field system and (owing to the disregard of signals at frequencies of less than 10 Hz) shortened data acquisition time. PAMT uses three magnetometers pre-mounted on a rigid frame to measure magnetic fields and steel electrodes to measure electric fields. Field tests proved that the system is stable enough to measure AMT fields in the given frequency range. A PAMT test measurement was carried out on Gotland, Sweden along a 3.5 km profile to study the ground conductivity and to map shallow Silurian marlstone and limestone formations, deeper Silurian, Ordovician and Cambrian sedimentary structures and crystalline basement. RMT data collected along a coincident profile and regional airborne very low frequency (VLF) data support the interpretation of our PAMT data. While only the RMT and VLF data constrain a shallow ( 20-50 m deep) transition between Silurian conductive ( 1000 Ωm resistivity) limestone, the single-method inversion models of both the PAMT and the RMT data show a transition into a conductive layer of 3 to 30 Ωm resistivity at 80 m depth suggesting the compatibility of the two data sets. This conductive layer is interpreted as saltwater saturated succession of Silurian, Ordovician and Cambrian sedimentary units. Towards the lower boundary of this succession (at 600 m depth according to boreholes), only the PAMT data constrain the structure. As supported by modelling tests and sensitivity analysis, the PAMT data only contain a vague indication of the underlying crystalline basement. A PAMT and RMT

  9. 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.

  10. Passive Guaranteed Simulation of Analog Audio Circuits: A Port-Hamiltonian Approach

    Directory of Open Access Journals (Sweden)

    Antoine Falaize

    2016-09-01

    Full Text Available We present a method that generates passive-guaranteed stable simulations of analog audio circuits from electronic schematics for real-time issues. On one hand, this method is based on a continuous-time power-balanced state-space representation structured into its energy-storing parts, dissipative parts, and external sources. On the other hand, a numerical scheme is especially designed to preserve this structure and the power balance. These state-space structures define the class of port-Hamiltonian systems. The derivation of this structured system associated with the electronic circuit is achieved by an automated analysis of the interconnection network combined with a dictionary of models for each elementary component. The numerical scheme is based on the combination of finite differences applied on the state (with respect to the time variable and on the total energy (with respect to the state. This combination provides a discrete-time version of the power balance. This set of algorithms is valid for both the linear and nonlinear case. Finally, three applications of increasing complexities are given: a diode clipper, a common-emitter bipolar-junction transistor amplifier, and a wah pedal. The results are compared to offline simulations obtained from a popular circuit simulator.

  11. No, there is no 150 ms lead of visual speech on auditory speech, but a range of audiovisual asynchronies varying from small audio lead to large audio lag.

    Directory of Open Access Journals (Sweden)

    Jean-Luc Schwartz

    2014-07-01

    Full Text Available An increasing number of neuroscience papers capitalize on the assumption published in this journal that visual speech would be typically 150 ms ahead of auditory speech. It happens that the estimation of audiovisual asynchrony in the reference paper is valid only in very specific cases, for isolated consonant-vowel syllables or at the beginning of a speech utterance, in what we call "preparatory gestures". However, when syllables are chained in sequences, as they are typically in most parts of a natural speech utterance, asynchrony should be defined in a different way. This is what we call "comodulatory gestures" providing auditory and visual events more or less in synchrony. We provide audiovisual data on sequences of plosive-vowel syllables (pa, ta, ka, ba, da, ga, ma, na showing that audiovisual synchrony is actually rather precise, varying between 20 ms audio lead and 70 ms audio lag. We show how more complex speech material should result in a range typically varying between 40 ms audio lead and 200 ms audio lag, and we discuss how this natural coordination is reflected in the so-called temporal integration window for audiovisual speech perception. Finally we present a toy model of auditory and audiovisual predictive coding, showing that visual lead is actually not necessary for visual prediction.

  12. The Effects of Audio-Visual Recorded and Audio Recorded Listening Tasks on the Accuracy of Iranian EFL Learners' Oral Production

    Science.gov (United States)

    Drood, Pooya; Asl, Hanieh Davatgari

    2016-01-01

    The ways in which task in classrooms has developed and proceeded have receive great attention in the field of language teaching and learning in the sense that they draw attention of learners to the competing features such as accuracy, fluency, and complexity. English audiovisual and audio recorded materials have been widely used by teachers and…

  13. The Effect of Bio/Neurofeedback Training on Performance, Audio and Visual Attention in Elite Shooters

    Directory of Open Access Journals (Sweden)

    Farzaneh Bagheri asl

    2017-10-01

    Full Text Available The aim of this study was the effect of Bio/Neurofeedback training on performance, audio and visual attention of elite shooters. In this study 36 elite shooters of Kermanshah Province participated. They divided in three groups. Two groups were experimental groups how participated biofeedback and neurofeedback training and one group was control group. All participants were tried that their trainings as well as the number of shoots were closely controlled in order to assure their physical and special trainings. In this study, for attention affects the computerized Integrated Visual and Auditory test (IVA was used. This test has been considered as both a pretest and a posttest after the therapeutic intervention in three groups. The score of shooting also were collected before and after intervention. Each athlete in neurofeedback training group carried out the neurofeedback training for 20 sessions, each lasting 45 minutes. To do so, both auricles and T3 and PZ of each individual were cleaned using alcohol and new-perp gel to prepare for the neurofeedback training. The biofeedback training was heart rate and respiratory training.  To compare the results of the pretest and the posttest in each group, the dependent t-test was used. For compare three groups we used ANOVA test. The significance level was set at 0.05. The results indicated that there is a significant difference in three groups. It indicates a significant increase in the total score for attention after the implementation of the biofeedback and neurofeedback training. The results showed that the attention mean scores in three visual, audio, and total variables were higher in the posttest than in the pretest for two experimental groups. The results also indicated that the scores of the shoots were improved after training.  According the research finding, we can be said that the neurofeedback and biofeedback  training act on the waves of the sensory-motor beats and which are responsible

  14. 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.

  15. Multimodal interaction in the perception of impact events displayed via a multichannel audio and simulated structure-borne vibration

    Science.gov (United States)

    Martens, William L.; Woszczyk, Wieslaw

    2005-09-01

    For multimodal display systems in which realistic reproduction of impact events is desired, presenting structure-borne vibration along with multichannel audio recordings has been observed to create a greater sense of immersion in a virtual acoustic environment. Furthermore, there is an increased proportion of reports that the impact event took place within the observer's local area (this is termed ``presence with'' the event, in contrast to ``presence in'' the environment in which the event occurred). While holding the audio reproduction constant, varying the intermodal arrival time and level of mechanically displayed, synthetic whole-body vibration revealed a number of other subjective attributes that depend upon multimodal interaction in the perception of a representative impact event. For example, when the structure-borne component of the displayed impact event arrived 10 to 20 ms later than the airborne component, the intermodal delay was not only tolerated, but gave rise to an increase in the proportion of reports that the impact event had greater power. These results have enabled the refinement of a multimodal simulation in which the manipulation of synthetic whole-body vibration can be used to control perceptual attributes of impact events heard within an acoustic environment reproduced via a multichannel loudspeaker array.

  16. Real-time decreased sensitivity to an audio-visual illusion during goal-directed reaching.

    Directory of Open Access Journals (Sweden)

    Luc Tremblay

    Full Text Available In humans, sensory afferences are combined and integrated by the central nervous system (Ernst MO, Bülthoff HH (2004 Trends Cogn. Sci. 8: 162-169 and appear to provide a holistic representation of the environment. Empirical studies have repeatedly shown that vision dominates the other senses, especially for tasks with spatial demands. In contrast, it has also been observed that sound can strongly alter the perception of visual events. For example, when presented with 2 flashes and 1 beep in a very brief period of time, humans often report seeing 1 flash (i.e. fusion illusion, Andersen TS, Tiippana K, Sams M (2004 Brain Res. Cogn. Brain Res. 21: 301-308. However, it is not known how an unfolding movement modulates the contribution of vision to perception. Here, we used the audio-visual illusion to demonstrate that goal-directed movements can alter visual information processing in real-time. Specifically, the fusion illusion was linearly reduced as a function of limb velocity. These results suggest that cue combination and integration can be modulated in real-time by goal-directed behaviors; perhaps through sensory gating (Chapman CE, Beauchamp E (2006 J. Neurophysiol. 96: 1664-1675 and/or altered sensory noise (Ernst MO, Bülthoff HH (2004 Trends Cogn. Sci. 8: 162-169 during limb movements.

  17. Effects of Temporal Congruity Between Auditory and Visual Stimuli Using Rapid Audio-Visual Serial Presentation.

    Science.gov (United States)

    An, Xingwei; Tang, Jiabei; Liu, Shuang; He, Feng; Qi, Hongzhi; Wan, Baikun; Ming, Dong

    2016-10-01

    Combining visual and auditory stimuli in event-related potential (ERP)-based spellers gained more attention in recent years. Few of these studies notice the difference of ERP components and system efficiency caused by the shifting of visual and auditory onset. Here, we aim to study the effect of temporal congruity of auditory and visual stimuli onset on bimodal brain-computer interface (BCI) speller. We designed five visual and auditory combined paradigms with different visual-to-auditory delays (-33 to +100 ms). Eleven participants attended in this study. ERPs were acquired and aligned according to visual and auditory stimuli onset, respectively. ERPs of Fz, Cz, and PO7 channels were studied through the statistical analysis of different conditions both from visual-aligned ERPs and audio-aligned ERPs. Based on the visual-aligned ERPs, classification accuracy was also analyzed to seek the effects of visual-to-auditory delays. The latencies of ERP components depended mainly on the visual stimuli onset. Auditory stimuli onsets influenced mainly on early component accuracies, whereas visual stimuli onset determined later component accuracies. The latter, however, played a dominate role in overall classification. This study is important for further studies to achieve better explanations and ultimately determine the way to optimize the bimodal BCI application.

  18. Online dissection audio-visual resources for human anatomy: Undergraduate medical students' usage and learning outcomes.

    Science.gov (United States)

    Choi-Lundberg, Derek L; Cuellar, William A; Williams, Anne-Marie M

    2016-11-01

    In an attempt to improve undergraduate medical student preparation for and learning from dissection sessions, dissection audio-visual resources (DAVR) were developed. Data from e-learning management systems indicated DAVR were accessed by 28% ± 10 (mean ± SD for nine DAVR across three years) of students prior to the corresponding dissection sessions, representing at most 58% ± 20 of assigned dissectors. Approximately 50% of students accessed all available DAVR by the end of semester, while 10% accessed none. Ninety percent of survey respondents (response rate 58%) generally agreed that DAVR improved their preparation for and learning from dissection when used. Of several learning resources, only DAVR usage had a significant positive correlation (P = 0.002) with feeling prepared for dissection. Results on cadaveric anatomy practical examination questions in year 2 (Y2) and year 3 (Y3) cohorts were 3.9% (P learning outcomes of more students. Anat Sci Educ 9: 545-554. © 2016 American Association of Anatomists. © 2016 American Association of Anatomists.

  19. 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...

  20. Audio Classification in Speech and Music: A Comparison between a Statistical and a Neural Approach

    Directory of Open Access Journals (Sweden)

    Alessandro Bugatti

    2002-04-01

    Full Text Available We focus the attention on the problem of audio classification in speech and music for multimedia applications. In particular, we present a comparison between two different techniques for speech/music discrimination. The first method is based on Zero crossing rate and Bayesian classification. It is very simple from a computational point of view, and gives good results in case of pure music or speech. The simulation results show that some performance degradation arises when the music segment contains also some speech superimposed on music, or strong rhythmic components. To overcome these problems, we propose a second method, that uses more features, and is based on neural networks (specifically a multi-layer Perceptron. In this case we obtain better performance, at the expense of a limited growth in the computational complexity. In practice, the proposed neural network is simple to be implemented if a suitable polynomial is used as the activation function, and a real-time implementation is possible even if low-cost embedded systems are used.