WorldWideScience

Sample records for audio frequency

  1. 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…

  2. Audio Classification from Time-Frequency Texture

    OpenAIRE

    Yu, Guoshen; Slotine, Jean-Jacques

    2008-01-01

    Time-frequency representations of audio signals often resemble texture images. This paper derives a simple audio classification algorithm based on treating sound spectrograms as texture images. The algorithm is inspired by an earlier visual classification scheme particularly efficient at classifying textures. While solely based on time-frequency texture features, the algorithm achieves surprisingly good performance in musical instrument classification experiments.

  3. Calibration of an audio frequency noise generator

    DEFF Research Database (Denmark)

    Diamond, Joseph M.

    1966-01-01

    A noise generator of known output is very convenient in noise measurement. At low audio frequencies, however, all devices, including noise sources, may be affected by excess noise (1/f noise). It is therefore very desirable to be able to check the spectral density of a noise source before it is...... 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...

  4. 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.

  5. Experimental Investigation of Low Pressure Audio Frequency Discharge in Argon

    International Nuclear Information System (INIS)

    Experimental data obtained on audio frequency (100–10000 Hz) discharge in argon at four pressures 50, 60, 70, and 80 mTorr are presented. The data show significant changes of the discharge current waveform with frequency. These changes seem to be associated with the glow discharge profile and colour. An empirical model based on the assumption of a frequency-dependent breakdown voltage is used to describe the experimental data

  6. Audio Effects Based on Biorthogonal Time-Varying Frequency Warping

    Directory of Open Access Journals (Sweden)

    Cavaliere Sergio

    2001-01-01

    Full Text Available We illustrate the mathematical background and musical use of a class of audio effects based on frequency warping. These effects alter the frequency content of a signal via spectral mapping. They can be implemented in dispersive tapped delay lines based on a chain of all-pass filters. In a homogeneous line with first-order all-pass sections, the signal formed by the output samples at a given time is related to the input via the Laguerre transform. However, most musical signals require a time-varying frequency modification in order to be properly processed. Vibrato in musical instruments or voice intonation in the case of vocal sounds may be modeled as small and slow pitch variations. Simulation of these effects requires techniques for time-varying pitch and/or brightness modification that are very useful for sound processing. The basis for time-varying frequency warping is a time-varying version of the Laguerre transformation. The corresponding implementation structure is obtained as a dispersive tapped delay line, where each of the frequency dependent delay element has its own phase response. Thus, time-varying warping results in a space-varying, inhomogeneous, propagation structure. We show that time-varying frequency warping is associated to an expansion over biorthogonal sets generalizing the discrete Laguerre basis. Slow time-varying characteristics lead to slowly varying parameter sequences. The corresponding sound transformation does not suffer from discontinuities typical of delay lines based on unit delays.

  7. Implementation of virtual reality demonstrations with time-frequency-domain audio engine

    OpenAIRE

    Paasonen, Juhani

    2015-01-01

    Directional audio coding (DirAC) is a system originally developed for analyzing recordings of spatial audio and synthesizing them with arbitrary loudspeaker setups or headphones. It is based on knowledge of human hearing, psychoacoustics. In analysis, the input is divided to frequency bands. For each frequency band, the direction of arrival and the diffuseness of sound are estimated. In synthesis, the result is divided to diffuse and non-diffuse stream. The diffuse stream is non-directional, ...

  8. 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....... By choosing the optimal switching frequency based on the proposed analysis, the experimental results show that system power losses of the reference design are minimized and an efficiency improvement of 8 % in maximum is achieved without compromising audio performances....

  9. Method of measuring the amplitude directivity pattern of parabolic mirrored antennas in the audio frequency range

    Directory of Open Access Journals (Sweden)

    Sadchenko A. V.

    2016-02-01

    Full Text Available Directivity pattern (DP or graphical representation of the dependence of gain factor (directivity gain of antennas on the direction of the antenna in the target plane is the main characteristic that describes its directional properties. Running DP measurements directly in the microwave range is very expensive. While generating and receiving devices for the acoustic frequency range are reasonably priced. In this paper, we propose a method for measuring the amplitude directivity pattern of parabolic mirrored antennas on the basis of sound equivalent, which is based on the identity of the numerical values of the directivity gain of microwave range, and at audio frequencies. The paper presents analytical expressions for the calculation of equivalent frequency and defines the requirements for the minimum size of the antenna. The paper contains a modified block diagram for an amplitude directivity pattern meter for parabolic mirrored antennas in the audio frequency range.

  10. Audio-frequency heating of particulate magnetic systems

    Institute of Scientific and Technical Information of China (English)

    B.E. Kashevsky; I.V. Prokhorov; S.B. Kashevsky

    2007-01-01

    This paper presents theoretical and experimental studies on the magnetodynamics and energy dissipation in suspensions of small ferromagnetic particles with magnetic hysteresis and mechanical mobility in an AC magnetic field. Energy absorption by particles suspended in a solid, liquid or gas environment and subjected to high frequency magnetic fields is of great interest for cancer treatment by hyperthermia, chemical technology,biotechnology and smart materials science.Sub-micron needle-like γ-Fe2O3 particles dispersed in liquid were subjected in this study to a 430 Hz magnetic field with an intensity of up to 105 A/m. Dynamic magnetization loops were measured in parallel to the energy dissipated in the samples. Combined magnetomechanical dynamics of particle dispersions was simulated by using a chain-of-spheres model allowing for incoherent magnetic field reversal. In liquid dispersions,within the kilohertz frequency range, the mechanical mobility of particles does not interfere with their hysteretic magnetic reversal that makes heat release comparable to that observed with solids; for instance, in the present study using γ-Fe2O3 particles in liquid subjected to 104 Hz field exhibited heat release rates from 250 up to 600 W per 1 cm3 of the dry particle content.

  11. Arabic Audio News Retrieval System Using Dependent Speaker Mode, Mel Frequency Cepstral Coefficient and Dynamic Time Warping Techniques

    Directory of Open Access Journals (Sweden)

    Hasan Muaidi

    2014-06-01

    Full Text Available Recently, audio data has increasingly becomes one of the prevalent source of information, especially after the exponential growth of using Internet, digital libraries systems and digital mobile devices. The currently massive amount of audio data stimulates working on developing custom audio retrieval tools to facilitate the audio retrieval tasks. The most familiar audio retrieval systems are based on searching using keyword, title or authors. This study presents the feasibility of using MEL Frequency Cepstral Coefficients (MFCCs to extract features and Dynamic Time Warping (DTW to compare the test patterns for Arabic audio news. The study proposes and implements architecture for content based audio retrieval system that is dedicated for the Arabic Audio News. The proposed architecture (ARANEWS utilizes automatic speech recognition for isolated Arabic keyword speech mode; template based automatic speech recognition approach, MFCCs and DTW. ARANEWS presents a style of retrieval system that based on modeling signal waves and measuring the similarity between features that are extracted from spoken queries and spoken keywords. One of the major components that compose ARANEWS system is feature Database (ARANEWSDB. ARANEWSDB stores the extracted features (MFCCs from the spoken keywords that are prepared to retrieve Arabic audio news. ARANEWS supports using Query by Humming (QBH and Query by Example (QBE instead of using query by text.

  12. Audio-frequency noise emissions from high-voltage overhead power lines

    International Nuclear Information System (INIS)

    This article discusses the noise-emissions caused by high-voltage overhead power lines that can occur under certain atmospheric conditions. These emissions, caused by electric discharges around the conductors, can achieve disturbing values, depending on the conditions prevailing at the time in question. The causes of the discharges are examined and the ionisation processes involved are looked at. The parameters influencing the discharges are discussed and measures that can be taken to reduce such audio-frequency emissions are looked at. The authors note that a reduction of peripheral field strengths can reduce emissions and that hydrophilic coatings can lead to faster reduction of such effects after rainfall

  13. Audio-Band Frequency-Dependent Squeezing for Gravitational-Wave Detectors

    Science.gov (United States)

    Oelker, Eric; Isogai, Tomoki; Miller, John; Tse, Maggie; Barsotti, Lisa; Mavalvala, Nergis; Evans, Matthew

    2016-01-01

    Quantum vacuum fluctuations impose strict limits on precision displacement measurements, those of interferometric gravitational-wave detectors among them. Introducing squeezed states into an interferometer's readout port can improve the sensitivity of the instrument, leading to richer astrophysical observations. However, optomechanical interactions dictate that the vacuum's squeezed quadrature must rotate by 90° around 50 Hz. Here we use a 2-m-long, high-finesse optical resonator to produce frequency-dependent rotation around 1.2 kHz. This demonstration of audio-band frequency-dependent squeezing uses technology and methods that are scalable to the required rotation frequency and validates previously developed theoretical models, heralding application of the technique in future gravitational-wave detectors.

  14. PERFORMANCE ANALYSIS OF GATED RING OSCILLATOR DESIGNED FOR AUDIO FREQUENCY RANGE ASYNCHRONOUS ADC

    Directory of Open Access Journals (Sweden)

    Anita Arvind Deshmukh

    2014-12-01

    Full Text Available This paper presents performance analysis of Gated Ring Oscillator (GRO. Proposed GRO is designed to employ in implementation of Time to Digital Converter (TDC block of Asynchronous ADC. For an audio frequency range ADC, minimum GRO stages are designed using asynchronous technique. So leads to reduced area and power. Compared to conventional Ring Oscillator (RO, we avoided to employ the gated clock; to evade clock design related problems like jitter, additional area and power. Instead we preferred gating of ring oscillator itself. Consequently during sleep mode, GRO disables automatically which saves the dynamic power. Furthermore it also provides first order noise shaping of the quantization and mismatch noise. Proposed GRO is implemented with 0.18µm CMOS Digital Technology in Cadence Virtuso environment. GRO performance analysis shows oscillation frequency as 286 KHz with 327ps jitter and average power consumption of 1.08µW.

  15. Comparison of level discrimination, increment detection, and comodulation masking release in the audio- and envelope-frequency domains

    DEFF Research Database (Denmark)

    Nelson, Paul C.; Ewert, Stephan; Carney, Laurel H.; Dau, Torsten

    2007-01-01

    In general, the temporal structure of stimuli must be considered to account for certain observations made in detection and masking experiments in the audio-frequency domain. Two such phenomena are (1) a heightened sensitivity to amplitude increments with a temporal fringe compared to gated level ...

  16. Comparison of intensity discrimination, increment detection, and comodulation masking release in the envelope and audio-frequency domains

    DEFF Research Database (Denmark)

    Nelson, Paul C.; Ewert, Stephan; Carney, Laurel H.; Dau, Torsten

    In the audio-frequency domain, the envelope apparently plays an important role in detection of intensity increments and in comodulation masking release (CMR). The current study addressed the question whether the second-order envelope ("venelope") contributes similarly for comparable experiments i...

  17. Whistlers and audio-frequency emissions monthly summaries of whistlers and emissions for the period July 1957 - December 1958

    CERN Document Server

    Morgan, M G

    1965-01-01

    Annals of the International Geophysical Year, Volume 37: Whistlers and Audio-Frequency Emissions presents the principal results obtained in Whistlers-East synoptic program publications. Although whistlers can be observed at any time of day, it is found that they occur primarily at night. The greatest incidence of whistlers during the International Geophysical Year (IGY) period occurred in both hemispheres in the geomagnetic latitude range 50-60ʻ. The day-to-day correlation of whistler activity at geomagnetically conjugate stations was sometimes very low and sometimes remarkably high. This book

  18. Movie Piracy Detection Based on Audio Features Using Mel-Frequency Cepstral Coefficients and Vector Quantization

    OpenAIRE

    Srinivas, B.; K.Venkata Rao; P. Suresh Varma

    2012-01-01

    Along with the increase in the advancement oftechnology in movie industry over internet, there is also anincrease in the movie piracy via internet which affects factors likeeconomy and repudiation of movie industry. Internet movie piracyis the most common means for pirates as well as downloader’s tobreak copyright laws by anonymous illegal uploads/downloads. Inthis paper we proposed an automated internet movie piracydetection mechanism based on audio fingerprint, whichimplements two famous al...

  19. Audio 2008: Audio Fixation

    Science.gov (United States)

    Kaye, Alan L.

    2008-01-01

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

  20. Systematic design of output filters for audio class-D amplifiers via Simplified Real Frequency Technique

    Science.gov (United States)

    Hintzen, E.; Vennemann, T.; Mathis, W.

    2014-11-01

    In this paper a new filter design concept is proposed and implemented which takes into account the complex loudspeaker impedance. By means of techniques of broadband matching, that has been successfully applied in radio technology, we are able to optimize the reconstruction filter to achieve an overall linear frequency response. Here, a passive filter network is inserted between source and load that matches the complex load impedance to the complex source impedance within a desired frequency range. The design and calculation of the filter is usually done using numerical approximation methods which are known as Real Frequency Techniques (RFT). A first approach to systematic design of reconstruction filters for class-D amplifiers is proposed, using the Simplified Real Frequency Technique (SRFT). Some fundamental considerations are introduced as well as the benefits and challenges of impedance matching between class-D amplifiers and loudspeakers. Current simulation data using MATLAB is presented and supports some first conclusions.

  1. Observation of Edge Harmonic Oscillation in NSTX and Theoretical Study of its Active Control Using HHFW Antenna at Audio Frequencies

    International Nuclear Information System (INIS)

    Full text: Edge localized modes (ELMs) can generate unacceptable heat loads to plasma facing components in a reactor scale tokamak or spherical torus, and therefore ELM control is a critical issue in ITER. ELM control using non-axisymmetric (3D) fields is a promising concept, but the 3D coil requirements are demanding in cost and engineering. An alternative means may be to use internally driven 3D field oscillations such as edge harmonic oscillations (EHOs), but the relevant operational window is possibly more limited than the external 3D field applications. The disadvantages of each approach can be mitigated if the external and the internal drive of 3D fields can be constructively combined. This paper presents two important topics for this vision: Experimental observations of edge harmonic oscillations in NSTX (not necessarily the same as EHOs in DIII-D), and theoretical study of its audio-frequency drive using the high harmonic fast wave (HHFW) antenna as 3D field coils. Edge harmonic oscillations were observed particularly well in NSTX ELM-free operation with low n = 1 core modes, with various diagnostics confirming n = 4 - 6 coherent oscillations in 2 - 8 kHz frequency range. These oscillations, which share some characteristics with the n = 1 dominated modes observed in small-ELM regimes in NSTX, seem to have a favored operational window in rotational shear, similarly to EHOs in DIII-D QH modes. However, in NSTX, they are not observed to provide particle or impurity control, possibly due to their weak amplitudes, of a few mm displacements, as measured by reflectometry. The external drive of these modes has been proposed in NSTX, by utilizing audio-frequency currents in the HHFW antenna straps. Analysis shows that the HHFW straps can be optimized to maximize n = 4 - 6 while minimizing n = 1 - 3. Also, IPEC calculations show that the optimized configuration with only 1 kAt current can produce twice larger displacements than the observed internal modes, ∼ 6 mm

  2. Magnetic Force Nanoprobe for Direct Observation of Audio Frequency Tonotopy of Hair Cells.

    Science.gov (United States)

    Kim, Ji-Wook; Lee, Jae-Hyun; Ma, Ji-Hyun; Chung, Eunna; Choi, Hongsuh; Bok, Jinwoong; Cheon, Jinwoo

    2016-06-01

    Sound perception via mechano-sensation is a remarkably sensitive and fast transmission process, converting sound as a mechanical input to neural signals in a living organism. Although knowledge of auditory hair cell functions has advanced over the past decades, challenges remain in understanding their biomechanics, partly because of their biophysical complexity and the lack of appropriate probing tools. Most current studies of hair cells have been conducted in a relatively low-frequency range (perception of 20 kHz or higher. Here, we demonstrate that the magnetic force nanoprobe (MFN) has superb spatiotemporal capabilities to mechanically stimulate spatially-targeted individual hair cells with a temporal resolution of up to 9 μs, which is equivalent to approximately 50 kHz; therefore, it is possible to investigate avian hair cell biomechanics at different tonotopic regions of the cochlea covering a full hearing frequency range of 50 to 5000 Hz. We found that the variation of the stimulation frequency and amplitude of hair bundles creates distinct mechanical responsive features along the tonotopic axis, where the kinetics of the hair bundle recovery motion exhibits unique frequency-dependent characteristics: basal, middle, and apical hair bundles can effectively respond at their respective ranges of frequency. We revealed that such recovery kinetics possesses two different time constants that are closely related to the passive and active motilities of hair cells. The use of MFN is critical for the kinetics study of free-standing hair cells in a spatiotemporally distinct tonotopic organization. PMID:27215487

  3. 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.

  4. 穴位按压配合音频治疗手术后尿潴留%Audio frequency and acupuncture point pressing to treat retention of urine

    Institute of Scientific and Technical Information of China (English)

    2001-01-01

    @@Background: Retention of urine is a command complication for the postoperative patients who recepte the general or vertebral canal anesthesia.Because the micturition reflex center is temporarily disturbed by the anesthetic , the vegetative nerve system that control the bladder is functional disorder.The urinary bladder sphincter relatively contracts,and the detrusor urinae of bladder relatively relax. Objective: To discuss the effect of audio frequency and acupuncture point pressing to treat retention of urine. Unit: General Hospital of Shenyang Military Region.

  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. On the Use of Time–Frequency Reassignment and SVM-Based Classifier for Audio Surveillance Applications

    Directory of Open Access Journals (Sweden)

    Souli S. Sameh

    2014-11-01

    Full Text Available In this paper, we propose a robust environmental sound spectrogram classification approach. Its purpose is surveillance and security applications based on the reassignment method and log-Gabor filters. Besides, the reassignment method is applied to the spectrogram to improve the readability of the time-frequency representation, and to assure a better localization of the signal components. Our approach includes three methods. In the first two methods, the reassigned spectrograms are passed through appropriate log-Gabor filter banks and the outputs are averaged and underwent an optimal feature selection procedure based on a mutual information criterion. The third method uses the same steps but applied only to three patches extracted from each reassigned spectrogram. The proposed approach is tested on a large database consists of 1000 sounds belonging to ten classes. The recognition is based on Multiclass Support Vector Machines.

  7. 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 ...

  8. Instant audio processing with Web Audio

    CERN Document Server

    Khoo, Chris

    2013-01-01

    Filled with practical, step-by-step instructions and clear explanations for the most important and useful tasks. A concise, recipe-based approach to use Web Audio's automation functionality to produce interesting audio effects such as audio stitching and ducking.This book is designed for developers with some HTML and JavaScript programming experience who are seeking to learn about Web Audio. Experience with AJAX and web server installation/configuration is a plus but is not a necessity in order to follow the content of the book.

  9. Enhancing Manual Scan Registration Using Audio Cues

    Science.gov (United States)

    Ntsoko, T.; Sithole, G.

    2014-04-01

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

  10. 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.

  11. 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

  12. 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 modality...... short trajectories are constructed to rep- resent the motion of players. From these, four motion fea- tures are extracted and combined directly with audio fea- tures for classification. A k-nearest neighbour classifier is applied for classification of 180 1-minute video sequences from three sports types...

  13. Audio Source Separation Using a Deep Autoencoder

    OpenAIRE

    Jang, Giljin; Kim, Han-Gyu; Oh, Yung-Hwan

    2014-01-01

    This paper proposes a novel framework for unsupervised audio source separation using a deep autoencoder. The characteristics of unknown source signals mixed in the mixed input is automatically by properly configured autoencoders implemented by a network with many layers, and separated by clustering the coefficient vectors in the code layer. By investigating the weight vectors to the final target, representation layer, the primitive components of the audio signals in the frequency domain are o...

  14. Principles of Audio Watermarking

    Directory of Open Access Journals (Sweden)

    Martin Hrncar

    2008-01-01

    Full Text Available The article contains a brief overview of modern methods for embedding additional data in audio signals. It could have many reasons - for the purposes of access control or identification related to particular type of audio. This secret information is not “visible” for a user. This concept utilizes the imperfection of human auditory system. Simple data hiding into audio file has been proved in MATLAB.

  15. Categorizing Video Game Audio

    DEFF Research Database (Denmark)

    Westerberg, Andreas Rytter; Schoenau-Fog, Henrik

    2015-01-01

    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...... 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....

  16. 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.

  17. 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.

  18. Portable Audio Design

    DEFF Research Database (Denmark)

    Groth, Sanne Krogh

    2014-01-01

    The chapter presents a methodological approach to the early process of producing portable audio design. The chapter high lights audio walks and audio guides, but can also be of inspiration when working with graphical and video production for portable devices. The final products can be presented...... within online and physical institutional contexts. The approach focuses especially on the relationship to specific sites, and how an awareness of the relationship between the site and the production can be part of the design process. Such awareness entails several approaches: the necessity of paying...

  19. 音乐与蟋蟀鸣声的混合声频对食用菌生长的影响%Influence of audio frequency mixing of music and cricket voice on growth of edible mushrooms

    Institute of Scientific and Technical Information of China (English)

    姜仕仁; 黄俊; 韩省华; 曾宪霖

    2011-01-01

    为了考察声波对食用菌生长、产量及营养成分等方面的影响,采用自行开发的声频设备,播放古典音乐与蟋蟀鸣声混合而成的声频,对茶树菇、高温姬菇、黑平、杏鲍菇、秀珍菇、小白菇等6种食用菌的菌丝体进行了7次声波助长试验,对姬菇、黑平和姬菇18号3种食用菌的子实体进行了4次试验.结果表明,此声频可使食用菌的菌丝体生长速度加快10.2%~21%;使子实体提早出菇,提前l~5 d采菇,并可延长采菇天数;4次子实体试验的产量分别增加了15.76%、13.38%、13.05%和7.95%.经对2种子实体成分检测比较表明,姬菇18号的脂肪、蛋白质和粗多糖质量分数分别增加5.88%、8.74%和2.78%,黑平的蛋白质、粗多糖的质量分数分别提高2.37%和43.27%.研究结果为声波助长技术在食用菌生产上的推广应用提供科学依据.%In order to investigate audio frequency influence on the growth, yield and nutrient component of edible mushroom, the audio stimulating technology was applied to the mycelium of six kinds of edible mushroom (Agrocybe Cylindracea, high-temperature Pleurotu corucopiae, Pleurotus ap., Pleurotus eryngii, Pleurotu cornucopiae and Pleurocybella poprrigens) and the fruiting body of three edible mushrooms (Pleurotu corucopiae, Pleurotus ap. And Pleurotu corucopiae 18). The audio was generated by mixing classical music and cricket voice with a self-developed audio player equipment. The results showed that the sound increased the mycelium growth of all the six mushrooms by 10.2%~21%, accelerated their fruiting, advanced the body fruiting harvest time by 1-5 days and extended the picking period by about 3-8 days. The audio treatment also increased the yields of edible mushrooms by 15.76%, 13.38%, 13.05% and 7.95% in four tests respectively. By comparison of fruiting body nutrient component, the mass fraction of fat, protein and polysaccharide of Pleurotu corucopiae 18

  20. A Novel Algorithm for Robust Audio Watermarking in Wavelet Domain

    Institute of Scientific and Technical Information of China (English)

    FU Yu; WANG Bao-bao; LI Chun-ru; QUAN Ning-qiang

    2004-01-01

    A novel algorithm for digital audio watermarking in wavelet domain is proposed. First,an original audio signal is decomposed by discrete wavelet transform at three levels. Then, a discrete watermark is embedded into the coefficients of its intermediate frequencies. Finally, the watermarked audio signal is obtained by wavelet reconstruction. The proposed algorithm makes good use of the multiresolution characteristics of wavelet transform. The original audio signal is not needed when detecting the watermark correlatively. Simulation results show that the algorithm is inaudible and robust to noise, filtering and resampling.

  1. Introduction to AVS Audio

    Institute of Scientific and Technical Information of China (English)

    Hao-Jun Ai; Shui-Xian Chen; Rui-Min Hu

    2006-01-01

    This paper describes a general audio coding algorithm which has been recently standardized by AVS, China.The algorithm is based on a perceptual coding technique. The codec delivers near CD-quality audio at 128kb/s. This paper describes the coder structure in detail and discusses the reasons for specific design methods. A summary of the subjective test results are presented for the prototype codec. Comparison Mean Opinion Score (CMOS) test indicates that the quality of the AVS audio coder is comparable with MPEG Layer-3 audio coder. A real-time decoder was used for the characterization test,which is based on a 16-bit fixed-point DSP. The performance of the DSP solution was demonstrated, including computational complexity and storage characteristics.

  2. Forensic audio watermark detection

    Science.gov (United States)

    Steinebach, Martin; Zmudzinski, Sascha; Petrautzki, Dirk

    2012-03-01

    Digital audio watermarking detection is often computational complex and requires at least as much audio information as required to embed a complete watermark. In some applications, especially real-time monitoring, this is an important drawback. The reason for this is the usage of sync sequences at the beginning of the watermark, allowing a decision about the presence only if at least the sync has been found and retrieved. We propose an alternative method for detecting the presence of a watermark. Based on the knowledge of the secret key used for embedding, we create a mark for all potential marking stages and then use a sliding window to test a given audio file on the presence of statistical characteristics caused by embedding. In this way we can detect a watermark in less than 1 second of audio.

  3. Structure Learning in Audio

    OpenAIRE

    Nielsen, Andreas Brinch; Hansen, Lars Kai

    2009-01-01

    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 using pitch dynamics is suggested. The other approach is finding structures between the mixings of multiple sources based on an assumption of statistical independence of the sources. Three different aud...

  4. Spatial domain entertainment audio decompression/compression

    Science.gov (United States)

    Chan, Y. K.; Tam, Ka Him K.

    2014-02-01

    The ARM7 NEON processor with 128bit SIMD hardware accelerator requires a peak performance of 13.99 Mega Cycles per Second for MP3 stereo entertainment quality decoding. For similar compression bit rate, OGG and AAC is preferred over MP3. The Patent Cooperation Treaty Application dated 28/August/2012 describes an audio decompression scheme producing a sequence of interleaving "min to Max" and "Max to min" rising and falling segments. The number of interior audio samples bound by "min to Max" or "Max to min" can be {0|1|…|N} audio samples. The magnitudes of samples, including the bounding min and Max, are distributed as normalized constants within the 0 and 1 of the bounding magnitudes. The decompressed audio is then a "sequence of static segments" on a frame by frame basis. Some of these frames needed to be post processed to elevate high frequency. The post processing is compression efficiency neutral and the additional decoding complexity is only a small fraction of the overall decoding complexity without the need of extra hardware. Compression efficiency can be speculated as very high as source audio had been decimated and converted to a set of data with only "segment length and corresponding segment magnitude" attributes. The PCT describes how these two attributes are efficiently coded by the PCT innovative coding scheme. The PCT decoding efficiency is obviously very high and decoding latency is basically zero. Both hardware requirement and run time is at least an order of magnitude better than MP3 variants. The side benefit is ultra low power consumption on mobile device. The acid test on how such a simplistic waveform representation can indeed reproduce authentic decompressed quality is benchmarked versus OGG(aoTuv Beta 6.03) by three pair of stereo audio frames and one broadcast like voice audio frame with each frame consisting 2,028 samples at 44,100KHz sampling frequency.

  5. Multimodal audio guide for museums and exhibitions

    Science.gov (United States)

    Gebbensleben, Sandra; Dittmann, Jana; Vielhauer, Claus

    2006-02-01

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

  6. A content-based digital audio watermarking algorithm

    Science.gov (United States)

    Zhang, Liping; Zhao, Yi; Xu, Wen Li

    2015-12-01

    Digital audio watermarking embeds inaudible information into digital audio data for the purposes of copyright protection, ownership verification, covert communication, and/or auxiliary data carrying. In this paper, we present a novel watermarking scheme to embed a meaningful gray image into digital audio by quantizing the wavelet coefficients (using integer lifting wavelet transform) of audio samples. Our audio-dependent watermarking procedure directly exploits temporal and frequency perceptual masking of the human auditory system (HAS) to guarantee that the embedded watermark image is inaudible and robust. The watermark is constructed by utilizing still image compression technique, breaking each audio clip into smaller segments, selecting the perceptually significant audio segments to wavelet transform, and quantizing the perceptually significant wavelet coefficients. The proposed watermarking algorithm can extract the watermark image without the help from the original digital audio signals. We also demonstrate the robustness of that watermarking procedure to audio degradations and distortions, e.g., those that result from noise adding, MPEG compression, low pass filtering, resampling, and requantization.

  7. 应用可控源音频大地电磁法的土壤电阻率测量%Analysis on Soil Resistivity Measurement Based on Controlled Source Audio-frequency Magneto-telluric

    Institute of Scientific and Technical Information of China (English)

    苏杰; 吴广宁; 曹晓斌; 马御棠; 李瑞芳; 周炜明

    2011-01-01

    Grounding grid that is designed based on accurately measured soil resistivity can make the design error minimum, and it is significant to reliable operation of power system. Controlled source audio-frequency magneto-telluric (CSAMT) is a geophysical technology that has been successfully applied to explore the resources such as coal,petroleum and natural gas. Based on basic priciple of CSAMT method and Schlumberger measurement method a soil resistivity measurement model is built. Under the condition of homogeneous half-space, by use of famout grounding analyzing software CDEGS the simulative measurement of soil resistivity is implemented. Measured results show that when the measured soil resistance basiclly remains constant in a wide frequency range and is close to actual soil resistivity, it indicates that the measured soil is uniform; however the error of results measured by Schlumberger method is larger than actual soil resistivity, in addition, the measured results by Schlumberger method under higher frequencies are not receivable at all, and this fact confirms that CSAMT method is an accurate method for the measurement of soil resistivity.%采用准确测量土壤电阻率设计的地网可使设计误差达到最小,对电力系统的可靠运行具有重要意义.可控源音频大地电磁法(controlled source audio-frequency magnetotelluric,CSAMT)是一种成功应用于煤、油、气等资源探测的物探技术.基于CSAMT法及Schlumberger法基本原理建立了测量土壤电阻率的模型,在均匀半空间条件下,利用接地分析软件CDEGS实现了土壤电阻率的仿真测量.结果表明,CSAMT法所测土壤电阻率在很宽一段频率范围内基本保持不变,且与真实土壤电阻率的误差很小,表明所测土壤为均匀土壤;而Schlumberger法所测结果与真实值误差较大,且在频率较高时所得结果完全不可信,证实了CSAMT 法测量土壤电阻率的高准确性.

  8. Reception of infrasound and audio current in derma nerves

    Institute of Scientific and Technical Information of China (English)

    Jianwen Li; Ziyu Li; Xuezong Ma

    2010-01-01

    Determining the frequency range of derma nerve that responds to audio current is fundamental for the development of skin-hearing technology.Previous studies have shown that the range of derma nerve responding to audio current is 15-15 000 Hz,because audio amplification is not separated from the step-up transformer.Therefore,the present study used a signal generator which directly drives plane electrodes,simplified the original experimental environment for skin-hearing,measured lower limit voltage of frequency for derma nerve receiving pulse current signals,and revealed that the frequency range of human derma nerve response was as wide as 0.1-30 000 Hz.Results demonstrate that human derma nerve receives audio signals and infrasound within a wide frequency range.

  9. Virtual Audio - Three-Dimensional Audio in Virtual Environments

    OpenAIRE

    Adler, Daniel

    1996-01-01

    Three-dimensional interactive audio has a variety ofpotential uses in human-machine interfaces. After lagging seriously behind the visual components, the importance of sound is now becoming increas-ingly accepted. This paper mainly discusses background and techniques to implement three-dimensional audio in computer interfaces. A case study of a system for three-dimensional audio, implemented by the author, is described in great detail. The audio system was moreover integra...

  10. Structure Learning in Audio

    DEFF Research Database (Denmark)

    Nielsen, Andreas Brinch

    using pitch dynamics is suggested. The other approach is finding structures between the mixings of multiple sources based on an assumption of statistical independence of the sources. Three different audio classification tasks have been investigated. Audio classification into three classes, music, noise...... and speech, using novel features based on pitch dynamics. Within instrument classification two different harmonic models have been compared. Finally voiced/unvoiced segmentation of popular music is done based on MFCC’s and AR coefficients. The structures in the mixings of multiple sources have been...

  11. 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

  12. 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…

  13. Efectos digitales de audio con Web Audio API

    OpenAIRE

    GARCÍA CHAPARRO, SAMUEL

    2015-01-01

    El presente trabajo consiste en un estudio de la capacidad de Web Audio API para el procesado de efectos de audio en tiempo real. De todos los efectos de audio posibles se han elegido el wah-wah, el flanger y el choris, efectos ampliamente empleados con guitarra eléctrica. Se crean funciones de lenguaje JavaScript que modelan el comportamiento de los efectos de audio elegidos, haciéndolas funcionar sobre una plataforma web HTML5. García Chaparro, S. (2015). Efectos digitales de audio con W...

  14. 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

  15. Portable audio electronics for impedance-based measurements in microfluidics

    International Nuclear Information System (INIS)

    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)

  16. Active absorber for damping audio-frequency noise emissions at wind power plants. Active vibration damping; Aktiver Tilger zur Unterdrueckung tonaler Schallemissionen an Windenergieanlagen. Aktive Schwingungstilgung

    Energy Technology Data Exchange (ETDEWEB)

    Neugebauer, R.; Linke, M.; Kunze, H.; Ullrich, M. [Fraunhofer Institut fuer Werkzeugmaschinen und Umformtechnik (IWU), Chemnitz / Dresden (Germany)

    2010-07-01

    Structural vibrations of a wind turbine's drive train are one of the main reasons for noise emissions. Mechanical vibrations are transferred through the structure and emitted as noise by large surfaces, e.g. tower and nacelle. Dominant vibration excitation is caused for example by the gear mesh. If the gear mesh frequency is coinciding with the frequency of a structural resonance, the emitted noise contains noticeable single tones. German immission control law requires a ''tonal penalty'' up to 6 dB, if the emitted noise contains annoying tones. To ensure compliance with immission limits those tones must be reduced or eliminated. For wind turbines running with variable speed an active vibration absorber has been developed, whose absorber frequency and damping is adapted corresponding to the alternating vibration excitation. (orig.)

  17. 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.

  18. A Single Core Hardware Approach of MPEG Audio Decoder for Real-Time Transmission

    Directory of Open Access Journals (Sweden)

    M.B.I. Reaz

    2012-04-01

    Full Text Available The decoding of the voice audio bit stream is an issue in terms of real-time transmission of high quality voice audio over the Internet. A stand-alone chip to perform decoding is a better solution over software approach. The MPEG audio compression provides high compression with minimal loss. This study describes a VHDL model of MPEG audio layer 1 decoder that perform concurrent processing while receiving voice quality audio input bit stream at a constant bit rate and simultaneously producing a stream of 8-bit monopole PCM samples at a constant sampling frequency in real time.

  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......-decision scheme. The Mel-Frequency Cepstral Coefficients and the vertical mouth opening are the chosen audio and visual features respectively, both augmented with their first-order derivatives. The proposed system is assessed using far-field recordings from four different speakers and under various levels of...

  20. Digital Audio Watermarking: An Overview

    OpenAIRE

    Bhuvnesh Kumar Singh; Alok Kumar Singh

    2013-01-01

    Digital watermarking is a very recent research area. Digital audio watermarking is a method to embed or hide the Watermark (Information signal) into a digital signal i.e. Image, audio, text or video data. The watermark is difficult to remove from the audio signal. If the signal is copied, the information or watermark is also carried in the copy. A signal may carry several different watermarks at the same time. It is used to protecting multimedia data from unauthorized copying, piracy, ownersh...

  1. Parametric Coding of Stereo Audio

    Directory of Open Access Journals (Sweden)

    Erik Schuijers

    2005-06-01

    Full Text Available Parametric-stereo coding is a technique to efficiently code a stereo audio signal as a monaural signal plus a small amount of parametric overhead to describe the stereo image. The stereo properties are analyzed, encoded, and reinstated in a decoder according to spatial psychoacoustical principles. The monaural signal can be encoded using any (conventional audio coder. Experiments show that the parameterized description of spatial properties enables a highly efficient, high-quality stereo audio representation.

  2. 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

  3. Efficient audio signal processing for embedded systems

    Science.gov (United States)

    Chiu, Leung Kin

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

  4. 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)

  5. 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...

  6. Audio Watermarking with Error Correction

    Directory of Open Access Journals (Sweden)

    Aman Chadha

    2011-09-01

    Full Text Available In recent times, communication through the internet has tremendously facilitated the distribution of multimedia data. Although this is indubitably a boon, one of its repercussions is that it has also given impetus to the notorious issue of online music piracy. Unethical attempts can also be made to deliberately alter such copyrighted data and thus, misuse it. Copyright violation by means of unauthorized distribution, as well as unauthorized tampering of copyrighted audio data is an important technological and research issue. Audio watermarking has been proposed as a solution to tackle this issue. The main purpose of audio watermarking is to protect against possible threats to the audio data and in case of copyright violation or unauthorized tampering, authenticity of such data can be disputed by virtue of audio watermarking.

  7. Audio Watermarking with Error Correction

    CERN Document Server

    Chadha, Aman; Goel, Rishabh; Dave, Hiren; Roja, M Mani

    2011-01-01

    In recent times, communication through the internet has tremendously facilitated the distribution of multimedia data. Although this is indubitably a boon, one of its repercussions is that it has also given impetus to the notorious issue of online music piracy. Unethical attempts can also be made to deliberately alter such copyrighted data and thus, misuse it. Copyright violation by means of unauthorized distribution, as well as unauthorized tampering of copyrighted audio data is an important technological and research issue. Audio watermarking has been proposed as a solution to tackle this issue. The main purpose of audio watermarking is to protect against possible threats to the audio data and in case of copyright violation or unauthorized tampering, authenticity of such data can be disputed by virtue of audio watermarking.

  8. Digital Audio Collections

    Directory of Open Access Journals (Sweden)

    Jason Tenter

    2010-11-01

    Full Text Available

    This paper is about the possibility of libraries creating digital music or audio collections based on the current state of the digital music industry, and in comparison with the difficulties librarians have found in adding e-books to collections. In comparing the e-book and digital music markets, factors such as digital rights management (DRM and the differences in both markets’ relationships with customers are examined. This juxtaposition suggests that where e-books have been difficult to include in library collections because publishers want to maintain control over their content, music publishers have had to resign some of the control over their products because of file-sharing, and so may work with libraries to develop these collections in a more constructive way than e-book venders. At the end of the paper, some models are suggested for developing these collections.

  9. 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

  10. Digital Audio Watermarking: An Overview

    Directory of Open Access Journals (Sweden)

    Bhuvnesh Kumar Singh

    2013-10-01

    Full Text Available Digital watermarking is a very recent research area. Digital audio watermarking is a method to embed or hide the Watermark (Information signal into a digital signal i.e. Image, audio, text or video data. The watermark is difficult to remove from the audio signal. If the signal is copied, the information or watermark is also carried in the copy. A signal may carry several different watermarks at the same time. It is used to protecting multimedia data from unauthorized copying, piracy, ownership, inventions, authentication etc. in this paper we present the watermarking methods and applications

  11. 2.5维可控源音频大地电磁法Occam反演理论及应用%2.5 D controlled source audio-frequency magnetotellurics OCCAM inversion

    Institute of Scientific and Technical Information of China (English)

    何梅兴; 胡祥云; 叶益信; 罗文行

    2011-01-01

    Occam inversion is a regularization inversion to generate smooth models. The smoothest model is sought to the criterion that minimizes the misfit to the data. As resistivity of underground medium changes continually, Occam inversion considers both lateral and vertical roughness of underground medium. It avoids resistivity of model be interrupted arbitrarily. The paper takes a layered model and with two abnormities models for example, uses finite element methods of 2. 5 dimensional controlled source audio-frequency magnetotellurics to calculate response, then finds the models that fitting the response data by Occam inversion method. The results show that Occam inversion to CSAMT data is stable and convergent, the app-resistivity lines with high coherence, the misfit of a set data less than 3 %, and the smoothness is presented in inversion results.%Occam反演是一种正则化的光滑模型反演方法,它在寻找最小拟合差的同时追求最光滑模型.因地下介质的电性通常是连续变化的,Occam反演考虑了地下介质横向和纵向的光滑情况,应用Occam反演避免了模型电性参数被随意间断.通过建立层状和存在两个异常体的模型,利用2.5维可控源音频大地电磁有限元法作正演响应计算,对响应数据基于Occam反演理论拟合反演.结果表明,Occam反演对可控源音频大地电磁法响应数据反演稳定收敛,理论模型与反演结果的视电阻率曲线形态基本一致,测点数据的拟合误差小于3%,反演结果能反映出模型电阻率的光滑效果.

  12. 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,

  13. Tag Based Audio Search Engine

    Directory of Open Access Journals (Sweden)

    Parameswaran Vellachu

    2012-03-01

    Full Text Available The volume of the music database is increasing day by day. Getting the required song as per the choice of the listener is a big challenge. Hence, it is really hard to manage this huge quantity, in terms of searching, filtering, through the music database. It is surprising to see that the audio and music industry still rely on very simplistic metadata to describe music files. However, while searching audio resource, an efficient "Tag Based Audio Search Engine" is necessary. The current research focuses on two aspects of the musical databases 1. Tag Based Semantic Annotation Generation using the tag based approach.2. An audio search engine, using which the user can retrieve the songs based on the users choice. The proposed method can be used to annotation and retrieve songs based on musical instruments used , mood of the song, theme of the song, singer, music director, artist, film director, instrument, genre or style and so on.

  14. 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...

  15. 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.

  16. 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

  17. 47 CFR Figure 2 to Subpart N of... - Typical Audio Wave

    Science.gov (United States)

    2010-10-01

    ... 47 Telecommunication 1 2010-10-01 2010-10-01 false Typical Audio Wave 2 Figure 2 to Subpart N of Part 2 Telecommunication FEDERAL COMMUNICATIONS COMMISSION GENERAL FREQUENCY ALLOCATIONS AND RADIO... Audio Wave EC03JN91.006...

  18. Python for audio signal processing

    OpenAIRE

    Glover, John C.; Lazzarini, Victor; Timoney, Joseph

    2011-01-01

    This paper discusses the use of Python for developing audio signal processing applications. Overviews of Python language, NumPy, SciPy and Matplotlib are given, which together form a powerful platform for scientic computing. We then show how SciPy was used to create two audio programming libraries, and describe ways that Python can be integrated with the SndObj library and Pure Data, two existing environments for music composition and signal processing.

  19. 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....

  20. Modeling Audio Fingerprints: Structure, Distortion, Capacity

    OpenAIRE

    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, and ingested into a database, together with all relevant metadata. In the identification phase, unknown audio content is fingerprinted, and the fingerprints form the query to the database. The que...

  1. A Reproducible Research Framework for Audio Inpainting

    OpenAIRE

    Adler, Amir; Emiya, Valentin; Jafari, Maria,; Elad, Michael; Gribonval, Rémi; Plumbley, Mark D.

    2011-01-01

    International audience We introduce a unified framework for the restoration of distorted audio data, leveraging the Image Inpainting concept and covering existing audio applications. In this framework, termed Audio Inpainting, the distorted data is considered missing and its location is assumed to be known. We further introduce baseline approaches based on sparse representations. For this new audio inpainting concept, we provide reproducible-research tools including: the handling of audio ...

  2. Development of an audio input toolkit for multiple sources

    OpenAIRE

    Kosch, Thomas

    2013-01-01

    Audio services, like voice over IP or several voice recognition systems, are developing very fast and since they are easy to use nearly everybody is linked to such systems. In this thesis about the processing of multiple audio inputs, an audio toolkit for processing multiple audio inputs has to be developed. Used audio input devices are bluetooth headsets, which can send audio via UDP to the audio toolkit. This audio toolkit is able to process these multiple audio inputs and determines a domi...

  3. Concept Framework for Audio Information Retrieval: ARF

    Institute of Scientific and Technical Information of China (English)

    LI GuoHui(李国辉); WU DeFeng(武德峰); ZHANG Jun(张军)

    2003-01-01

    The majority of researches on content-based retrieval focused on visual media.However audio is also an important medium and information carrier from the viewpoint of humanauditory perception, so it is needed to retrieve for audio collection. Audio is handled by conven-tional methods as an opaque stream medium, which is not suitable for information retrieval byits content. In fact, audio carries rich aural information with the form of speech, musical, andsound effects, so it could be retrieved based on its aural content, such as acoustic features, musicalmelodies and associated semantics. In this paper, a concept framework (ARF) for content-basedaudio retrieval is proposed from systematic perspectives, which describes audio content model,audio retrieval architecture and audio query schemes. Audio contents are represented by a hier-archical model and a set of formal descriptions from physical to acoustic to semantic level, whichdepict acoustic features, logical structure and semantics of audio and audio objects. The archi-tecture consisting of audio meta-database, populating and accessing modules presents a systemstructure view of audio information retrieval. The query schemes give generalized approaches andmodes concerning how users deliver audio information needs to audio collections. Finally, an audioretrieval example implemented is used to explain and specify the application of the components in the proposed ARF.

  4. 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 d...

  5. Multi Carrier Modulation Audio Power Amplifier with Programmable Logic

    DEFF Research Database (Denmark)

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

    2009-01-01

    amplifiers while keeping the performance measures to excellent levels is therefore of high interest. In this paper a class D audio amplifier utilising Multi Carrier Modulation (MCM) will be analysed, and a prototype Master-Slave Multi Carrier Modulated (MS MCM) amplifier has been constructed and measured for...... performance and out of band spectral amplitudes. The basic principle in MCM is to use programmable logic to combine two or more Pulse Width Modulated (PWM) audio signals at different switching frequencies. In this way the out of band spectrum will be lowered compared with conventional class D amplifiers...

  6. 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

  7. Digital Audio Radio Broadcast Systems Laboratory Testing Nearly Complete

    Science.gov (United States)

    2005-01-01

    Radio history continues to be made at the NASA Lewis Research Center with the completion of phase one of the digital audio radio (DAR) testing conducted by the Consumer Electronics Group of the Electronic Industries Association. This satellite, satellite/terrestrial, and terrestrial digital technology will open up new audio broadcasting opportunities both domestically and worldwide. It will significantly improve the current quality of amplitude-modulated/frequency-modulated (AM/FM) radio with a new digitally modulated radio signal and will introduce true compact-disc-quality (CD-quality) sound for the first time. Lewis is hosting the laboratory testing of seven proposed digital audio radio systems and modes. Two of the proposed systems operate in two modes each, making a total of nine systems being tested. The nine systems are divided into the following types of transmission: in-band on-channel (IBOC), in-band adjacent-channel (IBAC), and new bands. The laboratory testing was conducted by the Consumer Electronics Group of the Electronic Industries Association. Subjective assessments of the audio recordings for each of the nine systems was conducted by the Communications Research Center in Ottawa, Canada, under contract to the Electronic Industries Association. The Communications Research Center has the only CCIR-qualified (Consultative Committee for International Radio) audio testing facility in North America. The main goals of the U.S. testing process are to (1) provide technical data to the Federal Communication Commission (FCC) so that it can establish a standard for digital audio receivers and transmitters and (2) provide the receiver and transmitter industries with the proper standards upon which to build their equipment. In addition, the data will be forwarded to the International Telecommunications Union to help in the establishment of international standards for digital audio receivers and transmitters, thus allowing U.S. manufacturers to compete in the

  8. 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....

  9. Lossless Audio Watermarking Based on the Alpha Statistic Modulation

    Directory of Open Access Journals (Sweden)

    Sunita V. Dhavale

    2012-09-01

    Full Text Available In this paper, we propose a high capacity, self-synchronized, lossless audio watermarking algorithm based on the alpha (‘α’ statistic modulation. Here ‘α’ is related to the correlation among any given sequence i.e audio samples and it is modulated according to the watermark bit stream. The embedding scheme is tested in both the time domain and DWT domain. Though the time domain embedding reduces the computational time in searching the synchronization codes, the time-frequency localization capability of DWT provides good trade off between the computational complexity and robustness of synchronization codes. In case of DWT, ‘α’ related to the 2nd level DWT coarse wavelet components is used for embedding the watermark. The offset value used for embedding is made adaptive to the required SNR for the final watermarked audio signal. After extraction of the embedded watermark using a watermark key, original audio can be recovered with minimal distortion. The watermarking method presented here does not require the use of the original signal for watermark detection. Also high embedding capacity is achieved by using small sizedaudio frames. Experimental results reveal that the proposed watermarking scheme maintains high audio quality and is simultaneously highly robust to pirate attacks, including MP3 compression, cropping, filtering, re-sampling, and re-quantization.

  10. 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

  11. Survey Musik und Medien 2012: Audio Media Usage in Germany - Audio Emitters - Audio Emitters used in 2012 - Versatile Traditionalists

    OpenAIRE

    Lepa, Steffen

    2013-01-01

    Where did the sound emanate from in 2012? Audio Emitters comprise those technical objects that are connected to Audio Devices in order to make Audio Sources audible. This category includes headphones and loudspeakers with both diverse sound formats (mono, stereo, surround) as well as various device types (Headphones: small, standard, HiFi; Loudspeakers: integrated models, single components and docking stations). Versatile Traditionalists constitute the most prevalent audio repertoire with...

  12. Audio watermark a comprehensive foundation using Matlab

    CERN Document Server

    Lin, Yiqing

    2015-01-01

    This book illustrates the commonly used and novel approaches of audio watermarking for copyrights protection. The author examines the theoretical and practical step by step guide to the topic of data hiding in audio signal such as music, speech, broadcast. The book covers new techniques developed by the authors are fully explained and MATLAB programs, for audio watermarking and audio quality assessments and also discusses methods for objectively predicting the perceptual quality of the watermarked audio signals. Explains the theoretical basics of the commonly used audio watermarking techniques Discusses the methods used to objectively and subjectively assess the quality of the audio signals Provides a comprehensive well tested MATLAB programs that can be used efficiently to watermark any audio media

  13. Digital Audio Application to Short Wave Broadcasting

    Science.gov (United States)

    Chen, Edward Y.

    1997-01-01

    Digital audio is becoming prevalent not only in consumer electornics, but also in different broadcasting media. Terrestrial analog audio broadcasting in the AM and FM bands will be eventually be replaced by digital systems.

  14. QRDA: Quantum Representation of Digital Audio

    Science.gov (United States)

    Wang, Jian

    2016-03-01

    Multimedia refers to content that uses a combination of different content forms. It includes two main medias: image and audio. However, by contrast with the rapid development of quantum image processing, quantum audio almost never been studied. In order to change this status, a quantum representation of digital audio (QRDA) is proposed in this paper to present quantum audio. QRDA uses two entangled qubit sequences to store the audio amplitude and time information. The two qubit sequences are both in basis state: |0> and |1>. The QRDA audio preparation from initial state |0> is given to store an audio in quantum computers. Then some exemplary quantum audio processing operations are performed to indicate QRDA's usability.

  15. Audio Steganography Using GA with Multilevel Security

    OpenAIRE

    K. Bhowal; Sarkar, D.; Biswas, S.; P.P. Sarkar

    2013-01-01

    In this paper we present a novel method for digital audio steganography where messages are embedded into image and image is embedded into the host audio. “Audio Steganography using GA with multilevel security” is a proposed system which is based on Steganography and Encryption; the system ensures secured data transfer between the source and destination. Here a novel approach is presented to resolve the remained problems of substitution technique of audio Steganography. In the first level of s...

  16. 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...

  17. 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...

  18. 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…

  19. Audio-Visual Aids: Historians in Blunderland.

    Science.gov (United States)

    Decarie, Graeme

    1988-01-01

    A history professor relates his experiences producing and using audio-visual material and warns teachers not to rely on audio-visual aids for classroom presentations. Includes examples of popular audio-visual aids on Canada that communicate unintended, inaccurate, or unclear ideas. Urges teachers to exercise caution in the selection and use of…

  20. 36 CFR 2.12 - Audio disturbances.

    Science.gov (United States)

    2010-07-01

    ... 36 Parks, Forests, and Public Property 1 2010-07-01 2010-07-01 false Audio disturbances. 2.12... RESOURCE PROTECTION, PUBLIC USE AND RECREATION § 2.12 Audio disturbances. (a) The following are prohibited..., motorized toy, or an audio device, such as a radio, television set, tape deck or musical instrument, in...

  1. 50 CFR 27.72 - Audio equipment.

    Science.gov (United States)

    2010-10-01

    ... 50 Wildlife and Fisheries 6 2010-10-01 2010-10-01 false Audio equipment. 27.72 Section 27.72 Wildlife and Fisheries UNITED STATES FISH AND WILDLIFE SERVICE, DEPARTMENT OF THE INTERIOR (CONTINUED) THE... Audio equipment. The operation or use of audio devices including radios, recording and playback...

  2. AudioRegent: Exploiting SimpleADL and SoX for Digital Audio Delivery

    OpenAIRE

    Nitin Arora

    2010-01-01

    AudioRegent is a command-line Python script currently being used by the University of Alabama Libraries’ Digital Services to create web-deliverable MP3s from regions within archival audio files. In conjunction with a small-footprint XML file called SimpleADL and SoX, an open-source command-line audio editor, AudioRegent batch processes archival audio files, allowing for one or many user-defined regions, particular to each audio file, to be extracted with additional audio processing in a trans...

  3. 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.

  4. Hi fi digital audio tape to SUN workstation transfer system for digital audio data

    OpenAIRE

    Gartenlaub, Arie Gal

    1994-01-01

    This thesis describes a subsystem developed to provide for the transfer of digital audio signals from a SUN SPARCstation 10 workstation to a digital audio tape (DAT) and vice versa. The new system expands the audio recording/reproduction options available in the laboratory by integrating an analog tape deck and a digital tape deck with the SUN workstation. The desired connection enables working with a larger audio bandwidth to achieve better audio performance and resolution in comparison to t...

  5. 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), and...... then map the DOA estimates to a location. In practice, however, the individual nodes contain few microphones, limiting the DOA estimation accuracy and, thereby, also the localization performance. We investigate a new approach, where range estimates are also obtained and utilized from each node, e.......g., using 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....

  6. 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.

  7. Digital Augmented Reality Audio Headset

    OpenAIRE

    Jussi Rämö; Vesa Välimäki

    2012-01-01

    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 ...

  8. WAVE : an audio virtual environment

    OpenAIRE

    Valbom, Leonel; Marcos, Adérito

    2004-01-01

    This paper outlines the basis and gives a description of the project WAVE that is starting in the Department of Information Systems of the University of Minho in co-operation with a research group in the Computer Graphics Centre - ZGDV, Guimaraes. The project aims to set up an immersive environment of virtual reality, where the music, sound and audio (3D or not) plays an important role in a virtual musical/sound instrument for performances, education, entertainment or experimentat...

  9. Audio Watermarking with Error Correction

    OpenAIRE

    Aman Chadha; Sandeep Gangundi; Rishabh Goel; Hiren Dave; M.Mani Roja

    2011-01-01

    In recent times, communication through the internet has tremendously facilitated the distribution of multimedia data. Although this is indubitably a boon, one of its repercussions is that it has also given impetus to the notorious issue of online music piracy. Unethical attempts can also be made to deliberately alter such copyrighted data and thus, misuse it. Copyright violation by means of unauthorized distribution, as well as unauthorized tampering of copyrighted audio data is an important ...

  10. Audio Interfaces for Improved Accessibility

    OpenAIRE

    Duarte, Carlos; Carrico, Lu&#;s

    2008-01-01

    This chapter focused on how endowing interfaces with audio interaction capabilities can improve their accessibility. To exemplify this outcome the development of several versions of a Digital Talking Book player was presented. This allowed us to show it is possible to maintain the same set of features while stripping the interface of visual components, and still keep it usable for the visually impaired population. The interface development concerns focused on both ends of the interaction spec...

  11. A novel fiber audio transmission system for secure communication

    Institute of Scientific and Technical Information of China (English)

    SU Ke; JIA Bo

    2005-01-01

    A new,simple and efficient fiber audio transmission method for the long distance secure communication is presented, which performs signal modulation by the strain-optic effects and signal demodulation by the all-fiber interferometer. The interferometer is a truly path-matched device, which eliminates much of the undesirable noise by combining the reference and the sensing arms within the same optical fiber. The sinusoidal signals adopted in the experiment are in a frequency range of 300 HZ-3 400 HZ and of the multi-frequency, and the system shows good capabilities, robust security and maintenance of audio integrity. The device may be applicable in the field of point to point secure communication of 40 kilometer long transmission range.

  12. C Implementation & comparison of companding & silence audio compression techniques

    OpenAIRE

    Dangarwala, Kruti; Shah, Jigar

    2010-01-01

    Just about all the newest living room audio-video electronics and PC multimedia products being designed today will incorporate some form of compressed digitized-audio processing capability. Audio compression reduces the bit rate required to represent an analog audio signal while maintaining the perceived audio quality. Discarding inaudible data reduces the storage, transmission and compute requirements of handling high-quality audio files. This paper covers wave audio file format & algorithm ...

  13. Scalable audio separation with light kernel additive modelling

    OpenAIRE

    Liutkus, Antoine; Fitzgerald, Derry; Rafii, Zafar

    2015-01-01

    Recently, Kernel Additive Modelling (KAM) was proposed as a unified framework to achieve multichannel audio source separation. Its main feature is to use kernel models for locally describing the spectrograms of the sources. Such kernels can capture source features such as repetitivity, stability over time and/or frequency, self-similarity, etc. KAM notably subsumes many popular and effective methods from the state of the art, including REPET and harmonic/percussive separation with median filt...

  14. Survey Musik und Medien 2012: Audio Media Usage in Germany - Audio Devices - Audio Devices used in 2012 - Selective Traditionalists

    OpenAIRE

    Lepa, Steffen

    2013-01-01

    By what means was music played back in 2012? Audio Devices comprise technical devices that permit access to and enable playback of Audio Sources. This includes CD players, record players, cassette recorders, MP3 player and smartphones but also computers and various multimedia entertainment devices that allow music use. 11,6 % of our participants may be described as Selective Traditionalists who are typically born between 1955 and 1975. The radio is the dominant audio source used at least ...

  15. Survey Musik und Medien 2012: Audio Media Usage in Germany - Audio Emitters - Audio Emitters in used in 2012

    OpenAIRE

    Lepa, Steffen

    2013-01-01

    Where did the sound emanate from in 2012? Audio Emitters comprise those technical objects that are connected to Audio Devices in order to make Audio Sources audible. This category includes headphones and loudspeakers with both diverse sound formats (mono, stereo, surround) as well as various device types (Headphones: small, standard, HiFi; Loudspeakers: integrated models, single components and docking stations). How do the Germans listen to music nowadays? Survey Musik und Medien 2012 deli...

  16. Survey Musik und Medien 2012: Audio Media Usage in Germany - Audio Devices - Audio Devices used in 2012

    OpenAIRE

    Lepa, Steffen

    2013-01-01

    By what means was music played back in 2012? Audio Devices comprise technical devices that permit access to and enable playback of Audio Sources. This includes CD players, record players, cassette recorders, MP3 player and smartphones but also computers and various multimedia entertainment devices that allow music use. How do the Germans listen to music nowadays? Survey Musik und Medien 2012 delivers representative data on actual audio media usage of German population. These data allow the...

  17. Survey Musik und Medien 2012: Audio Media Usage in Germany - Audio Emitters - Audio Emitters used in 2012 - Radio Traditionalists

    OpenAIRE

    Lepa, Steffen

    2013-01-01

    Where did the sound emanate from in 2012? Audio Emitters comprise those technical objects that are connected to Audio Devices in order to make Audio Sources audible. This category includes headphones and loudspeakers with both diverse sound formats (mono, stereo, surround) as well as various device types (Headphones: small, standard, HiFi; Loudspeakers: integrated models, single components and docking stations). Radio Traditionalists are represented in various age groups, and may be born ...

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

    Directory of Open Access Journals (Sweden)

    Theodoros Giannakopoulos

    Full Text Available 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.

  19. 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. PMID:26656189

  20. Using the ENF Criterion for Determining the Time of Recording of Short Digital Audio Recordings

    Science.gov (United States)

    Huijbregtse, Maarten; Geradts, Zeno

    The Electric Network Frequency (ENF) Criterion is a recently developed forensic technique for determining the time of recording of digital audio recordings, by matching the ENF pattern from a questioned recording with an ENF pattern database. In this paper we discuss its inherent limitations in the case of short - i.e., less than 10 minutes in duration - digital audio recordings. We also present a matching procedure based on the correlation coefficient, as a more robust alternative to squared error matching.

  1. Audio representations of multi-channel EEG: a new tool for diagnosis of brain disorders

    OpenAIRE

    Vialatte, François B.; Dauwels, Justin; Musha, Toshimitsu; Cichocki, Andrzej

    2012-01-01

    Objective: The objective of this paper is to develop audio representations of electroencephalographic (EEG) multichannel signals, useful for medical practitioners and neuroscientists. The fundamental question explored in this paper is whether clinically valuable information contained in the EEG, not available from the conventional graphical EEG representation, might become apparent through audio representations. Methods and Materials: Music scores are generated from sparse time-frequency maps...

  2. 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.

  3. 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 the...... 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 or...

  4. Implementing Audio-CASI on Windows’ Platforms

    OpenAIRE

    Cooley, Philip C.; Turner, Charles F.

    1998-01-01

    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 ...

  5. Weakly Supervised Scalable Audio Content Analysis

    OpenAIRE

    Kumar, Anurag; Raj, Bhiksha

    2016-01-01

    Audio Event Detection is an important task for content analysis of multimedia data. Most of the current works on detection of audio events is driven through supervised learning approaches. We propose a weakly supervised learning framework which can make use of the tremendous amount of web multimedia data with significantly reduced annotation effort and expense. Specifically, we use several multiple instance learning algorithms to show that audio event detection through weak labels is feasible...

  6. Loudness Control by Intelligent Audio Content Analysis

    OpenAIRE

    Sabri, Hossein

    2013-01-01

    Automatic audio segmentation aims at extracting information of audio signals. In the case of music tracks, detecting segment boundaries, labelling segments, and detecting repeated segments could be performed. These information can be used in different applications such as creating song summaries and facilitating browsing in music collections. This thesis studies the Foote method which is one of the automatic audio segmentation algorithms. Numerous experiments are carried out to improve the pe...

  7. MODIS: an audio motif discovery software

    OpenAIRE

    Catanese, Laurence; Souviraà-Labastie, Nathan; Qu, Bingqing; Campion, Sébastien; Gravier, Guillaume; Vincent, Emmanuel; Bimbot, Frédéric

    2013-01-01

    International audience MODIS is a free speech and audio motif discovery software developed at IRISA Rennes. Motif discovery is the task of discovering and collecting occurrences of repeating patterns in the absence of prior knowledge, or training material. MODIS is based on a generic approach to mine repeating audio sequences, with tolerance to motif variability. The algorithm implementation allows to process large audio streams at a reasonable speed where motif discovery often requires hu...

  8. New audio applications of beryllium metal

    International Nuclear Information System (INIS)

    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)

  9. Three-Dimensional Audio Client Library

    Science.gov (United States)

    Rizzi, Stephen A.

    2005-01-01

    The Three-Dimensional Audio Client Library (3DAudio library) is a group of software routines written to facilitate development of both stand-alone (audio only) and immersive virtual-reality application programs that utilize three-dimensional audio displays. The library is intended to enable the development of three-dimensional audio client application programs by use of a code base common to multiple audio server computers. The 3DAudio library calls vendor-specific audio client libraries and currently supports the AuSIM Gold-Server and Lake Huron audio servers. 3DAudio library routines contain common functions for (1) initiation and termination of a client/audio server session, (2) configuration-file input, (3) positioning functions, (4) coordinate transformations, (5) audio transport functions, (6) rendering functions, (7) debugging functions, and (8) event-list-sequencing functions. The 3DAudio software is written in the C++ programming language and currently operates under the Linux, IRIX, and Windows operating systems.

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

    OpenAIRE

    Theodoros Giannakopoulos

    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 wid...

  11. Audio description and audio subtitling in a dubbing country: Case studies

    OpenAIRE

    Benecke, Bernd

    2012-01-01

    In many European countries foreign films are not dubbed but subtitled. An audio describer has to include all the written subtitles in his script and try to make the description fit in between. Dubbing countries like Spain, Italy and Germany are also used to combining audio description and audio subtitling – for different reasons. This presentation shows how audio subtitling affects the work of describers in a dubbing country like Germany. It will present examples from daily wor...

  12. Audio-visual gender recognition

    Science.gov (United States)

    Liu, Ming; Xu, Xun; Huang, Thomas S.

    2007-11-01

    Combining different modalities for pattern recognition task is a very promising field. Basically, human always fuse information from different modalities to recognize object and perform inference, etc. Audio-Visual gender recognition is one of the most common task in human social communication. Human can identify the gender by facial appearance, by speech and also by body gait. Indeed, human gender recognition is a multi-modal data acquisition and processing procedure. However, computational multimodal gender recognition has not been extensively investigated in the literature. In this paper, speech and facial image are fused to perform a mutli-modal gender recognition for exploring the improvement of combining different modalities.

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

    NARCIS (Netherlands)

    Doets, P.J.O.; Menot Gisbert, M.; Lagendijk, R.L.

    2006-01-01

    Audio fingerprints can be seen as hashes of the perceptual content of an audio excerpt. Applications include linking metadata to unlabeled audio, watermark support, and broadcast monitoring. Existing systems identify a song by comparing its fingerprint to pre-computed fingerprints in a database. Sma

  14. The Effect Of 3D Audio And Other Audio Techniques On Virtual Reality Experience

    NARCIS (Netherlands)

    Brinkman, W.P.; Hoekstra, A.R.D.; Van Egmond, R.

    2015-01-01

    Three studies were conducted to examine the effect of audio on people's experience in a virtual world. The first study showed that people could distinguish between mono, stereo, Dolby surround and 3D audio of a wasp. The second study found significant effects for audio techniques on people's self-re

  15. Audio Sensing Aid based Wireless Microphone Emulation Attacks Detection

    Directory of Open Access Journals (Sweden)

    Wang Shan-shan

    2013-10-01

    Full Text Available The wireless microphone network is an important PU network for CRN, but there is no effective technology to solve the problem of microphone evaluation attacks. Therefore, this paper propose ASA algorithm, which utilizes three devices to detect MUs, and they are loudspeaker audio sensor (LAS, environment audio sensor (EAS, and radio frequency fingerprint detector (RFFD. LASs are installed near loudspeakers, which have two main effects: One is to sense loudspeakers’ output, and the other is to broadcast warning information to all SUs through the common control channel when detecting valid output. EASs are pocket voice captures provided to SU, and utilized to sense loudspeaker sound at SU’s location. Utilizing EASs and energy detections in SU can detect primary user emulation attack (PUEA fast. But to acquire the information of attacked channels, we need explore RFFDs to analyze the features of PU transmitters. The results show that the proposed algorithm can detect PUEA well.    

  16. 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.

  17. Audio-Visual Aids in Universities

    Science.gov (United States)

    Douglas, Jackie

    1970-01-01

    A report on the proceedings and ideas expressed at a one day seminar on "Audio-Visual Equipment--Its Uses and Applications for Teaching and Research in Universities." The seminar was organized by England's National Committee for Audio-Visual Aids in Education in conjunction with the British Universities Film Council. (LS)

  18. 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 for...

  19. Internet Audio Products (3/3)

    Science.gov (United States)

    Schwartz, Linda; de Schutter, Adrienne; Fahrni, Patricia; Rudolph, Jim

    2004-01-01

    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…

  20. Synchronization and comparison of Lifelog audio recordings

    DEFF Research Database (Denmark)

    Nielsen, Andreas Brinch; Hansen, Lars Kai

    2008-01-01

    We investigate concurrent ‘Lifelog’ audio recordings to locate segments from the same environment. We compare two techniques earlier proposed for pattern recognition in extended audio recordings, namely cross-correlation and a fingerprinting technique. If successful, such alignment can be used as a...

  1. Audio Quality for a Simple Forward Error Correcting Code

    OpenAIRE

    Calas, Yvan; Jean-Marie, Alain

    2004-01-01

    International audience The aim of this paper is to study the audio quality offered by a simple Forward Error Correction (FEC) code used in audio applications like Freephone or Rat. This coding technique consists in adding to every audio packet a redundant information concerning a preceding audio packet which belongs to the same audio flow. We show that the audio quality depends not only on the number of FEC flows and the utility function associated to the quantity of information received, ...

  2. Progressive Syntax-Rich Coding of Multichannel Audio Sources

    OpenAIRE

    Yang Dai; Ai Hongmei; Kyriakakis Chris; Kuo C-C Jay

    2003-01-01

    Being able to transmit the audio bitstream progressively is a highly desirable property for network transmission. MPEG- version audio supports fine grain bit rate scalability in the generic audio coder (GAC). It has a bit-sliced arithmetic coding (BSAC) tool, which provides scalability in the step of 1 Kbps per audio channel. There are also several other scalable audio coding methods, which have been proposed in recent years. However, these scalable audio tools are only available for mono ...

  3. Progressive Syntax-Rich Coding of Multichannel Audio Sources

    OpenAIRE

    Dai Yang; Hongmei Ai; Chris Kyriakakis; C.-C. Jay Kuo

    2003-01-01

    Being able to transmit the audio bitstream progressively is a highly desirable property for network transmission. MPEG-4 version 2 audio supports fine grain bit rate scalability in the generic audio coder (GAC). It has a bit-sliced arithmetic coding (BSAC) tool, which provides scalability in the step of 1 Kbps per audio channel. There are also several other scalable audio coding methods, which have been proposed in recent years. However, these scalable audio tools are only available for mono ...

  4. Stego-audio Using Genetic Algorithm Approach

    Directory of Open Access Journals (Sweden)

    V. Santhi

    2014-06-01

    Full Text Available With the rapid development of digital multimedia applications, the secure data transmission becomes the main issue in data communication system. So the multimedia data hiding techniques have been developed to ensure the secured data transfer. Steganography is an art of hiding a secret message within an image/audio/video file in such a way that the secret message cannot be perceived by hacker/intruder. In this study, we use RSA encryption algorithm to encrypt the message and Genetic Algorithm (GA to encode the message in the audio file. This study presents a method to access the negative audio bytes and includes the negative audio bytes in the message encoding and position embedding process. This increases the capacity of encoding message in the audio file. The use of GA operators in Genetic Algorithm reduces the noise distortions.

  5. C Implementation & comparison of companding & silence audio compression techniques

    CERN Document Server

    Dangarwala, Kruti

    2010-01-01

    Just about all the newest living room audio-video electronics and PC multimedia products being designed today will incorporate some form of compressed digitized-audio processing capability. Audio compression reduces the bit rate required to represent an analog audio signal while maintaining the perceived audio quality. Discarding inaudible data reduces the storage, transmission and compute requirements of handling high-quality audio files. This paper covers wave audio file format & algorithm of silence compression method and companding method to compress and decompress wave audio file. Then it compares the result of these two methods.

  6. 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.

  7. Complete guide to high-end audio

    CERN Document Server

    Harley, Robert

    2010-01-01

    In this newly updated directory, the latest in cutting-edge audio equipment is provided, including how to choose the best audio equipment on a budget, how to get the best sound for the money, and how to set up a system for maximum performance. Revised and expanded to include all the latest audio technologies, this book is packed with expert advice how to make speakers sound up to 50 percent better at no cost, avoid the most common system set-up mistakes, and how to choose the one speaker in 50 worth owning. Among the new topics covered are computer-based music servers, wireless streaming of au

  8. 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...

  9. Survey Musik und Medien 2012: Audio Media Usage in Germany - Audio Devices - Audio Devices used in 2012 - Radio Traditionalists

    OpenAIRE

    Lepa, Steffen

    2013-01-01

    By what means was music played back in 2012? Audio Devices comprise technical devices that permit access to and enable playback of Audio Sources. This includes CD players, record players, cassette recorders, MP3 player and smartphones but also computers and various multimedia entertainment devices that allow music use. Radio Traditionalists are represented in various age groups, and may be born in 1920 as well as in 1959. They constitute 22,2 % of the German population between age 14 and ...

  10. Survey Musik und Medien 2012: Audio Media Usage in Germany - Audio Devices - Audio Devices used in 2012 - Digital Mobilists

    OpenAIRE

    Lepa, Steffen

    2013-01-01

    By what means was music played back in 2012? Audio Devices comprise technical devices that permit access to and enable playback of Audio Sources. This includes CD players, record players, cassette recorders, MP3 player and smartphones but also computers and various multimedia entertainment devices that allow music use. Typically born between 1979 and 1998, the Digital Mobilists constitute the youngest user type and comprise 16,1 % of the German Population. They access free video streaming...

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

    Science.gov (United States)

    Korycki, Rafal

    2014-05-01

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

  12. Audio Word2Vec: Unsupervised Learning of Audio Segment Representations using Sequence-to-sequence Autoencoder

    OpenAIRE

    Chung, Yu-An; Wu, Chao-Chung; Shen, Chia-Hao; Lee, Hung-Yi; Lee, Lin-shan

    2016-01-01

    The vector representations of fixed dimensionality for words (in text) offered by Word2Vec have been shown to be very useful in many application scenarios, in particular due to the semantic information they carry. This paper proposes a parallel version, the Audio Word2Vec. It offers the vector representations of fixed dimensionality for variable-length audio segments. These vector representations are shown to describe the sequential phonetic structures of the audio segments to a good degree, ...

  13. Security of audio secret sharing scheme encrypting audio secrets with bounded shares

    OpenAIRE

    鷲尾, 槙也; 渡邊, 曜大

    2014-01-01

    Secret sharing is a method of encrypting a secret into multiple pieces called shares so that only qualified sets of shares can be employed to reconstruct the secret. Audio secret sharing (ASS) is an example of secret sharing whose decryption can be performed by human ears. This paper examines the security of an audio secret sharing scheme encrypting audio secrets with bounded shares, and optimizes the security with respect to the probability distribution used in its encryption.

  14. A Morphological Analysis of Audio Objects and their Control Methods for 3D Audio

    OpenAIRE

    Mathew, Justin; Huot, Stéphane; Blum, Alan

    2014-01-01

    International audience Recent technological improvements in audio reproduction systems increased the possibilities to spatialize sources in a listening environment. The spatialization of reproduced audio is highly dependent on the recording technique, the rendering method, and the loudspeaker configuration. While object-based audio production reduces this dependency on loudspeaker configurations, related authoring tools are still difficult to interact with. In this paper, we investigate th...

  15. Implementation of Audio signal by using wavelet transform

    Directory of Open Access Journals (Sweden)

    Chakresh kumar,

    2010-10-01

    Full Text Available Audio coding is the technology to represent audio in digital form with as few bits as possible while maintaining the intelligibility and quality required for particular application. Interest in audio coding is motivated by the evolution to digital communications and the requirement to minimize bit rate, and hence conserve bandwidth. There is always a tradeoff between compression ratio and maintaining the delivered audio quality and intelligibility. Audio coding is widely used in application such as digital broadcasting, Internet audio or music database to reduce the bit rate of high quality audio signal without comprising the perceptual quality. In this dissertation work Design and implementation of a MPEG Lossless audio codec using wavelet transform has been proposed. The major issues concerning the development of audio codec are choosing optimal wavelets for audio signals, decomposition level in the digital wavelet transform and thresholding criteria for coefficient truncation which is the basis to provide compression ratio for audio with suitable peak signal to noise ratio (PSNR, wavelet packet compression technique has also been used to compare the performanceof audio codec using wavelet transform. A psychoacoustic model is used to improve the quality of audio signal. The proposed audio codec has been implemented on DSK6713 Starter Kit using MATLAB-7.3 and Link to Code Composer Studio and various audio signals of different time duration have been tested. Result obtained show that the proposed codec improves quality of the reconstructed audio signal.

  16. 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...

  17. 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...

  18. 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.

  19. Audio-visual affective expression recognition

    Science.gov (United States)

    Huang, Thomas S.; Zeng, Zhihong

    2007-11-01

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

  20. Spatial audio quality perception (part 1)

    DEFF Research Database (Denmark)

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

    2015-01-01

    . The choice of the SAP can have a large influence on the degree of degradation. Taken together these findings and the quality-annotated database can guide the development of a regression model of perceived overall spatial audio quality, incorporating previously developed spatially-relevant feature......Spatial audio processes (SAPs) commonly encountered in consumer audio reproduction systems are known to produce a range of impairments to spatial quality. By way of two listening tests, this paper investigated the degree of degradation of the spatial quality of six 5-channel audio recordings...... resulting from 48 such SAPs. Perceived degradation also depends on the particular listeners, the program content, and the listening location. For example, combining off-center listener with another SAP can reduce spatial quality significantly when compared to listening to that SAP from a central location...

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

    DEFF Research Database (Denmark)

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

    2014-01-01

    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......, annoyance, balance and blend, and confusion. Ratings using these attributes were collected in the fourth stage, and a principal component analysis performed. This suggested two dimensions underlying the perception of an audio-on-audio interference situation: The first dimension was labeled “distraction” and...

  2. Audio watermarking for live performance

    Science.gov (United States)

    Tachibana, Ryuki

    2003-06-01

    Audio watermarking has been used mainly for digitally stored content. Using real-time watermark embedding, its coverage can be extended to live broadcasts and live performances. In general, a conventional embedding algorithm receives a host signal (HS) and outputs the summation of the HS and a watermark signal (WS). However, when applied to real-time embedding, there are two problems: (1) delay of the HS, and (2) possible interruption of the broadcast. To solve these problems, we propose a watermark generation algorithm that outputs only a WS, and a system composition method in which a mixer outside the computer mixes the WS generated by the algorithm and the HS. In addition, we propose a new composition method "sonic watermarking." In this composition method, the sound of the HS and the sound of the WS are played separately by two speakers, and the sounds are mixed in the air. Using this composition method, it would be possible to generate a watermarking sound in a concerto hall so that the watermark could be detected from content recorded by audience members who have recording devices at their seats. We report on the results of experiments and discuss the merits and flaws of various real-time watermarking composition methods.

  3. Quantitative characterisation of audio data by ordinal symbolic dynamics

    Science.gov (United States)

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

    2013-06-01

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

  4. 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

  5. PENGGUNAAN MEDIA AUDIO DALAM PEMBELAJARAN STENOGRAFI

    Directory of Open Access Journals (Sweden)

    S Martono

    2011-06-01

    Full Text Available The objective this study is to know the effectivenes of using audio media in stenografi typing learning. The population  of this research was 30 students that divided into two groups; experimental and controlled group consisted of 15 students. Based on the first score in stenografi subject that the two groups have the same abillity but they were given different treatment. For experimental group, they got a treatment of audio media whereas the controlled group didn’t use audio media. The technique of collecting data were documentation technique and experimental tecnique. The instrument was stenografi speed typing. The final result showed that the using of audio media was more effective and can improve the study result better than controlled group. This result was expected to  give significance for the stenografi teachers to apply audio media in learning and input for the students that stenografi was not a memorizing subject but it was a skill subject that must be trained by joining the lesson. Thus, people can use stenografi typing to record each talk. Keywords: Learning, Audio Media, Stenografi

  6. Implementation of Audio signal by using wavelet transform

    OpenAIRE

    Chakresh kumar; Chandra Shekhar; Mrs. Ashu Soni; Bindu Thakral

    2010-01-01

    Audio coding is the technology to represent audio in digital form with as few bits as possible while maintaining the intelligibility and quality required for particular application. Interest in audio coding is motivated by the evolution to digital communications and the requirement to minimize bit rate, and hence conserve bandwidth. There is always a tradeoff between compression ratio and maintaining the delivered audio quality and intelligibility. Audio coding is widely used in application s...

  7. Digital Multicasting of Multiple Audio Streams

    Science.gov (United States)

    Macha, Mitchell; Bullock, John

    2007-01-01

    The Mission Control Center Voice Over Internet Protocol (MCC VOIP) system (see figure) comprises hardware and software that effect simultaneous, nearly real-time transmission of as many as 14 different audio streams to authorized listeners via the MCC intranet and/or the Internet. The original version of the MCC VOIP system was conceived to enable flight-support personnel located in offices outside a spacecraft mission control center to monitor audio loops within the mission control center. Different versions of the MCC VOIP system could be used for a variety of public and commercial purposes - for example, to enable members of the general public to monitor one or more NASA audio streams through their home computers, to enable air-traffic supervisors to monitor communication between airline pilots and air-traffic controllers in training, and to monitor conferences among brokers in a stock exchange. At the transmitting end, the audio-distribution process begins with feeding the audio signals to analog-to-digital converters. The resulting digital streams are sent through the MCC intranet, using a user datagram protocol (UDP), to a server that converts them to encrypted data packets. The encrypted data packets are then routed to the personal computers of authorized users by use of multicasting techniques. The total data-processing load on the portion of the system upstream of and including the encryption server is the total load imposed by all of the audio streams being encoded, regardless of the number of the listeners or the number of streams being monitored concurrently by the listeners. The personal computer of a user authorized to listen is equipped with special- purpose MCC audio-player software. When the user launches the program, the user is prompted to provide identification and a password. In one of two access- control provisions, the program is hard-coded to validate the user s identity and password against a list maintained on a domain-controller computer

  8. 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…

  9. Separate mechanisms for audio-tactile pitch and loudness interactions

    Directory of Open Access Journals (Sweden)

    JeffreyMYau

    2010-10-01

    Full Text Available A major goal in perceptual neuroscience is to understand how signals from different sensory modalities are combined to produce stable and coherent representations. We previously investigated interactions between audition and touch, motivated by the fact that both modalities are sensitive to environmental oscillations. In our earlier study, we characterized the effect of auditory distractors on tactile frequency and intensity perception. Here, we describe the converse experiments examining the effect of tactile distractors on auditory processing. Because the two studies employ the same psychophysical paradigm, we combined their results for a comprehensive view of how auditory and tactile signals interact and how these interactions depend on the perceptual task. Together, our results show that temporal frequency representations are perceptually linked regardless of the attended modality. In contrast, audio-tactile loudness interactions depend on the attended modality: Tactile distractors influence judgments of auditory intensity, but judgments of tactile intensity are impervious to auditory distraction. Lastly, we show that audio-tactile loudness interactions depend critically on stimulus timing, while pitch interactions do not. These results reveal that auditory and tactile inputs are combined differently depending on the perceptual task. That distinct rules govern the integration of auditory and tactile signals in pitch and loudness perception implies that the two are mediated by separate neural mechanisms. These findings underscore the complexity and specificity of multisensory interactions.

  10. Real-Time Conversion of Stereo Audio to 5.1 Channel Audio for Providing Realistic Sounds

    Directory of Open Access Journals (Sweden)

    Chan Jun Chun

    2009-12-01

    Full Text Available In this paper, we address issues associated with the real-time implementation of upmixing stereo audio into 5.1 channel audio in order to improve audio realism. First, we review four different upmixing methods, including a passive surround decoding method, a least-meansquare based upmixing method, a principal component analysis based upmixing method, and an adaptive panning method. After that, we implement a simulator that includes the upmixingmethods and audio controls to play both stereo and upmixed 5.1 channel audio signals. Finally, we carry out a MUSHRA test to compare the quality of the upmixed 5.1 channel audio signals to that of the original stereo audio signal. It is shown from the test that the upmixed 5.1 channel audio signals generated by the four different upmixing methods are preferred to the original stereo audio signals.

  11. Evaluation of Perceived Spatial Audio Quality

    Directory of Open Access Journals (Sweden)

    Jan Berg

    2006-04-01

    Full Text Available The increased use of audio applications capable of conveying enhanced spatial quality puts focus on how such a quality should be evaluated. Different approaches to evaluation of perceived quality are briefly discussed and a new technique is introduced. In a series of experiment, attributes were elicited from subjects, tested and subsequently used for derivation of evaluation scales that were feasible for subjective evaluation of the spatial quality of certain multichannel stimuli. The findings of these experiments led to the development of a novel method for evaluation of spatial audio in surround sound systems. Parts of the method were subsequently implemented in the OPAQUE software prototype designed to facilitate the elicitation process. The prototype was successfully tested in a pilot experiment. The experiments show that attribute scales derived from subjects' personal constructs are functional for evaluation of perceived spatial audio quality. Finally, conclusions on the importance of spatial quality evaluation of new applications are made.

  12. A high efficiency PWM CMOS class-D audio power amplifier

    International Nuclear Information System (INIS)

    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 mm2. With the good performance, the class-D audio power amplifier can be applied to several audio power systems.

  13. 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.

  14. The research on image encryption method based on parasitic audio watermark

    Science.gov (United States)

    Gao, Pei-pei; Zhu, Yao-ting; Zhang, Shi-tao

    2010-11-01

    In order to improve image encryption strength, an image encryption method based on parasitic audio watermark was proposed in this paper, which relies on double messages such as image domain and speech domain to do image encryption protection. The method utilizes unique Chinese phonetics synthesis algorithm to complete audio synthesis with embedded text, then separate this sentence information into prosodic phrase, obtains complete element set of initial consonant and compound vowel that reflects audio feature of statement. By sampling and scrambling the initial consonant and compound vowel element, synthesizing them with image watermark, and embedding the compound into the image to be encrypted in frequency domain, the processed image contains image watermark information and parasitizes audio feature information. After watermark extraction, using the same phonetics synthesis algorithm the audio information is synthesized and compared with the original. Experiments show that any decryption method in image domain or speech domain could not break encryption protection and image gains higher encryption strength and security level by double encryption.

  15. 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.

  16. Information Security using Audio Steganography -A Survey

    Directory of Open Access Journals (Sweden)

    B. Santhi

    2012-07-01

    Full Text Available The most important application of internet is data transmission. Unfortunately this is less secured because of advanced hacking technologies. So, for secured data transmission we make use of steganography. This is the art of hiding information where the existence of data is unknown. Any medium like music, video, text, speech, etc can be used. In this study, the selected medium is audio. This study discusses about the existing audio steganographic techniques along with their advantages and limitations. Also an algorithm implementing parity and LSB methods is proposed. This mitigates the limitations of the existing methods discussed, thus increasing security and reducing computational load and code complexity.

  17. 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...

  18. 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 by a...... qualitative analysis of subject responses. Distraction ratings were collected for one hundred randomly created audio-on-audio interference situations with music target and interferer programs. The selected features were related to the overall loudness, loudness ratio, perceptual evaluation of audio source...

  19. 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.

  20. 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...

  1. Flow control using audio tones in resonant microfluidic networks: towards cell-phone controlled lab-on-a-chip devices.

    Science.gov (United States)

    Phillips, Reid H; Jain, Rahil; Browning, Yoni; Shah, Rachana; Kauffman, Peter; Dinh, Doan; Lutz, Barry R

    2016-08-16

    Fluid control remains a challenge in development of portable lab-on-a-chip devices. Here, we show that microfluidic networks driven by single-frequency audio tones create resonant oscillating flow that is predicted by equivalent electrical circuit models. We fabricated microfluidic devices with fluidic resistors (R), inductors (L), and capacitors (C) to create RLC networks with band-pass resonance in the audible frequency range available on portable audio devices. Microfluidic devices were fabricated from laser-cut adhesive plastic, and a "buzzer" was glued to a diaphragm (capacitor) to integrate the actuator on the device. The AC flowrate magnitude was measured by imaging oscillation of bead tracers to allow direct comparison to the RLC circuit model across the frequency range. We present a systematic build-up from single-channel systems to multi-channel (3-channel) networks, and show that RLC circuit models predict complex frequency-dependent interactions within multi-channel networks. Finally, we show that adding flow rectifying valves to the network creates pumps that can be driven by amplified and non-amplified audio tones from common audio devices (iPod and iPhone). This work shows that RLC circuit models predict resonant flow responses in multi-channel fluidic networks as a step towards microfluidic devices controlled by audio tones. PMID:27416111

  2. A high performance switching audio amplifier using sliding mode control

    OpenAIRE

    Pillonnet, Gael; Cellier, Rémy; Abouchi, Nacer; Chiollaz, Monique

    2008-01-01

    International audience The switching audio amplifiers are widely used in various portable and consumer electronics due to their high efficiency, but suffers from low audio performances due to inherent nonlinearity. This paper presents an integrated class D audio amplifier with low consumption and high audio performances. It includes a power stage and an efficient control based on sliding mode technique. This monolithic class D amplifier is capable of delivering up to 1W into 8Ω load at les...

  3. Learning bimodal structure in audio-visual data

    OpenAIRE

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

    2009-01-01

    A novel model is presented to learn bimodally informative structures from audio-visual signals. The signal is represented as a sparse sum of audio- visual kernels. Each kernel is a bimodal function consisting of synchronous snippets of an audio waveform and a spatio-temporal visual basis function. To represent an audio-visual signal, the kernels can be positioned independently and arbitrarily in space and time. The proposed algorithm uses unsupervised learning to form dicti...

  4. Assessment of spatial audio quality based on sound attributes

    OpenAIRE

    LE BAGOUSSE, Sarah; Paquier, Mathieu; Colomes, Catherine

    2012-01-01

    International audience Spatial audio technologies become very important in audio broadcast services. But, there is a lack of methods for evaluating spatial audio quality. Standards do not take into account spatial dimension of sound and assessments are limited to the overall quality particularly in the context of audio coding. Through different elicitation methods, a long list of attributes has been established to characterize sound but it is difficult to include them in a listening test. ...

  5. 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.

  6. Study of audio speakers containing ferrofluid

    International Nuclear Information System (INIS)

    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

  7. 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.

  8. Audio-visual classification video browser

    OpenAIRE

    Scott, David; Zhang, ZhenXing; Albatal, Rami; McGuinness, Kevin; Acar, Esra; Hopfgartner, Frank; Gurrin, Cathal; O'Connor, Noel; Smeaton, Alan

    2014-01-01

    This paper presents our third participation in the Video Browser Showdown. Building on the experience that we gained while participating in this event, we compete in the 2014 showdown with a more advanced browsing system based on incorporating several audio- visual retrieval techniques. This paper provides a short overview of the features and functionality of our new system.

  9. 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. In...

  10. Structuring Broadcast Audio for Information Access

    Science.gov (United States)

    Gauvain, Jean-Luc; Lamel, Lori

    2003-12-01

    One rapidly expanding application area for state-of-the-art speech recognition technology is the automatic processing of broadcast audiovisual data for information access. Since much of the linguistic information is found in the audio channel, speech recognition is a key enabling technology which, when combined with information retrieval techniques, can be used for searching large audiovisual document collections. Audio indexing must take into account the specificities of audio data such as needing to deal with the continuous data stream and an imperfect word transcription. Other important considerations are dealing with language specificities and facilitating language portability. At Laboratoire d'Informatique pour la Mécanique et les Sciences de l'Ingénieur (LIMSI), broadcast news transcription systems have been developed for seven languages: English, French, German, Mandarin, Portuguese, Spanish, and Arabic. The transcription systems have been integrated into prototype demonstrators for several application areas such as audio data mining, structuring audiovisual archives, selective dissemination of information, and topic tracking for media monitoring. As examples, this paper addresses the spoken document retrieval and topic tracking tasks.

  11. Building Digital Audio Preservation Infrastructure and Workflows

    Science.gov (United States)

    Young, Anjanette; Olivieri, Blynne; Eckler, Karl; Gerontakos, Theodore

    2010-01-01

    In 2009 the University of Washington (UW) Libraries special collections received funding for the digital preservation of its audio indigenous language holdings. The university libraries, where the authors work in various capacities, had begun digitizing image and text collections in 1997. Because of this, at the onset of the project, workflows (a…

  12. Utilization of Nonlinear Converters for Audio Amplification

    DEFF Research Database (Denmark)

    Iversen, Niels; Birch, Thomas; Knott, Arnold

    2012-01-01

    introduction of non-linear converters for audio amplication defeats this limitation. A Cuk converter, designed to deliver an AC peak output voltage twice the supply voltage, is presented in this paper. A 3V prototype has been developed to prove the concept. The prototype shows that it is possible to achieve an...

  13. Providing Students with Formative Audio Feedback

    Science.gov (United States)

    Brearley, Francis Q.; Cullen, W. Rod

    2012-01-01

    The provision of timely and constructive feedback is increasingly challenging for busy academics. Ensuring effective student engagement with feedback is equally difficult. Increasingly, studies have explored provision of audio recorded feedback to enhance effectiveness and engagement with feedback. Few, if any, of these focus on purely formative…

  14. 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…

  15. Predistortion of a Bidirectional Cuk Audio Amplifier

    DEFF Research Database (Denmark)

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

    2014-01-01

    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....

  16. Structuring Broadcast Audio for Information Access

    Directory of Open Access Journals (Sweden)

    Jean-Luc Gauvain

    2003-02-01

    Full Text Available One rapidly expanding application area for state-of-the-art speech recognition technology is the automatic processing of broadcast audiovisual data for information access. Since much of the linguistic information is found in the audio channel, speech recognition is a key enabling technology which, when combined with information retrieval techniques, can be used for searching large audiovisual document collections. Audio indexing must take into account the specificities of audio data such as needing to deal with the continuous data stream and an imperfect word transcription. Other important considerations are dealing with language specificities and facilitating language portability. At Laboratoire d′Informatique pour la Mécanique et les Sciences de l′Ingénieur (LIMSI, broadcast news transcription systems have been developed for seven languages: English, French, German, Mandarin, Portuguese, Spanish, and Arabic. The transcription systems have been integrated into prototype demonstrators for several application areas such as audio data mining, structuring audiovisual archives, selective dissemination of information, and topic tracking for media monitoring. As examples, this paper addresses the spoken document retrieval and topic tracking tasks.

  17. 47 CFR 73.403 - Digital audio broadcasting service requirements.

    Science.gov (United States)

    2010-10-01

    ... 47 Telecommunication 4 2010-10-01 2010-10-01 false Digital audio broadcasting service requirements. 73.403 Section 73.403 Telecommunication FEDERAL COMMUNICATIONS COMMISSION (CONTINUED) BROADCAST RADIO SERVICES RADIO BROADCAST SERVICES Digital Audio Broadcasting § 73.403 Digital audio broadcasting...

  18. 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...

  19. Switching-mode Audio Power Amplifiers with Direct Energy Conversion

    DEFF Research Database (Denmark)

    Ljusev, Petar; Andersen, Michael Andreas E.

    2005-01-01

    This paper presents a new class of switching-mode audio power amplifiers, which are capable of direct energy conversion from the AC mains to the audio output. They represent an ultimate integration of a switching-mode power supply and a Class D audio power amplifier, where the intermediate DC bus...... component count, volume and cost....

  20. 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...

  1. The KUSC Classical Music Dataset for Audio Key Finding

    Directory of Open Access Journals (Sweden)

    Ching-Hua Chuan

    2014-08-01

    Full Text Available In this paper, we present a benchmark dataset based on the KUSC classical music collection and provide baseline key-finding comparison results. Audio key finding is a basic music information retrieval task; it forms an essential component of systems for music segmentation, similarity assessment, and mood detection. Due to copyright restrictions and a labor-intensive annotation process, audio key finding algorithms have only been evaluated using small proprietary datasets to date. To create a common base for systematic comparisons, we have constructed a dataset comprising of more than 3,000 excerpts of classical music. The excerpts are made publicly accessible via commonly used acoustic features such as pitch-based spectrograms and chromagrams. We introduce a hybrid annotation scheme that combines the use of title keys with expert validation and correction of only the challenging cases. The expert musicians also provide ratings of key recognition difficulty. Other meta-data include instrumentation. As demonstration of use of the dataset, and to provide initial benchmark comparisons for evaluating new algorithms, we conduct a series of experiments reporting key determination accuracy of four state-of-the-art algorithms. We further show the importance of considering factors such as estimated tuning frequency, key strength or confidence value, and key recognition difficulty in key finding. In the future, we plan to expand the dataset to include meta-data for other music information retrieval tasks.

  2. Survey Musik und Medien 2012: Audio Media Usage in Germany - Audio Repertoires by birth cohorts

    OpenAIRE

    Lepa, Steffen

    2013-01-01

    Audio Repertoires are widespread patterns regarding the use of audiotechnologies in everyday life which may also be interpreted as “user types”. They were identified in Survey Musik und Medien 2012 based on the nationwide collected representative Audio Usage Data. Nowadays, people listen to music by means of various different devices, infrastructures and technologies. Furthermore, people often tend to combine those options within their daily routines. Therefore, it is reasonable to analyz...

  3. Audio Oracle: A New Algorithm for Fast Learning of Audio Structures

    OpenAIRE

    Dubnov, Shlomo; Assayag, Gerard; Cont, Arshia

    2007-01-01

    International audience In this paper we present a new method for indexing of audio data in terms of repeating sub-clips of variable length that we call audio factors. The new structure allows fast retrieval and recombination of sub-clips in a manner that assures continuity between splice points. The resulting structure accomplishes effectively a new method for texture synthesis, where the amount of innovation is controlled by one of the synthesis parameters. In the paper we present the new...

  4. Audio-visual synchrony and feature-selective attention co-amplify early visual processing.

    Science.gov (United States)

    Keitel, Christian; Müller, Matthias M

    2016-05-01

    Our brain relies on neural mechanisms of selective attention and converging sensory processing to efficiently cope with rich and unceasing multisensory inputs. One prominent assumption holds that audio-visual synchrony can act as a strong attractor for spatial attention. Here, we tested for a similar effect of audio-visual synchrony on feature-selective attention. We presented two superimposed Gabor patches that differed in colour and orientation. On each trial, participants were cued to selectively attend to one of the two patches. Over time, spatial frequencies of both patches varied sinusoidally at distinct rates (3.14 and 3.63 Hz), giving rise to pulse-like percepts. A simultaneously presented pure tone carried a frequency modulation at the pulse rate of one of the two visual stimuli to introduce audio-visual synchrony. Pulsed stimulation elicited distinct time-locked oscillatory electrophysiological brain responses. These steady-state responses were quantified in the spectral domain to examine individual stimulus processing under conditions of synchronous versus asynchronous tone presentation and when respective stimuli were attended versus unattended. We found that both, attending to the colour of a stimulus and its synchrony with the tone, enhanced its processing. Moreover, both gain effects combined linearly for attended in-sync stimuli. Our results suggest that audio-visual synchrony can attract attention to specific stimulus features when stimuli overlap in space. PMID:26226930

  5. Survey Musik und Medien 2012: Audio Media Usage in Germany - Audio Sources - Audio Sources used in 2012 - Selective Adopters

    OpenAIRE

    Lepa, Steffen

    2013-01-01

    Where did everyday music come from in 2012? Audio Sources describe those distribution channels by means of which music is purchased, archived and made accessible. This includes physical recordings (CD, LP, MC etc.), electronic services in terms of downloading and streaming of digital music (iTunes, last.fm, Spotify etc.) as well as traditional radio reception and last but not least musical content on websites or digital storage media. Typically born between 1963 and 1980, the Selective ad...

  6. Survey Musik und Medien 2012: Audio Media Usage in Germany - Audio Sources - Audio Sources used in 2012 - Digital Mobilists

    OpenAIRE

    Lepa, Steffen

    2013-01-01

    Where did everyday music come from in 2012? Audio Sources describe those distribution channels by means of which music is purchased, archived and made accessible. This includes physical recordings (CD, LP, MC etc.), electronic services in terms of downloading and streaming of digital music (iTunes, last.fm, Spotify etc.) as well as traditional radio reception and last but not least musical content on websites or digital storage media. Typically born between 1979 and 1998, the Digital Mobi...

  7. Survey Musik und Medien 2012: Audio Media Usage in Germany - Audio Sources - Audio Sources used in 2012

    OpenAIRE

    Lepa, Steffen

    2013-01-01

    Where did everyday music come from in 2012? Audio Sources describe those distribution channels by means of which music is purchased, archived and made accessible. This includes physical recordings (CD, LP, MC etc.), electronic services in terms of downloading and streaming of digital music (iTunes, last.fm, Spotify etc.) as well as traditional radio reception and last but not least musical content on websites or digital storage media. How do the Germans listen to music nowadays? Survey Mus...

  8. A Unified Approach to Real Time Audio-to-Score and Audio-to-Audio Alignment Using Sequential Montecarlo Inference Techniques

    OpenAIRE

    Montecchio, Nicola; Cont, Arshia

    2011-01-01

    International audience We present a methodology for the real time alignment of music signals using sequential Montecarlo inference techniques. The alignment problem is formulated as the state tracking of a dynamical system, and differs from traditional Hidden Markov Model - Dynamic Time Warping based systems in that the hidden state is continuous rather than discrete. The major contribution of this paper is addressing both problems of audio-to-score and audio-to-audio alignment within the ...

  9. Three-dimensional audio using loudspeakers

    Science.gov (United States)

    Gardner, William G.

    1997-12-01

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

  10. Audio Mining with emphasis on Music Genre Classification

    DEFF Research Database (Denmark)

    Meng, Anders

    2004-01-01

    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......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...... etc. is receiving quite a lot of attention. The first breakthough in audio mining was created by MuscleFish in 1996, a simple audio retrieval system. With the increasing amount of audio material being accessible through the web, e.g. Apple's iTunes (700,000+ songs), Sony, Amazon, new methods...

  11. Unsupervised incremental online learning and prediction of musical audio signals

    DEFF Research Database (Denmark)

    Marxer, Richard; Purwins, Hendrik

    2016-01-01

    the next event in a musical sequence, given as audio input. The flow of the system is as follows: 1) segmentation by onset detection, 2) timbre representation of each segment by Mel frequency cepstrum coefficients, 3) discretization by incremental clustering, yielding a tree of different sound classes (e.......g. timbre categories/instruments) that can grow or shrink on the fly driven by the instantaneous sound events, resulting in a discrete symbol sequence, 4) extraction of statistical regularities of the symbol sequence, using hierarchical N-grams and the newly introduced conceptual Boltzmann machine...... that adapt to the dynamically changing clustering tree in 3), and 5) prediction of the next sound event in the sequence, given the last n previous events. The system’s robustness is assessed with respect to complexity and noisiness of the signal. Clustering in isolation yields an adjusted Rand index (ARI...

  12. Digital audio and video broadcasting by satellite

    Science.gov (United States)

    Yoshino, Takehiko

    In parallel with the progress of the practical use of satellite broadcasting and Hi-Vision or high-definition television technologies, research activities are also in progress to replace the conventional analog broadcasting services with a digital version. What we call 'digitalization' is not a mere technical matter but an important subject which will help promote multichannel or multimedia applications and, accordingly, can change the old concept of mass media, such as television or radio. NHK Science and Technical Research Laboratories has promoted studies of digital bandwidth compression, transmission, and application techniques. The following topics are covered: the trend of digital broadcasting; features of Integrated Services Digital Broadcasting (ISDB); compression encoding and transmission; transmission bit rate in 12 GHz band; number of digital TV transmission channels; multichannel pulse code modulation (PCM) audio broadcasting system via communication satellite; digital Hi-Vision broadcasting; and development of digital audio broadcasting (DAB) for mobile reception in Japan.

  13. A Physiologically Inspired Method for Audio Classification

    Directory of Open Access Journals (Sweden)

    David V. Anderson

    2005-06-01

    Full Text Available We explore the use of physiologically inspired auditory features with both physiologically motivated and statistical audio classification methods. We use features derived from a biophysically defensible model of the early auditory system for audio classification using a neural network classifier. We also use a Gaussian-mixture-model (GMM-based classifier for the purpose of comparison and show that the neural-network-based approach works better. Further, we use features from a more advanced model of the auditory system and show that the features extracted from this model of the primary auditory cortex perform better than the features from the early auditory stage. The features give good classification performance with only one-second data segments used for training and testing.

  14. Audio Steganography Techniques-A Survey

    Directory of Open Access Journals (Sweden)

    Navneet Kaur

    2014-06-01

    Full Text Available we can communicate with each other by passing messages which is not secure, but we make a communication be kept secret by embedding the message into carrier or by special tools such as invisible ink, microdots etc. Steganography is the science that involves communicating secret data in an appropriate carrier which is used from hundreds of years. In digital age new techniques of hiding the data inside the carrier are invented which are known as digital steganography. Nowadays, the carrier of the message can be an image, audio, video or a text file. In this paper we have purposed a method to enhance the security level in audio steganography and also improve the quality by making 2-level steganography.

  15. 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.

  16. Museum audio guides as an accessibility enhancer

    OpenAIRE

    Martins, Cláudia Susana Nunes

    2012-01-01

    Accessibility to museums is enhanced by various types of cultural mediation, such as the use of audio guides, which consist of a means for innovative mediation put forth to make the museum visit more autonomous and simultaneously replace the traditional guided visit. Their use is integrated in the tendency for museum democratisation felt in Europe between the 60s and the 80s of the 20th century, especially with the development of educational services at museums and their opening to schools. I...

  17. 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.

  18. Beamforming Techniques for Multichannel audio Signal Separation

    OpenAIRE

    Adel, Hidri; Souad, Meddeb; Alaqeeli, Abdulqadir; Hamid, Amiri

    2012-01-01

    Beamforming is a signal processing technique. It has been studied in many areas such as radar, sonar, seismology and wireless communications, to name but a few. It can be used for a myriad of purposes, such as detecting the presence of a signal, estimating the direction of arrival, and enhancing a desired signal from its measurements corrupted by noise, competing sources and reverberation. Actually, Beamforming has been adopted by the audio research society, mostly to separate or extract spee...

  19. Audio Modeling based on Delayed Sinusoids

    OpenAIRE

    Boyer, Remy; Abed-Meraim, Karim

    2004-01-01

    In this work, we present an evolution of the DDS (Damped & Delayed Sinusoidal) model introduced within the framework of the general signal modeling. This model is named the Partial Damped & Delayed Sinusoidal (PDDS) model and takes into account a single time delay parameter for a set (sum) of damped sinusoids. This modi- ¯cation is more consistent with the transient audio modeling problem. We show the validity of this approach by compari- son with the well-known EDS (Exponentially Damped Sinu...

  20. Aplicacion de RA-Audio I

    OpenAIRE

    Camacho Melero, Felipe

    2011-01-01

    Este documento explica el diseño y la implementación de un proyecto que ofrece realidad aumentada de audio. Es una combinación de tres servicios diferentes: red social, streaming y realidad aumentada. El sistema se basa en una arquitectura cliente-servidor con una base de datos donde se almacena la información. La parte cliente está implementada para ser accesible mediante una web y desde dispositivos con el sistema operativo Android.

  1. Capacity-optimized mp2 audio watermarking

    Science.gov (United States)

    Steinebach, Martin; Dittmann, Jana

    2003-06-01

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

  2. Synthecology: sound use of audio in teleimmersion

    Science.gov (United States)

    Baum, Geoffrey; Gotsis, Marientina; Chang, Benjamin; Drinkwater, Robb; St. Clair, Dan

    2006-02-01

    This paper examines historical audio applications used to provide real-time immersive sound for CAVE TM environments and discusses their relative strengths and weaknesses. We examine and explain issues of providing spatialized sound immersion in real-time virtual environments (VEs), some problems with currently used sound servers, and a set of requirements for an 'ideal' sound server. We present the initial configuration of a new cross-platform sound server solution using open source software and the Open Sound Control (OSC) specification for the creation of real-time spatialized audio with CAVE applications, specifically Ygdrasil (Yg) environments. The application, aNother Sound Server (NSS) establishes an application interface (API) using OSC, a logical server layer implemented in Python, and an audio engine using SuperCollider (SC). We discuss spatialization implementation and other features. Finally, we document the Synthecology project which premiered at WIRED NEXTFEST 2005 and was the first VE to use NSS. We also discuss various techniques that enhance presence in networked VEs, as well as possible and planned extensions of NSS.

  3. Le registrazioni audio dell’archivio Luigi Nono di Venezia

    Directory of Open Access Journals (Sweden)

    Luca Cossettini

    2009-11-01

    Full Text Available The audio recordings of the Luigi Nono Archive in Venice: guidelines for preservation and critical edition of audio documentsStudying audio recordings brings us back to ancient source verification problems that too often one thinks are overcome by the technical reproduction of sound. Au-dio signal is “fixed” on a specific carrier (tape, disc etc with a specific audio format (speed, number of tracks etc; the choice of support and format during the first “memorizing” process and the following copying processes is a subjective and, in case of copying, an interpretative operation conducted within a continuously evolv-ing audio technology. What we listen to today is the result of a transmission process that unavoidably transforms the original acoustic event and the documents that memorize it. Audio recording is no way a timeless and immutable fixing process. It is therefore necessary to study the transmission processes and to reconstruct the au-dio document tradition. The re-recording of the tapes of the Archivio Luigi Nono, conducted by the Audio Labs of the DAMS Musica of the University of Udine, of-fers clear examples of the technical and musicological interpretative problems one can find when he works with audio recordings.

  4. Applications of ENF criterion in forensic audio, video, computer and telecommunication analysis.

    Science.gov (United States)

    Grigoras, Catalin

    2007-04-11

    This article reports on the electric network frequency criterion as a means of assessing the integrity of digital audio/video evidence and forensic IT and telecommunication analysis. A brief description is given to different ENF types and phenomena that determine ENF variations. In most situations, to reach a non-authenticity opinion, the visual inspection of spectrograms and comparison with an ENF database are enough. A more detailed investigation, in the time domain, requires short time windows measurements and analyses. The stability of the ENF over geographical distances has been established by comparison of synchronized recordings made at different locations on the same network. Real cases are presented, in which the ENF criterion was used to investigate audio and video files created with secret surveillance systems, a digitized audio/video recording and a TV broadcasted reportage. By applying the ENF Criterion in forensic audio/video analysis, one can determine whether and where a digital recording has been edited, establish whether it was made at the time claimed, and identify the time and date of the registering operation. PMID:16884872

  5. C Implementation and comparison of companding and silence audio compression techniques

    OpenAIRE

    Kruti Dangarwala; Jigar Shah

    2010-01-01

    Just about all the newest living room audio-video electronics and PC multimedia products being designed today will incorporate some form of compressed digitized-audio processing capability. Audio compression reduces the bit rate required to represent an analog audio signal while maintaining the perceived audio quality. Discarding inaudible data reduces the storage, transmission and compute requirements of handling high-quality audio files. This paper covers wave audio file format and algorith...

  6. Differences in Human Audio Localization Performance between a HRTF- and a non-HRTF Audio System

    DEFF Research Database (Denmark)

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

    2013-01-01

    Spatial audio solutions have been around for a long time in real-time applications, but yielding spatial cues that more closely simulate real life accuracy has been a computational issue, and has often been solved by hardware solutions. This has long been a restriction, but now with more powerful...... computers this is becoming a lesser and lesser concern and software solutions are now applicable. Most current virtual environment applications do not take advantage of these im- plementations of accurate spatial cues, however. This paper compares a common implementation of spatial audio and a head......-related transfer function (HRTF) system implemen- tation in a study in relation to precision, speed and navi- gational performance in localizing audio sources in a virtual environment. We found that a system using HRTFs is signif- icantly better at all three performance tasks than a system using panning....

  7. An audio-based sports video segmentation and event detection algorithm

    OpenAIRE

    Baillie, M.; Jose, J.M.

    2004-01-01

    In this paper, we present an audio-based event detection algorithm shown to be effective when applied to Soccer video. The main benefit of this approach is the ability to recognise patterns that display high levels of crowd response correlated to key events. The soundtrack from a Soccer sequence is first parameterised using Mel-frequency Cepstral coefficients. It is then segmented into homogenous components using a windowing algorithm with a decision process based on Bayesian model selection....

  8. Phase recovery in NMF for audio source separation: an insightful benchmark

    OpenAIRE

    Magron, Paul; Badeau, Roland; David, Bertrand

    2016-01-01

    Nonnegative Matrix Factorization (NMF) is a powerful tool for decomposing mixtures of audio signals in the Time-Frequency (TF) domain. In applications such as source separation, the phase recovery for each extracted component is a major issue since it often leads to audible artifacts. In this paper, we present a methodology for evaluating various NMF-based source separation techniques involving phase reconstruction. For each model considered, a comparison between two approaches (blind separat...

  9. Android real-time audio communications over local wireless

    OpenAIRE

    Belda Ortega, Román; Arce Vila, Pau; De Fez Lava, Ismael; Fraile Gil, Francisco; Guerri Cebollada, Juan Carlos

    2012-01-01

    This paper describes an Android mobile application that allows voice communications through short-range wireless networks, mainly Bluetooth and Wi-Fi. The application is able to replicate as close as possible the behavior of a two-way radio device. The application is designed to receive audio streams from multiple devices simultaneously and to send them. The main design considerations of the application, such as audio recording and playing, audio coding or data transmission, are explained thr...

  10. Exploring relationships between audio features and emotion in music

    OpenAIRE

    Laurier, Cyril; Lartillot, Olivier; Eerola, Tuomas; Toiviainen, Petri

    2009-01-01

    In this paper, we present an analysis of the associations between emotion categories and audio features automatically extracted from raw audio data. This work is based on 110 excerpts from film soundtracks evaluated by 116 listeners. This data is annotated with 5 basic emotions (fear, anger, happiness, sadness, tenderness) on a 7 points scale. Exploiting state-of-the-art Music Information Retrieval (MIR) techniques, we extract audio features of different kind: timbral, rhythmic and tonal. Amo...

  11. Audio Classical Composer Identification by Deep Neural Network

    OpenAIRE

    Hu, Zhen; Fu, Kun; Zhang, Changshui

    2013-01-01

    Audio Classical Composer Identification (ACC) is an important problem in Music Information Retrieval (MIR) which aims at identifying the composer for audio classical music clips. The famous annual competition, Music Information Retrieval Evaluation eXchange (MIREX), also takes it as one of the four training&testing tasks. We built a hybrid model based on Deep Belief Network (DBN) and Stacked Denoising Autoencoder (SDA) to identify the composer from audio signal. As a matter of copyright, spon...

  12. 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...

  13. Standardization Promotes the Quality of Meteorological Audio & Video Service

    Institute of Scientific and Technical Information of China (English)

    2011-01-01

    As an important part of meteorological sector and a critical basis for enhancing the capability of meteorological disaster prevention and mitigation and climate change response,the meteorological standardization is a significant support for facilitating the good and quick development of meteorological sector.Huafeng Group,as a leading enterprise of meteorological audio & video service,has,for years,attached much importance to employing the standardization of meteorological audio & video service to improve its management level and quality of programs,enhance the quality of meteorological audio & video service,build the brand image,cultivate the highlevel backbone personnel,and facilitate the sustainable development of meteorological audio & video service.

  14. 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.

  15. 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

  16. Audio marketing v ČR

    OpenAIRE

    Timanov, Vladimir

    2015-01-01

    The aim of the work is processing and evaluation of the investment project. The project implies an establishment of the firm in Czech Republic. The branch of the entrepreneurship is sensory marketing or audio-visual marketing. The essence of this field of the marketing is encouragement of sales through the influence on emotional side of the client. Components of the work are market research, analysis of the competitors in this sphere, and the financial plan. As a result, the work will be stru...

  17. Presence and the utility of audio spatialization

    DEFF Research Database (Denmark)

    Bormann, Karsten

    2005-01-01

    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 the...... playing radio would be stationary (a passive sound source) while subjects searched for some other object. Independent of this, the music emitted by the radio would be either fully spatialized or directional but nonattenuated. Findings include that for subjects searching for the active sound source, being...... visuals....

  18. 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 that...

  19. On Steganography in Lost Audio Packets

    CERN Document Server

    Mazurczyk, Wojciech; Szczypiorski, Krzysztof

    2011-01-01

    The paper presents a new hidden data insertion procedure based on estimated probability of the remaining time of the call for steganographic method called LACK (Lost Audio PaCKets steganography). LACK provides hidden communication for real-time services like Voice over IP. The analytical results presented in this paper concern the influence of LACK's hidden data insertion procedures on the method's impact on quality of voice transmission and its resistance to steganalysis. The proposed hidden data insertion procedure is also compared to previous steganogram insertion approach based on estimated remaining average call duration.

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

    Science.gov (United States)

    Carter, Curtis W.

    2012-01-01

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

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

    Science.gov (United States)

    George, Rohini

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

  2. Simple Solutions for Space Station Audio Problems

    Science.gov (United States)

    Wood, Eric

    2016-01-01

    Throughout this summer, a number of different projects were supported relating to various NASA programs, including the International Space Station (ISS) and Orion. The primary project that was worked on was designing and testing an acoustic diverter which could be used on the ISS to increase sound pressure levels in Node 1, a module that does not have any Audio Terminal Units (ATUs) inside it. This acoustic diverter is not intended to be a permanent solution to providing audio to Node 1; it is simply intended to improve conditions while more permanent solutions are under development. One of the most exciting aspects of this project is that the acoustic diverter is designed to be 3D printed on the ISS, using the 3D printer that was set up earlier this year. Because of this, no new hardware needs to be sent up to the station, and no extensive hardware testing needs to be performed on the ground before sending it to the station. Instead, the 3D part file can simply be uploaded to the station's 3D printer, where the diverter will be made.

  3. 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.

  4. Subjective evaluation and electroacoustic theoretical validation of a new approach to audio upmixing

    Science.gov (United States)

    Usher, John S.

    Audio signal processing systems for converting two-channel (stereo) recordings to four or five channels are increasingly relevant. These audio upmixers can be used with conventional stereo sound recordings and reproduced with multichannel home theatre or automotive loudspeaker audio systems to create a more engaging and natural-sounding listening experience. This dissertation discusses existing approaches to audio upmixing for recordings of musical performances and presents specific design criteria for a system to enhance spatial sound quality. A new upmixing system is proposed and evaluated according to these criteria and a theoretical model for its behavior is validated using empirical measurements. The new system removes short-term correlated components from two electronic audio signals using a pair of adaptive filters, updated according to a frequency domain implementation of the normalized-least-means-square algorithm. The major difference of the new system with all extant audio upmixers is that unsupervised time-alignment of the input signals (typically, by up to +/-10 ms) as a function of frequency (typically, using a 1024-band equalizer) is accomplished due to the non-minimum phase adaptive filter. Two new signals are created from the weighted difference of the inputs, and are then radiated with two loudspeakers behind the listener. According to the consensus in the literature on the effect of interaural correlation on auditory image formation, the self-orthogonalizing properties of the algorithm ensure minimal distortion of the frontal source imagery and natural-sounding, enveloping reverberance (ambiance) imagery. Performance evaluation of the new upmix system was accomplished in two ways: Firstly, using empirical electroacoustic measurements which validate a theoretical model of the system; and secondly, with formal listening tests which investigated auditory spatial imagery with a graphical mapping tool and a preference experiment. Both electroacoustic

  5. Design of FIR Filter Using Particle Swarm Optimization Algorithm for Audio Processing

    Directory of Open Access Journals (Sweden)

    Amanpreet Kaur

    2012-08-01

    Full Text Available In this paper, an optimal design of linear phase digital finiteimpulse response (FIR filter using Modified Particle SwarmOptimization (MPSO has been presented. In the designprocess, the filter length, pass band and stop band frequencies,pass band and stop band ripple sizes are specified. Sometimesthe gradient based optimization techniques are not effectivefor designing filter. An evolutionary method is introduced tofind the optimal solution of FIR filter design problem. MPSOis a global stochastic searching technique that can find out theglobal optima of the problem. A simulation results reveals theoptimization efficacy of the algorithm for the solution of,highly non-linear, and constrained filter design problems. Thedesigned filter is then applied on the audio application for upsampling of the audio signal. MATLAB toolkit functions areused for implementation of proposed algorithm.

  6. 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...

  7. 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

  8. 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...

  9. 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...

  10. The Practical Audio-Visual Handbook for Teachers.

    Science.gov (United States)

    Scuorzo, Herbert E.

    The use of audio/visual media as an aid to instruction is a common practice in today's classroom. Most teachers, however, have little or no formal training in this field and rarely a knowledgeable coordinator to help them. "The Practical Audio-Visual Handbook for Teachers" discusses the types and mechanics of many of these media forms and proposes…

  11. Beyond Podcasting: Creative Approaches to Designing Educational Audio

    Science.gov (United States)

    Middleton, Andrew

    2009-01-01

    This paper discusses a university-wide pilot designed to encourage academics to creatively explore learner-centred applications for digital audio. Participation in the pilot was diverse in terms of technical competence, confidence and contextual requirements and there was little prior experience of working with digital audio. Many innovative…

  12. Effect of Audio vs. Video on Aural Discrimination of Vowels

    Science.gov (United States)

    McCrocklin, Shannon

    2012-01-01

    Despite the growing use of media in the classroom, the effects of using of audio versus video in pronunciation teaching has been largely ignored. To analyze the impact of the use of audio or video training on aural discrimination of vowels, 61 participants (all students at a large American university) took a pre-test followed by two training…

  13. 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 comp

  14. Using Audio Books to Improve Reading and Academic Performance

    Science.gov (United States)

    Montgomery, Joel R.

    2009-01-01

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

  15. A Case Study on Audio Feedback with Geography Undergraduates

    Science.gov (United States)

    Rodway-Dyer, Sue; Knight, Jasper; Dunne, Elizabeth

    2011-01-01

    Several small-scale studies have suggested that audio feedback can help students to reflect on their learning and to develop deep learning approaches that are associated with higher attainment in assessments. For this case study, Geography undergraduates were given audio feedback on a written essay assignment, alongside traditional written…

  16. Use of Audio Modification in Science Vocabulary Assessment

    Science.gov (United States)

    Adiguzel, Tufan

    2011-01-01

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

  17. 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…

  18. 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 subj

  19. Audio/Visual Aids: A Study of the Effect of Audio/Visual Aids on the Comprehension Recall of Students.

    Science.gov (United States)

    Bavaro, Sandra

    A study investigated whether the use of audio/visual aids had an effect upon comprehension recall. Thirty fourth-grade students from an urban public school were randomly divided into two equal samples of 15. One group was given a story to read (print only), while the other group viewed a filmstrip of the same story, thereby utilizing audio/visual…

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

    International Nuclear Information System (INIS)

    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

  1. On Building Immersive Audio Applications Using Robust Adaptive Beamforming and Joint Audio-Video Source Localization

    Directory of Open Access Journals (Sweden)

    Beracoechea JA

    2006-01-01

    Full Text Available This paper deals with some of the different problems, strategies, and solutions of building true immersive audio systems oriented to future communication applications. The aim is to build a system where the acoustic field of a chamber is recorded using a microphone array and then is reconstructed or rendered again, in a different chamber using loudspeaker array-based techniques. Our proposal explores the possibility of using recent robust adaptive beamforming techniques for effectively estimating the original sources of the emitting room. A joint audio-video localization method needed in the estimation process as well as in the rendering engine is also presented. The estimated source signal and the source localization information drive a wave field synthesis engine that renders the acoustic field again at the receiving chamber. The system performance is tested using MUSHRA-based subjective tests.

  2. Audio visual information materials for risk communication

    International Nuclear Information System (INIS)

    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. Survey Musik und Medien 2012: Audio Media Usage in Germany - Audio Sources - Radio Traditionalists

    OpenAIRE

    Lepa, Steffen

    2013-01-01

    Where did everyday music come from in 2012? Audio Sources describe those distribution channels by means of which music is purchased, archived and made accessible. This includes physical recordings (CD, LP, MC etc.), electronic services in terms of downloading and streaming of digital music (iTunes, last.fm, Spotify etc.) as well as traditional radio reception and last but not least musical content on websites or digital storage media. Radio Traditionalists are represented in various age g...

  4. 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.

  5. An inconclusive digital audio authenticity examination: a unique case.

    Science.gov (United States)

    Koenig, Bruce E; Lacey, Douglas S

    2012-01-01

    This case report sets forth an authenticity examination of 35 encrypted, proprietary-format digital audio files containing recorded telephone conversations between two codefendants in a criminal matter. The codefendant who recorded the conversations did so on a recording system he developed; additionally, he was both a forensic audio authenticity examiner, who had published and presented in the field, and was the head of a professional audio society's writing group for authenticity standards. The authors conducted the examination of the recordings following nine laboratory steps of the peer-reviewed and published 11-step digital audio authenticity protocol. Based considerably on the codefendant's direct involvement with the development of the encrypted audio format, his experience in the field of forensic audio authenticity analysis, and the ease with which the audio files could be accessed, converted, edited in the gap areas, and reconstructed in such a way that the processes were undetected, the authors concluded that the recordings could not be scientifically authenticated through accepted forensic practices. PMID:21854384

  6. Beyond podcasting: creative approaches to designing educational audio

    Directory of Open Access Journals (Sweden)

    Andrew Middleton

    2009-12-01

    Full Text Available This paper discusses a university-wide pilot designed to encourage academics to creatively explore learner-centred applications for digital audio. Participation in the pilot was diverse in terms of technical competence, confidence and contextual requirements and there was little prior experience of working with digital audio. Many innovative approaches were taken to using audio in a blended context including student-generated vox pops, audio feedback models, audio conversations and task-setting. A podcast was central to the pilot itself, providing a common space for the 25 participants, who were also supported by materials in several other formats. An analysis of podcast interviews involving pilot participants provided the data informing this case study. This paper concludes that audio has the potential to promote academic creativity in engaging students through media intervention. However, institutional scalability is dependent upon the availability of suitable timely support mechanisms that can address the lack of technical confidence evident in many staff. If that is in place, audio can be widely adopted by anyone seeking to add a new layer of presence and connectivity through the use of voice.

  7. Real-Time Audio Translation Module Between Iax And Rsw

    Directory of Open Access Journals (Sweden)

    Hadeel Saleh Haj Aliwi

    2014-06-01

    Full Text Available At the last few years, multimedia communication has been developed and improved rapidly in order to enable users to communicate between each other over the internet. Generally, multimedia communication consists of audio and video communication. However, this research concentrates on audio conferencing only. The audio translation between protocols is a very critical issue, because it solves the communication problems between any two protocols. So, it enables people around the world to talk with each other even they use different protocols. In this research, a real time audio translation module between two protocols has been done. These two protocols are: InterAsteri sk eXchange Protocol (IAX and Real-Time Switching Control Protocol (RSW, which they are widely used to provide two ways audio transfer feature. The solution here is to provide interworking between the two protocols which they have different media transports, audio codec’s, header formats and different transport protocols for the audio transmission. This translation will help bridging the gap between the two protocols by providing interworking capability between the two audio streams of IAX and RSW. Some related works have been done to provide translation between IAX and RSW control signalling messages. But, this research paper concentrates on the translation that depends on the media transfer. The proposed translation module was tested and evaluated in different scenarios in order to examine its performance. The obtained results showed that the Real- Time Audio Translation Module produces lower rates of packet delay and jitter than the acceptance values for each of the mentioned performance metrics.

  8. Applications of Wavelets in 3-D Audio Simulation

    Institute of Scientific and Technical Information of China (English)

    2000-01-01

    Wavelet has been used as a powerful tool in the signal processing and function approx imation recently. This paper presents the application of wavelets for solving two key problems in 3-1 audio simulation. First, we employ discrete wavelet transform (DWT) combined with vector quantisation (VQ) to compress audio data in order to reduce tremendous redundant data storage and transmission times. Secondly, we use wavelets as the activation functions in neural networks called feed-forward wavelet networks to approach auditory localisation information cues (head-related transfer functions (HRTFs) are used here). The experimental results demonstrate that the applica tion of wavelets is more efficientand useful in 3-D audio simulation.

  9. Audio system using binaural synthesis for multimodal telepresence applications

    DEFF Research Database (Denmark)

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

    2013-01-01

    microphones, headphones and loudspeakers as well as measurements of network latency and bandwidth requirements of the system. Furthermore, measurements were made to determine whether the level of echo and cross talk cause any issues. The overall system employs multiple modalities to virtually transport a......An audio system was developed as part of a multimodal system aiming to go beyond current state of the art in telepresence.This paper provides an overview of how the audio was implemented and documents measurements that were performed on the audio system. The measurements include equalization of...

  10. 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....

  11. 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....

  12. Audio Features Underlying Perceived Groove and Sensorimotor Synchronization in Music

    OpenAIRE

    Stupacher, Jan; Hove, Michael J.

    2015-01-01

    The experience of groove is associated with the urge to move to a musical rhythm. Here we focus on the relevance of audio features, obtained using music information retrieval (MIR) tools, for explaining the perception of groove and music-related movement. In the first of three studies, we extracted audio features from clips of real music previously rated on perceived groove. Measures of variability, such as the variance of the audio signal’s RMS curve and spectral flux (particularly in low fr...

  13. Technical Evaluation Report 31: Internet Audio Products (3/ 3)

    OpenAIRE

    Jim Rudolph; Patricia Fahrni; Linda Schwartz; Adrienne de Schutter

    2004-01-01

    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...

  14. Audio Codecs in VoIPv6: A Performance Analysis

    OpenAIRE

    Monjur Ahmed; Mohammad Sarwar Hossain Mollah

    2014-01-01

    Audio communications in IP based networks have been revolutionized by the introduction of VoIP applications. High cost-efficiency has made VoIP to be the communication means in today’s world; and this trend is anticipated to be continued on an ongoing basis. The performance of VoIP significantly depends on the efficiency of the audio codecs used in any communication scenario which make the study on the performance issues of audio codecs in VoIP applications worth investigating. IPv6 is the ne...

  15. 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 %....

  16. Deep Multimodal Learning for Audio-Visual Speech Recognition

    OpenAIRE

    Mroueh, Youssef; Marcheret, Etienne; Goel, Vaibhava

    2015-01-01

    In this paper, we present methods in deep multimodal learning for fusing speech and visual modalities for Audio-Visual Automatic Speech Recognition (AV-ASR). First, we study an approach where uni-modal deep networks are trained separately and their final hidden layers fused to obtain a joint feature space in which another deep network is built. While the audio network alone achieves a phone error rate (PER) of $41\\%$ under clean condition on the IBM large vocabulary audio-visual studio datase...

  17. 37 CFR 201.27 - Initial notice of distribution of digital audio recording devices or media.

    Science.gov (United States)

    2010-07-01

    ... distribution of digital audio recording devices or media. 201.27 Section 201.27 Patents, Trademarks, and... Initial notice of distribution of digital audio recording devices or media. (a) General. This section..., any digital audio recording device or digital audio recording medium in the United States....

  18. TNO at TRECVID 2008, Combining Audio and Video Fingerprinting for Robust Copy Detection

    NARCIS (Netherlands)

    Doets, P.J.; Eendebak, P.T.; Ranguelova, E.; Kraaij, W.

    2009-01-01

    TNO has evaluated a baseline audio and a video fingerprinting system based on robust hashing for the TRECVID 2008 copy detection task. We participated in the audio, the video and the combined audio-video copy detection task. The audio fingerprinting implementation clearly outperformed the video fing

  19. Audio CAPTCHA for SIP-Based VoIP

    Science.gov (United States)

    Soupionis, Yannis; Tountas, George; Gritzalis, Dimitris

    Voice over IP (VoIP) introduces new ways of communication, while utilizing existing data networks to provide inexpensive voice communications worldwide as a promising alternative to the traditional PSTN telephony. SPam over Internet Telephony (SPIT) is one potential source of future annoyance in VoIP. A common way to launch a SPIT attack is the use of an automated procedure (bot), which generates calls and produces audio advertisements. In this paper, our goal is to design appropriate CAPTCHA to fight such bots. We focus on and develop audio CAPTCHA, as the audio format is more suitable for VoIP environments and we implement it in a SIP-based VoIP environment. Furthermore, we suggest and evaluate the specific attributes that audio CAPTCHA should incorporate in order to be effective, and test it against an open source bot implementation.

  20. 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)

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

    DEFF Research Database (Denmark)

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

    communication is challenged by the fact that patients tend to forget or misunderstand a great deal of the information given. The primary objective of this study is to investigate the effects of providing patients with an audio recording of the consultation. Methods A randomized controlled trial involving four...... 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...... conducted from February 2011 to August 2011. Discussion/Implication for field The study is expected to contribute with knowledge about the impact of audio recording in general and among subgroups. This will clarify whether specific groups of patients benefit in particular from the intervention....

  2. 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...

  3. Can audio recording improve patients' recall of outpatient consultations?

    DEFF Research Database (Denmark)

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

    objective we want to investigate if replay of the consultations improves the patients’ recall of the information given. Methods Interviews are carried out with 40 patients whose consultations have been audio recorded. Patients are divided into two groups, those who have listened to their consultation 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) with the...... information provided in the consultation (audio recording). Results Preliminary findings from 5 patients indicate that patients remember 50-70% of the information given. Whether this difference in recall is correlated with patients listening to the audio recording of their consultation or due to other...

  4. Audio-visual voice activity detection

    Institute of Scientific and Technical Information of China (English)

    LIU Peng; WANG Zuo-ying

    2006-01-01

    In speech signal processing systems,frame-energy based voice activity detection (VAD) method may be interfered with the background noise and non-stationary characteristic of the frame-energy in voice segment.The purpose of this paper is to improve the performance and robustness of VAD by introducing visual information.Meanwhile,data-driven linear transformation is adopted in visual feature extraction,and a general statistical VAD model is designed.Using the general model and a two-stage fusion strategy presented in this paper,a concrete multimodal VAD system is built.Experiments show that a 55.0% relative reduction in frame error rate and a 98.5% relative reduction in sentence-breaking error rate are obtained when using multimodal VAD,compared to frame-energy based audio VAD.The results show that using multimodal method,sentence-breaking errors are almost avoided,and flame-detection performance is clearly improved, which proves the effectiveness of the visual modal in VAD.

  5. CAVA (human Communication: an Audio-Visual Archive)

    OpenAIRE

    Mahon, M. S.

    2009-01-01

    In order to investigate human communication and interaction, researchers need hours of audio-visual data, sometimes recorded over periods of months or years. The process of collecting, cataloguing and transcribing such valuable data is time-consuming and expensive. Once it is collected and ready to use, it makes sense to get the maximum value from it by reusing it and sharing it among the research community. But unlike highly-controlled experimental data, natural audio-visual data tends t...

  6. Distributed audio recording using OFDR with double interrogation

    Science.gov (United States)

    Gabai, Haniel; Eyal, Avishay

    2014-05-01

    We introduce a phase sensitive, dynamic and long range fiber-optic sensing system with fully distributed audio recording capabilities. The proposed system implements a recently developed OFDR design, which is based on double interrogation of a sensing fiber with equally-spaced discrete reflectors. In this paper, the ability of each sensing segment to operate as an independent, purely optical audio recorder with little cross-talk artifacts is demonstrated.

  7. Acoustic Neurinoma With Bilateral Audio Logical Complication; a Case Report

    OpenAIRE

    Saeed Farahani

    1998-01-01

    Many of the CP angle tumors are acoustic neuroma, vestibular schowanoma or 8th nerve tumor. This kind of tumor is benign histologically. Big size ones can cause neurological symptoms such as cerebellar imbalance, edema and cranial nerves dysfunction. Acoustic neuroma is mostly unilateral and audio logical findings manifest a unilateral hearing loss. Although big size tumors can lead to bilateral audio logical symptoms which can affect the findings of hearing assessment. Here, a 31 year-old pa...

  8. Handreiking multimediaformaten: naar optimale toegang van audio, video en afbeeldingen

    OpenAIRE

    Folmer, E.J.A.; Wams, N.; Knubben, B.

    2010-01-01

    Multimedia maken meer en meer deel uit van de manier waarop we ons dagelijks uitdrukken; audio en video maken inmiddels het overgrote deel uit van het internetverkeer. Daarbij maken we gebruik van allerhande formaten, soms zonder daar goed bij stil te staan. Deze handreiking geeft achtergrond bij de de keuzes die u kunt maken om video en audio beschikbaar te stellen. Open Standaarden zijn daarbij (nog) minder gangbaar dan gesloten standaarden, maar zijn wel in opkomst en dragen bovendien bete...

  9. Underdetermined Blind Audio Source Separation Using Modal Decomposition

    OpenAIRE

    Abdeldjalil Aïssa-El-Bey; Karim Abed-Meraim; Yves Grenier

    2007-01-01

    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 signa...

  10. 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...

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

    DEFF Research Database (Denmark)

    Sturm, Bob L.; Daudet, Laurent

    2011-01-01

    We investigate recursive nearest neighbor search in a sparse domain at the scale of audio signals. Essentially, to approximate the cosine distance between the signals we make pairwise comparisons between the elements of localized sparse models built from large and redundant multiscale dictionaries...... of time-frequency atoms. Theoretically, error bounds on these approximations provide efficient means for quickly reducing the search space to the nearest neighborhood of a given data; but we demonstrate here that the best bound defined thus far involving a probabilistic assumption does not provide a...

  12. A Havoc Proof for Secure and Robust Audio Watermarking

    Directory of Open Access Journals (Sweden)

    K.Vaitheki

    2010-01-01

    Full Text Available The audio watermarking involves the concealment of data within a discrete audio file. Audio watermarking technology affords an opportunity to generate copies of a recording which are perceived by listeners as identical to the original but which may differ from one another on the basis of the embedded information. A highly confidential audio watermarking scheme using multiple scrambling is presented Superior to other audio watermarking techniques, the proposed scheme is self-secured by integrating multiple scrambling operations into the embedding stage. To ensure that unauthorized detection without correct secret keys is nearly impossible, the watermark is encrypted by a coded-image; certain frames are randomly selected from the total frames of the audio signal for embedding and their order of coding is further randomized. Adaptive synchronization is improves the robustness against hazardous synchronization attacks, such as random samples cropping/inserting and pitch-invariant time stretching. The efficient watermarking schemes make it impossible to be detected and robust even though the watermarking algorithm is open to the public.

  13. Experiences with audio feedback in a veterinary curriculum.

    Science.gov (United States)

    Rhind, Susan M; Pettigrew, Graham W; Spiller, Jo; Pearson, Geoff T

    2013-01-01

    On a national scale in the United Kingdom, student surveys have served to highlight areas within higher education that are not achieving high student satisfaction. Of particular concern to the veterinary and medical disciplines are the persistently poor levels of student satisfaction with academic feedback compared to students in other subjects. In this study we describe experiences with audio feedback trials in a veterinary curriculum. Students received audio feedback on either an in-course laboratory practical report or on an in-course multiple-choice test. Shortly after receiving their feedback, students were surveyed using an electronic questionnaire. In both courses, more students strongly agreed that audio feedback was helpful compared to either text-based (course A) or whole-class (course B) feedback. When asked to reflect on the helpfulness of various types of feedback they had received, audio feedback was rated less helpful than individual discussion with a member of staff (course A and course B), more helpful than peer discussion or automated feedback (course A and course B), and more helpful than written comments or whole-class review sessions (course B). From a faculty perspective, in course A, use of audio feedback was more efficient than handwritten feedback. In course B, the additional time commitment required was approximately 5 hours. Major themes in the qualitative data included the personal and individual nature of the feedback, quantity of feedback, improvement in students' insight into the process of marking, and the capacity of audio feedback to encourage and motivate. PMID:23470242

  14. Audio Codecs in VoIPv6: A Performance Analysis

    Directory of Open Access Journals (Sweden)

    Monjur Ahmed

    2014-04-01

    Full Text Available Audio communications in IP based networks have been revolutionized by the introduction of VoIP applications. High cost-efficiency has made VoIP to be the communication means in today’s world; and this trend is anticipated to be continued on an ongoing basis. The performance of VoIP significantly depends on the efficiency of the audio codecs used in any communication scenario which make the study on the performance issues of audio codecs in VoIP applications worth investigating. IPv6 is the new version of IP, which will gradually replace the current IPv4 as the transition from IPv4 to IPv6 is already in place. This demands the scrutiny of the audio codecs being used in IPv4 to be tested for their compatibility in IPv6 in terms of desired performance. This paper presents the study on the performance of selected audio codecs that are widely used in VoIPv4. G.711, G.729A and G.723.1 codecs were chosen for the study in VoIPv6 based scenarios presented in this paper. The selected audio codecs were applied in IPv6 based voice communication network scenarios to determine their performance efficiency by observing various QoS parameters. The study was done by means of simulation using OPNET.

  15. FABRICATION OF MESSAGE DIGEST TO AUTHENTICATE AUDIO SIGNALS WITH ALTERNATION OF COEFFICIENTS OF HARMONICS IN MULTI-STAGES (MDAC

    Directory of Open Access Journals (Sweden)

    Uttam Kr. Mondal

    2011-11-01

    Full Text Available Providing security to audio songs for maintaining its intellectual property right (IPR is one of chanllenging fields in commercial world especially in creative industry. In this paper, an effective approach has been incorporated to fabricate authentication of audio song through application of message digest method with alternation of coefficients of harmonics in multi-stages of higher frequency domain without affecting its audible quality. Decomposing constituent frequency components of song signal using Fourier transform with generating secret code via applying message digest followed by alternating coefficients of specific harmonics in multi-stages generates a secret code and this unique code is utilized to detect the originality of the song. A comparative study has been made with similar existing techniques and experimental results are also supported with mathematical formula based on Microsoft WAVE (".wav" stereo sound file.

  16. The state of audio descriptions in the United Kingdom – from description to narration

    OpenAIRE

    Steve Finbow

    2010-01-01

    Audio description and audio narration – this article focuses on the problematic relationship between description and narration, re-examining the pronouncements of objectivity in broadcast guidelines for audio describers in the United Kingdom. Using narratological theory and works by members of the various schools of continental philosophy by way of explication, the article calls for a more subjective and cultural-historical reference point for audio describers in the UK. Audio descriptions – ...

  17. Multichannel audio signal source separation based on an Interchannel Loudness Vector Sum

    OpenAIRE

    Park, Taejin; Lee, Taejin

    2015-01-01

    In this paper, a Blind Source Separation (BSS) algorithm for multichannel audio contents is proposed. Unlike common BSS algorithms targeting stereo audio contents or microphone array signals, our technique is targeted at multichannel audio such as 5.1 and 7.1ch audio. Since most multichannel audio object sources are panned using the Inter-channel Loudness Difference (ILD), we employ the ILVS (Inter-channel Loudness Vector Sum) concept to cluster common signals (such as background music) from ...

  18. An Open Dataset for Research on Audio Field Recording Archives: freefield1010

    OpenAIRE

    Stowell, D; Plumbley, MD

    2014-01-01

    We introduce a free and open dataset of 7690 audio clips sampled from the field-recording tag in the Freesound audio archive. The dataset is designed for use in research related to data mining in audio archives of field recordings / soundscapes. Audio is standardised, and audio and metadata are Creative Commons licensed. We describe the data preparation process, characterise the dataset descriptively, and illustrate its use through an auto-tagging experiment.

  19. Guidelines for the design of location-based audio for mobile learning

    OpenAIRE

    FitzGerald, Elizabeth; Sharples, Mike; Jones, Robert; Priestnall, Gary

    2010-01-01

    In this paper, we discuss the value of location-based and movement-sensitive audio for learning. We distinguish three types of audio learning experience: audio vignettes, movement-based guides and mobile narratives. An analysis of projects in these three areas has resulted in the formulation of guidelines for the design of audio experiences. We offer a case study of a novel audio experience, called "A Chaotic Encounter", that delivers an adaptive story based on the pattern of movements of the...

  20. Control of a velocity-sensitive audio-band quantum non-demolition interferometer

    CERN Document Server

    Leavey, S S; Gläfke, A; Barr, B W; Bell, A S; Gräf, C; Hennig, J -S; Houston, E A; Huttner, S H; Lück, H; Pascucci, D; Somiya, K; Sorazu, B; Spencer, A; Steinlechner, S; Strain, K A; Wright, J; Zhang, T; Hild, S

    2016-01-01

    The Sagnac speed meter interferometer topology can potentially provide enhanced sensitivity to gravitational waves in the audio-band compared to equivalent Michelson interferometers. A challenge with the Sagnac speed meter interferometer arises from the intrinsic lack of sensitivity at low frequencies where the velocity-proportional signal is smaller than the noise associated with the sensing of the signal. Using as an example the on-going proof-of-concept Sagnac speed meter experiment in Glasgow, we quantify the problem and present a solution involving the extraction of a small displacement-proportional signal. This displacement signal can be combined with the existing velocity signal to enhance low frequency sensitivity, and we derive optimal filters to accomplish this for different signal strengths. We show that the extraction of the displacement signal for low frequency control purposes can be performed without reducing significantly the quantum non-demolition character of this type of interferometer.

  1. Video equipment of tele dosimetry and audio

    International Nuclear Information System (INIS)

    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)

  2. Time and spectral analysis methods with machine learning for the authentication of digital audio recordings.

    Science.gov (United States)

    Korycki, Rafal

    2013-07-10

    This paper addresses the problem of tampering detection and discusses new methods that can be used for authenticity analysis of digital audio recordings. Nowadays, the only method referred to digital audio files commonly approved by forensic experts is the ENF criterion. It consists in fluctuation analysis of the mains frequency induced in electronic circuits of recording devices. Therefore, its effectiveness is strictly dependent on the presence of mains signal in the recording, which is a rare occurrence. This article presents the existing methods of time and spectral analysis along with their modifications as proposed by the author involving spectral analysis of residual signal enhanced by machine learning algorithms. The effectiveness of tampering detection methods described in this paper is tested on a predefined music database. The results are compared graphically using ROC-like curves. Furthermore, time-frequency plots are presented and enhanced by reassignment method in purpose of visual inspection of modified recordings. Using this solution, enables analysis of minimal changes of background sounds, which may indicate tampering. PMID:23481673

  3. A dual mode charge pump with adaptive output used in a class G audio power amplifier

    International Nuclear Information System (INIS)

    A dual mode charge pump to produce an adaptive power supply for a class G audio power amplifier is presented. According to the amplitude of the input signals, the charge pump has two level output voltage rails available to save power. It operates both in current mode at high output load and in pulse frequency modulation (PFM) at light load to reduce the power dissipation. Also, dynamic adjustment of the power stage transistor size based on load current at the PFM mode is introduced to reduce the output voltage ripple and prevent the switching frequency from audio range. The prototype is implemented in 0.18 μm 3.3 V CMOS technology. Experimental results show that the maximum power efficiency of the charge pump is 79.5% - 0.5x mode and 83.6% - 1x mode. The output voltage ripple is less than 15 mV while providing 120 mA of the load current at PFM control and less than 18 mV while providing 300 mA of the load current at current mode control. An analytical model for ripple voltage and efficiency calculation of the proposed PFM control demonstrates reasonable agreement with measured results. (semiconductor integrated circuits)

  4. A dual mode charge pump with adaptive output used in a class G audio power amplifier

    Science.gov (United States)

    Yong, Feng; Zhenfei, Peng; Shanshan, Yang; Zhiliang, Hong; Yang, Liu

    2011-04-01

    A dual mode charge pump to produce an adaptive power supply for a class G audio power amplifier is presented. According to the amplitude of the input signals, the charge pump has two level output voltage rails available to save power. It operates both in current mode at high output load and in pulse frequency modulation (PFM) at light load to reduce the power dissipation. Also, dynamic adjustment of the power stage transistor size based on load current at the PFM mode is introduced to reduce the output voltage ripple and prevent the switching frequency from audio range. The prototype is implemented in 0.18 μm 3.3 V CMOS technology. Experimental results show that the maximum power efficiency of the charge pump is 79.5% @ 0.5x mode and 83.6% @ 1x mode. The output voltage ripple is less than 15 mV while providing 120 mA of the load current at PFM control and less than 18 mV while providing 300 mA of the load current at current mode control. An analytical model for ripple voltage and efficiency calculation of the proposed PFM control demonstrates reasonable agreement with measured results.

  5. The Fungible Audio-Visual Mapping and its Experience

    Directory of Open Access Journals (Sweden)

    Adriana Sa

    2014-12-01

    Full Text Available This article questions how different sorts of audio-visual mappings may be perceived. 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; the present investigation seeks to glean its constitution and aspect. We 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, and posed quantitative and qualitative questions. These questions respect to 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. 

  6. Audio-video feature correlation: faces and speech

    Science.gov (United States)

    Durand, Gwenael; Montacie, Claude; Caraty, Marie-Jose; Faudemay, Pascal

    1999-08-01

    This paper presents a study of the correlation of features automatically extracted from the audio stream and the video stream of audiovisual documents. In particular, we were interested in finding out whether speech analysis tools could be combined with face detection methods, and to what extend they should be combined. A generic audio signal partitioning algorithm as first used to detect Silence/Noise/Music/Speech segments in a full length movie. A generic object detection method was applied to the keyframes extracted from the movie in order to detect the presence or absence of faces. The correlation between the presence of a face in the keyframes and of the corresponding voice in the audio stream was studied. A third stream, which is the script of the movie, is warped on the speech channel in order to automatically label faces appearing in the keyframes with the name of the corresponding character. We naturally found that extracted audio and video features were related in many cases, and that significant benefits can be obtained from the joint use of audio and video analysis methods.

  7. Content-based audio search: from fingerprinting to semantic audio retrieval

    OpenAIRE

    Cano Vila, Pedro

    2007-01-01

    Aquesta tesi tracta de cercadors d'audio basats en contingut. Específicament, tracta de desenvolupar tecnologies que permetin fer més estret l'interval semàntic o --semantic gap' que, a avui dia, limita l'ús massiu de motors de cerca basats en contingut. Els motors de cerca d'àudio fan servir metadades, en la gran majoria generada per editors, per a gestionar col.leccions d'àudio. Tot i ser una tasca àrdua i procliu a errors, l'anotació manual és la pràctica més habitual. Els mètodes basats e...

  8. Tagging and Linking Lecture Audio Recordings: Goals and Practice

    CERN Document Server

    Gray, Norman; Honeychurch, Sarah; Draper, Steve; Barr, Niall

    2013-01-01

    Making and distributing audio recordings of lectures is cheap and technically straightforward, and these recordings represent an underexploited teaching resource. We explore the reasons why such recordings are not more used; we believe the barriers inhibiting such use should be easily overcome. Students can listen to a lecture they missed, or re-listen to a lecture at revision time, but their interaction is limited by the affordances of the replaying technology. Listening to lecture audio is generally solitary, linear, and disjoint from other available media. In this paper, we describe a tool we are developing at the University of Glasgow, which enriches students' interactions with lecture audio. We describe our experiments with this tool in session 2012--13. Fewer students used the tool than we expected would naturally do so, and we discuss some possible explanations for this.

  9. Symmetric Key based Audio Steganography for Mobile Network

    Directory of Open Access Journals (Sweden)

    Anwesha Mukherjee

    2012-06-01

    Full Text Available This paper presents an innovative Symmetric Key based Audio Steganography algorithms for providing security in mobile network. In this paper the data to be transmitted is encrypted using proposed symmetric key cryptography algorithm. The encrypted text is hidden within an audio file during transmission. Fibonacci series based bit replacement technique is used for the purpose of hiding the data. The audio and the data file are converted into stream of bits. Encryption is performed to convert the plain text to cipher text using the proposed symmetric algorithm using 128-bit key. The bits of the cipher text to be replaced are selected based on Fibonacci series. The time complexity of the proposed algorithm is 75�0less than the previous approach

  10. 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.

  11. 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.

  12. 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 switch......-mode audio power amplifiers while keeping the performance measures to excellent levels is therefore of high general interest. A modulator utilizing multiple carrier signals to generate a two level pulse train will be shown in this paper. The performance of the modulator will be compared in simulation to...... existing modulation topologies. The lower EMI as well as the preserved audio performance will be shown in simulation as well as in measurement results on a prototype....

  13. Highlight summarization in golf videos using audio signals

    Science.gov (United States)

    Kim, Hyoung-Gook; Kim, Jin Young

    2008-01-01

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

  14. 声定向聚能系统性能测试研究%Research of the capability testing of directional and energy-concentrated audio system

    Institute of Scientific and Technical Information of China (English)

    马春庭; 谭业双; 李荣祥; 杜杰

    2012-01-01

    Directional and energy-concentrated audio system is the key undeathful weapon the the antiterrorism. The frequency characteristics to the reproduction audio signal, system directivity and the far field sound pressure level of system are the important guidelines. The capability testing of the directional and energy-concentrated audio system is satisfied to the ideal requirement through the testing. The error is also analyzed.%声定向聚能系统是未来重要的反恐防暴非致命武器之一,系统的再现声频信号的频率特性、系统的指向性、系统远场声压级是重要指标,通过测试声定向聚能系统性能基本达到理想要求,并分析了误差原因.

  15. Cover signal specific steganalysis: the impact of training on the example of two selected audio steganalysis approaches

    Science.gov (United States)

    Kraetzer, Christian; Dittmann, Jana

    2008-02-01

    The main goals of this paper are to show the impact of the basic assumptions for the cover channel characteristics as well as the impact of different training/testing set generation strategies on the statistical detectability of exemplary chosen audio hiding approaches known from steganography and watermarking. Here we have selected exemplary five steganography algorithms and four watermarking algorithms. The channel characteristics for two different chosen audio cover channels (an application specific exemplary scenario of VoIP steganography and universal audio steganography) are formalised and their impact on decisions in the steganalysis process, especially on the strategies applied for training/ testing set generation, are shown. Following the assumptions on the cover channel characteristics either cover dependent or cover independent training and testing can be performed, using either correlated or non-correlated training and test sets. In comparison to previous work, additional frequency domain features are introduced for steganalysis and the performance (in terms of classification accuracy) of Bayesian classifiers and multinomial logistic regression models is compared with the results of SVM classification. We show that the newly implemented frequency domain features increase the classification accuracy achieved in SVM classification. Furthermore it is shown on the example of VoIP steganalysis that channel character specific evaluation performs better than tests without focus on a specific channel (i.e. universal steganalysis). A comparison of test results for cover dependent and independent training and testing shows that the latter performs better for all nine algorithms evaluated here and the used SVM based classifier.

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

    Directory of Open Access Journals (Sweden)

    Zhang Qiang

    2015-01-01

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

  17. Entorno de Audio usando la nueva API de HTML 5

    OpenAIRE

    LATORRE PLAYÁN, JAVIER

    2015-01-01

    Este trabajo tiene como objetivo el diseño y programación de una aplicación de audio sobre la nueva API de audio de HTML 5. Para ello, utilizamos el programa SoundCool, que es propiedad de la Universidad Politécnica de Valencia y, a partir de los módulos que implementa, los adaptaremos al lenguaje antes mencionado, con el propósito de hacerlo más accesible y atractivo visualmente. Para poder llevar a cabo lo mencionado anteriormente, se ha realizado, en primer lugar, un trabajo de investig...

  18. 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

  19. An audio file tagging mobile game, mTagATune

    OpenAIRE

    Díaz, Francisco Javier; Queiruga, Claudia Alejandra; Ferraresso, Alejandro; Larghi, José

    2011-01-01

    mTagATune is a mobile game based on TagATune. mTagATune implements the concept of GWAP and seizes the capabilities and wide acceptance of current smartphones. GWAP promotes the creation of computer games that encourage people to do voluntary work. mTagATune implements a game that collects information on audio files to facilitate future searches on them. By means of a collaborative game, mTagATune enables an ubiquitous collection of information on audio files that can later be used in searc...

  20. 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. In...... this paper, we propose to use the desired audio signal instead. Specifically, we treat the case of estimating the distance between two loudspeakers playing back a stereo music or speech signal. In this connection, we develop a real-time maximum likelihood estimator and demonstrate that it has a...

  1. Audio Steganography Coding Using the Discrete Wavelet Transforms

    Directory of Open Access Journals (Sweden)

    Siwar Rekik

    2012-02-01

    Full Text Available The performance of audio steganography compression system using discrete wavelet transform(DWT is investigated. Audio steganography coding is the technology of transforming stegospeechinto efficiently encoded version that can be decoded in the receiver side to produce aclose representation of the initial signal (non compressed. Experimental results prove theefficiency of the used compression technique since the compressed stego-speech areperceptually intelligible and indistinguishable from the equivalent initial signal, while being able torecover the initial stego-speech with slight degradation in the quality .

  2. Video-assisted segmentation of speech and audio track

    Science.gov (United States)

    Pandit, Medha; Yusoff, Yusseri; Kittler, Josef; Christmas, William J.; Chilton, E. H. S.

    1999-08-01

    Video database research is commonly concerned with the storage and retrieval of visual information invovling sequence segmentation, shot representation and video clip retrieval. In multimedia applications, video sequences are usually accompanied by a sound track. The sound track contains potential cues to aid shot segmentation such as different speakers, background music, singing and distinctive sounds. These different acoustic categories can be modeled to allow for an effective database retrieval. In this paper, we address the problem of automatic segmentation of audio track of multimedia material. This audio based segmentation can be combined with video scene shot detection in order to achieve partitioning of the multimedia material into semantically significant segments.

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

    Science.gov (United States)

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

    2015-11-01

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

  4. MedlinePlus FAQ: Is audio description available for videos on MedlinePlus?

    Science.gov (United States)

    ... https://medlineplus.gov/faq/audiodescription.html Question: Is audio description available for videos on MedlinePlus? To use ... features on this page, please enable JavaScript. Answer: Audio description of videos helps make the content of ...

  5. Audio Environment Recognition using Zero Crossing Features and MPEG-7 Descriptors

    OpenAIRE

    Saleh Al-Zhrani; Mubarak AlQahtani

    2010-01-01

    Problem statement: This study investigated zero crossing features and selected MPEG-7 audio descriptors for environment sound recognition applications such as audio forensics. Approach: The study implemented several experiments focusing on the problems of environment recognition from audio particularly for forensic applications. Results: It was investigated the effect of the temporal zero crossing feature as well as selected MPEG-7 audio low level descriptors on environment sound recognition....

  6. Evaluation of MPEG-7-Based Audio Descriptors for Animal Voice Recognition over Wireless Acoustic Sensor Networks

    OpenAIRE

    Joaquín Luque; Diego F. Larios; Enrique Personal; Julio Barbancho; Carlos León

    2016-01-01

    Environmental audio monitoring is a huge area of interest for biologists all over the world. This is why some audio monitoring system have been proposed in the literature, which can be classified into two different approaches: acquirement and compression of all audio patterns in order to send them as raw data to a main server; or specific recognition systems based on audio patterns. The first approach presents the drawback of a high amount of information to be stored in a main server. Moreove...

  7. Using Touch Screen Audio-CASI to Obtain Data on Sensitive Topics

    OpenAIRE

    Cooley, Philip C.; Rogers, Susan M; Turner, Charles F.; Al-Tayyib, Alia A.; Willis, Gordon; Ganapathi, Laxminarayana

    2001-01-01

    This paper describes a new interview data collection system that uses a laptop personal computer equipped with a touch-sensitive video monitor. The touch-screen-based audio computer-assisted self-interviewing system, or touch screen audio-CASI, enhances the ease of use of conventional audio CASI systems while simultaneously providing the privacy of self-administered questionnaires. We describe touch screen audio-CASI design features and operational characteristics. In addition, we present dat...

  8. Developing a Framework for Effective Audio Feedback: A Case Study

    Science.gov (United States)

    Hennessy, Claire; Forrester, Gillian

    2014-01-01

    The increase in the use of technology-enhanced learning in higher education has included a growing interest in new approaches to enhance the quality of feedback given to students. Audio feedback is one method that has become more popular, yet evaluating its role in feedback delivery is still an emerging area for research. This paper is based on a…

  9. 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...

  10. 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…

  11. 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.

  12. 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...

  13. Audio Card Systems. Technical Information Bulletin No. 13.

    Science.gov (United States)

    Gasser, P.

    This examination of audio card systems for computers begins by identifying the three information processing systems for sound: sound digitizing, synthesis of text, and word recognition. Specific pedagogical applications of digitized sound are then briefly discussed. The remainder of the document focuses on specifications for the working of vocal…

  14. Mediatheque - digitization and preservation of audio content in RTV Slovenia

    Directory of Open Access Journals (Sweden)

    Martin Žvelc

    2011-01-01

    Full Text Available RTV Slovenia’s archives contain large amounts of audio and video materials, various documents and music scores, and most of them are still in the analogue format. Widespread digitization has revolutionized the processes and ways of creating content in the digital format, recorded on different media. Such records also require new ways of preservation. In the article the development and structure of the Mediateque department at RTV Slovenia is presented. Also an overview to the preservation model of audio content is given. Due to rapid technological changes the audio content was the most critical and the first to be digitized. The intensive work in Mediatheque began in 2008 and after two years Radio Slovenia has developed modern system of permanent storage of audio content. Radio Slovenia’s Digital Archive meets all the standards and regulations applicable to modern archival systems. In the article the application of Mediarc software is also presented, which as it could be used for digitizing and permanent storage of TV Slovenia’s video archives.

  15. Audio-visual perception system for a humanoid robotic head.

    Science.gov (United States)

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

    2014-01-01

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

  16. Comparative study of Audio-lingual method and CLT

    Institute of Scientific and Technical Information of China (English)

    2013-01-01

    For language teaching,various teaching methods and approaches have been proposed. But no one teaching approach is one-for-al that is good enough to be used as the standard of teaching. Among so many methods this paper mainly concerns the audio-lingual method and CLT.

  17. 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...

  18. Audio-haptic interaction in simulated walking experiences

    DEFF Research Database (Denmark)

    Serafin, Stefania

    2011-01-01

    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 interc...

  19. 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 stim...

  20. Effectiveness of Audio on Screen Captures in Software Application Instruction

    Science.gov (United States)

    Veronikas, Susan Walsh; Maushak, Nancy

    2005-01-01

    Presentation of software instruction has been supported by manuals and textbooks consisting of screen captures, but a multimedia approach may increase learning outcomes. This study investigated the effects of modality (text, audio, or dual) on the achievement and attitudes of college students learning a software application through the computer.…

  1. Utilization of non-linear converters for audio amplification

    DEFF Research Database (Denmark)

    Iversen, Niels Elkjær; Birch, Thomas; Knott, Arnold

    2012-01-01

    . The introduction of non-linear converters for audio amplification defeats this limitation. A Cuk converter, designed to deliver an AC peak output voltage twice the supply voltage, is presented in this paper. A 3V prototype has been developed to prove the concept. The prototype shows that it is...

  2. Audio-Described Educational Materials: Ugandan Teachers' Experiences

    Science.gov (United States)

    Wormnaes, Siri; Sellaeg, Nina

    2013-01-01

    This article describes and discusses a qualitative, descriptive, and exploratory study of how 12 visually impaired teachers in Uganda experienced audio-described educational video material for teachers and student teachers. The study is based upon interviews with these teachers and observations while they were using the material either…

  3. Integrated Spacesuit Audio System Enhances Speech Quality and Reduces Noise

    Science.gov (United States)

    Huang, Yiteng Arden; Chen, Jingdong; Chen, Shaoyan Sharyl

    2009-01-01

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

  4. 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...

  5. Audio and Video Reflections to Promote Social Justice

    Science.gov (United States)

    Boske, Christa

    2011-01-01

    Purpose: The purpose of this paper is to examine how 15 graduate students enrolled in a US school leadership preparation program understand issues of social justice and equity through a reflective process utilizing audio and/or video software. Design/methodology/approach: The study is based on the tradition of grounded theory. The researcher…

  6. 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...

  7. Real-Time Audio-Visual Analysis for Multiperson Videoconferencing

    Directory of Open Access Journals (Sweden)

    Petr Motlicek

    2013-01-01

    Full Text Available We describe the design of a system consisting of several state-of-the-art real-time audio and video processing components enabling multimodal stream manipulation (e.g., automatic online editing for multiparty videoconferencing applications in open, unconstrained environments. The underlying algorithms are designed to allow multiple people to enter, interact, and leave the observable scene with no constraints. They comprise continuous localisation of audio objects and its application for spatial audio object coding, detection, and tracking of faces, estimation of head poses and visual focus of attention, detection and localisation of verbal and paralinguistic events, and the association and fusion of these different events. Combined all together, they represent multimodal streams with audio objects and semantic video objects and provide semantic information for stream manipulation systems (like a virtual director. Various experiments have been performed to evaluate the performance of the system. The obtained results demonstrate the effectiveness of the proposed design, the various algorithms, and the benefit of fusing different modalities in this scenario.

  8. 77 FR 16890 - Second Meeting: RTCA Special Committee 226, Audio Systems and Equipment

    Science.gov (United States)

    2012-03-22

    ... Federal Aviation Administration Second Meeting: RTCA Special Committee 226, Audio Systems and Equipment... meeting RTCA Special Committee 226, Audio Systems and Equipment. SUMMARY: The FAA is issuing this notice to advise the public of the second meeting of RTCA Special Committee 226, Audio Systems and...

  9. 76 FR 79755 - First Meeting: RTCA Special Committee 226 Audio Systems and Equipment

    Science.gov (United States)

    2011-12-22

    ... Federal Aviation Administration First Meeting: RTCA Special Committee 226 Audio Systems and Equipment... RTCA Special Committee 226, Audio Systems and Equipment. SUMMARY: The FAA is issuing this notice to advise the public of a meeting of RTCA Special Committee 226, Audio Systems and Equipment, for the...

  10. 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…

  11. Audio Use in E-Learning: What, Why, When, and How?

    Science.gov (United States)

    Calandra, Brendan; Barron, Ann E.; Thompson-Sellers, Ingrid

    2008-01-01

    Decisions related to the implementation of audio in e-learning are perplexing for many instructional designers, and deciphering theory and principles related to audio use can be difficult for practitioners. Yet, as bandwidth on the Internet increases, digital audio is becoming more common in online courses. This article provides a review of…

  12. 16 CFR 307.8 - Requirements for disclosure in audiovisual and audio advertising.

    Science.gov (United States)

    2010-01-01

    ... 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... and in graphics so that it is easily legible. If the advertisement has an audio component, the...

  13. 47 CFR 73.9005 - Compliance requirements for covered demodulator products: Audio.

    Science.gov (United States)

    2010-10-01

    ... products: Audio. 73.9005 Section 73.9005 Telecommunication FEDERAL COMMUNICATIONS COMMISSION (CONTINUED....9005 Compliance requirements for covered demodulator products: Audio. Except as otherwise provided in §§ 73.9003(a) or 73.9004(a), covered demodulator products shall not output the audio portions...

  14. 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...

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

    Science.gov (United States)

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

    2016-03-01

    Audio Visual Stimulation (AVS), a form of neurofeedback, is a non-pharmacological intervention that has been used for both performance enhancement and symptom management. We review the history of AVS, its two sub-types (close- and open-loop), and discuss its clinical implications. We also describe a promising new application of AVS to improve sleep, and potentially decrease pain. AVS research can be traced back to the late 1800s. AVS's efficacy has been demonstrated for both performance enhancement and symptom management. Although AVS is commonly used in clinical settings, there is limited literature evaluating clinical outcomes and mechanisms of action. One of the challenges to AVS research is the lack of standardized terms, which makes systematic review and literature consolidation difficult. Future studies using AVS as an intervention should; (1) use operational definitions that are consistent with the existing literature, such as AVS, Audio-visual Entrainment, or Light and Sound Stimulation, (2) provide a clear rationale for the chosen training frequency modality, (3) use a randomized controlled design, and (4) follow the Consolidated Standards of Reporting Trials and/or related guidelines when disseminating results. PMID:26294268

  16. Robust Watermarking Using Compressed Sensing Framework with Application to MP3 Audio

    Directory of Open Access Journals (Sweden)

    Mohamed Waleed Fakhr

    2012-12-01

    Full Text Available In this paper a watermark embedding and recovery technique is proposed based on the compressed sensing framework. Both the watermark and the host signal are sparse, each in its own domain. In recovery, the L1-minimization is used to recover the watermark and the host signal almost perfectly in clean conditions. The proposed technique is tested on MP3 audio compression-decompression attack and additive noise attack. Bit error rates are compared with standard spread spectrum embedding. The proposed technique is implemented for both time domain and frequency domain embedding with a unified approach. The WalshHadamard transform (WHT, the discrete cosine transform (DCT and the Karhunen-Loeve transform (KLT are compared in the host signal sparsifying process. Significant performance improvements in all tested conditions are achieved against the spread spectrum embedding. A payload as high as 172bps in additive noise attacks, 86bps in 128kbps MP3 attacks and 11bps in 64kbps MP3 attacks are achieved at small bit error rates and acceptable MP3 audio signal quality.

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

    Directory of Open Access Journals (Sweden)

    Pakarinen Jyri

    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.

  18. Robust Watermarking Using Compressed Sensing Framework with Application to MP3 Audio

    Directory of Open Access Journals (Sweden)

    Mohamed Waleed Fakhr

    2013-01-01

    Full Text Available In this paper a watermark embedding and recovery technique is proposed based on the compressed sensingframework. Both the watermark and the host signal are sparse, each in its own domain. In recovery, theL1-minimization is used to recover the watermark and the host signal almost perfectly in clean conditions.The proposed technique is tested on MP3 audio compression-decompression attack and additive noiseattack. Bit error rates are compared with standard spread spectrum embedding. The proposed technique isimplemented for both time domain and frequency domain embedding with a unified approach. The Walsh-Hadamard transform (WHT, the discrete cosine transform (DCT and the Karhunen-Loeve transform(KLT are compared in the host signal sparsifying process. Significant performance improvements in alltested conditions are achieved against the spread spectrum embedding. A payload as high as 172bps inadditive noise attacks, 86bps in 128kbps MP3 attacks and 11bps in 64kbps MP3 attacks are achieved atsmall bit error rates and acceptable MP3 audio signal quality.

  19. 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"--

  20. Modelling and extraction of fundamental frequency in speech signals

    OpenAIRE

    Pawi, Alipah

    2014-01-01

    This thesis was submitted for the degree of Doctor of Philosophy and awarded by Brunel University. One of the most important parameters of speech is the fundamental frequency of vibration of voiced sounds. The audio sensation of the fundamental frequency is known as the pitch. Depending on the tonal/non-tonal category of language, the fundamental frequency conveys intonation, pragmatics and meaning. In addition the fundamental frequency and intonation carry speaker gender, age, identity, s...

  1. Analysis of a digital technique for frequency transposition of speech

    OpenAIRE

    DiGirolamo, Vincent

    1985-01-01

    Frequency transposition is the process of raising or lowering the frequency content (pitch) of an audio signal. The hearing impaired community has the greatest interest in the application of frequency transposition. Though several analog and digital frequency transposing hearing aid systems have been built and tested, this investigates a possible digital processing alternative. Pole shifting, in the z-domain of an autoregressive (all pole) model of speech was proven to be a viable theory f...

  2. 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.

  3. A sub-milliwatt audio-processing platform for digital hearing aids

    International Nuclear Information System (INIS)

    We present a novel audio-processing platform, FlexEngine, which is composed of a 24-bit application-specific instruction-set processor (ASIP) and five dedicated accelerators. Acceleration instructions, compact instructions and repeat instruction are added into the ASIP's instruction set to deal with some core tasks of hearing aid algorithms. The five configurable accelerators are used to execute several of the most common functions of hearing aids. Moreover, several low power strategies, such as clock gating, data isolation, memory partition, bypass mode, sleep mode, are also applied in this platform for power reduction. The proposed platform is implemented in CMOS 130 nm technology, and test results show that power consumption of FlexEngine is 0.863 mW with the clock frequency of 8 MHz at Vdd = 1.0 V. (semiconductor integrated circuits)

  4. Audio Linguistic Disorders in Autistic Children

    International Nuclear Information System (INIS)

    Objective: To explore auditory function abnormalities and language disorder in autistic children. Twelve children with criteria of infantile autism were tested using Pure Tone Audiometry (PTA), Immitancemetry, Transient Evoked Otoacoustic Emission Test (TEOAE), Auditory Brainstem Response (ABR), Standardized Arabic Test of Early Language Development (for both receptive and expressive language). For comparison twlive normal children were chosen as control group. Statistically significant increase in hearing threshold level for the autistic children at low frequency region 250, 500 and 1000 Hz, significant reduction of the amplitude of TEOAE test and significant increase in wave I and V latency and I-V inter-peak latency at both RR 21.2 and 51.2 msec when compared to the control group. A positive correlation was found in this study between the changes in ABR latency and the severity of verbal disability. These resuts leed to the conclusion that Auditory dysfunction in autistic children can be verified through the presence of cochlear involvement and a delay in the brain stem transmission time in those patients. Disturbed verbal communication can be due to dysfunction in the auditory processing mechanisms

  5. “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.

  6. Acoustic Neurinoma With Bilateral Audio Logical Complication; a Case Report

    Directory of Open Access Journals (Sweden)

    Saeed Farahani

    1998-03-01

    Full Text Available Many of the CP angle tumors are acoustic neuroma, vestibular schowanoma or 8th nerve tumor. This kind of tumor is benign histologically. Big size ones can cause neurological symptoms such as cerebellar imbalance, edema and cranial nerves dysfunction. Acoustic neuroma is mostly unilateral and audio logical findings manifest a unilateral hearing loss. Although big size tumors can lead to bilateral audio logical symptoms which can affect the findings of hearing assessment. Here, a 31 year-old patient suffering right ear vestibular schowanoma have been reported. changes in left ear pure tone results, acoustic reflex measurements and ABR in addition to hearing loss in the right ear have been demonstrated.

  7. Adaptive audio watermarking based on SNR in localized regions

    Institute of Scientific and Technical Information of China (English)

    WU Guo-min; ZHUANG Yue-ting; WU Fei; PAN Yun-he

    2005-01-01

    In this paper, a novel localized audio watermarking scheme based on signal to noise ratio (SNR) to determine a scaling parameter α is proposed. The basic idea is to embed watermark in selected high inflexion regions, and the intensity of embedded watermarks are modified by adaptively adjusting α. As these high inflexion local regions usually correspond to music edges like sound of percussion instruments, explosion or transition of mixed music, which represent the music rhythm or tempo and are very important to human auditory perception, the embedded watermark is especially expected to escape the distortions caused by time domain synchronization attacks. Taking advantage of localization and SNR, the method shows strong robustness against common problems in audio signal processing, random cropping, time scale modification, etc.

  8. Random Numbers Generated from Audio and Video Sources

    Directory of Open Access Journals (Sweden)

    I-Te Chen

    2013-01-01

    Full Text Available Random numbers are very useful in simulation, chaos theory, game theory, information theory, pattern recognition, probability theory, quantum mechanics, statistics, and statistical mechanics. The random numbers are especially helpful in cryptography. In this work, the proposed random number generators come from white noise of audio and video (A/V sources which are extracted from high-resolution IPCAM, WEBCAM, and MPEG-1 video files. The proposed generator applied on video sources from IPCAM and WEBCAM with microphone would be the true random number generator and the pseudorandom number generator when applied on video sources from MPEG-1 video file. In addition, when applying NIST SP 800-22 Rev.1a 15 statistics tests on the random numbers generated from the proposed generator, around 98% random numbers can pass 15 statistical tests. Furthermore, the audio and video sources can be found easily; hence, the proposed generator is a qualified, convenient, and efficient random number generator.

  9. 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.

  10. 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.

  11. EMOTION ANALYSIS OF SONGS BASED ON LYRICAL AND AUDIO FEATURES

    Directory of Open Access Journals (Sweden)

    Adit Jamdar

    2015-05-01

    Full Text Available In this paper, a method is proposed to detect the emotion of a song based on its lyrical and audio features. Lyrical features are generated by segmentation of lyrics during the process of data extraction. ANEW and WordNet knowledge is then incorporated to compute Valence and Arousal values. In addition to this, linguistic association rules are applied to ensure that the issue of ambiguity is properly addressed. Audio features are used to supplement the lyrical ones and include attributes like energy, tempo, and danceability. These features are extracted from The Echo Nest, a widely used music intelligence platform. Construction of training and test sets is done on the basis of social tags extracted from the last.fm website. The classification is done by applying feature weighting and stepwise threshold reduction on the k-Nearest Neighbors algorithm to provide fuzziness in the classification.

  12. Comparing Audio Features and Playlist Statistics for Music Classification

    OpenAIRE

    Vatolkin, Igor; Bonnin, Geoffray; Jannach, Dietmar

    2014-01-01

    In recent years, a number of approaches have been developed for the automatic recognition of music genres, but also more specific categories (styles, moods, personal preferences, etc.). Among the different sources for building classification models, features extracted from the audio signal play an important role in the literature. Although such features can be extracted from any digitized music piece independently of the availability of other information sources, their extraction can require ...

  13. Prioritizing signals for selective real-time audio processing

    OpenAIRE

    Gallo, Emmanuel; Lemaitre, Guillaume; Tsingos, Nicolas

    2005-01-01

    This paper studies various priority metrics that can be used to progressively select sub-parts of a number of audio signals for realtime processing. In particular, five level-related metrics were examined: RMS level, A-weighted level, Zwicker and Moore loudness models and a masking threshold-based model. We conducted a pilot subjective evaluation study aimed at evaluating which metric would perform best at reconstructing mixtures of various types (speech, ambient and music) using only a budge...

  14. 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.

  15. Random Numbers Generated from Audio and Video Sources

    OpenAIRE

    I-Te Chen

    2013-01-01

    Random numbers are very useful in simulation, chaos theory, game theory, information theory, pattern recognition, probability theory, quantum mechanics, statistics, and statistical mechanics. The random numbers are especially helpful in cryptography. In this work, the proposed random number generators come from white noise of audio and video (A/V) sources which are extracted from high-resolution IPCAM, WEBCAM, and MPEG-1 video files. The proposed generator applied on video sources from IPCAM ...

  16. Audio Message Transmitter Secured Through Elliptical Curve Cryptosystem

    OpenAIRE

    Luma, Artan; Selimi, Besnik; Ameti, Lirim

    2014-01-01

    Securing a communication is always a challenge for participants in it. A lot of applications available in the market claim to enable secure audio communication, but not always show the details of the technology used behind to encrypt the data. It is important for end users to understand the techniques used for encrypting the data, in order to trust it. Elliptic curve cryptography, an approach to public key cryptography, is now widely used in cryptographic systems. Hence, in this paper we prop...

  17. Folksonomy-based tag recommendation for online audio clip sharing

    OpenAIRE

    Font, Frederic; Serr?? Juli??, Joan; Serra, Xavier

    2012-01-01

    Collaborative tagging has emerged as an efficient way to semantically describe online resources shared by a community of users. However, tag descriptions present some drawbacks such as tag scarcity or concept inconsistencies. In these situations, tag recommendation strategies can help users in adding meaningful tags to the resources being described. Freesound is an online audio clip sharing site that uses collaborative tagging to describe a collection of more than 140,...

  18. Navigation for the Blind through Audio-Based Virtual Environments

    OpenAIRE

    Sánchez, Jaime; Sáenz, Mauricio; Pascual-Leone, Alvaro; Merabet, Lotfi

    2010-01-01

    We present the design, development and an initial study changes and adaptations related to navigation that take place in the brain, by incorporating an Audio-Based Environments Simulator (AbES) within a neuroimaging environment. This virtual environment enables a blind user to navigate through a virtual representation of a real space in order to train his/her orientation and mobility skills. Our initial results suggest that this kind of virtual environment could be highly efficient as a testi...

  19. Phase Synchronization in Human EEG During Audio-Visual Stimulation

    Czech Academy of Sciences Publication Activity Database

    Teplan, M.; Šušmáková, K.; Paluš, Milan; Vejmelka, Martin

    2009-01-01

    Roč. 28, - (2009), s. 80-84. ISSN 1536-8378 Grant ostatní: Bilateral project between Slovak AS and AS CR(CZ-SK) Modern methods for evaluation of electrophysiological signals Source of funding: V - iné verejné zdroje Keywords : synchronization * EEG * wavelet * audio-visual stimulation Subject RIV: FH - Neurology Impact factor: 0.729, year: 2009

  20. Audio Processing Solution for Video Conference Based Aerobics

    OpenAIRE

    Berggren, Magnus; Stjernberg, Louise; Lindström, Fredric; Claesson, Ingvar

    2010-01-01

    In this paper an audio processing solution for video conference based aerobics is presented. The proposed solution leaves the workout music unaltered by separating it from the speech and processing each signal separately. The speech signal processing is also performed at a lower sample rate, which saves computational power. Real time evaluation of the system shows that high quality music as well as a good two-way communication is maintained during the aerobic session.

  1. NFL Films audio, video, and film production facilities

    Science.gov (United States)

    Berger, Russ; Schrag, Richard C.; Ridings, Jason J.

    2003-04-01

    The new NFL Films 200,000 sq. ft. headquarters is home for the critically acclaimed film production that preserves the NFL's visual legacy week-to-week during the football season, and is also the technical plant that processes and archives football footage from the earliest recorded media to the current network broadcasts. No other company in the country shoots more film than NFL Films, and the inclusion of cutting-edge video and audio formats demands that their technical spaces continually integrate the latest in the ever-changing world of technology. This facility houses a staggering array of acoustically sensitive spaces where music and sound are equal partners with the visual medium. Over 90,000 sq. ft. of sound critical technical space is comprised of an array of sound stages, music scoring stages, audio control rooms, music writing rooms, recording studios, mixing theaters, video production control rooms, editing suites, and a screening theater. Every production control space in the building is designed to monitor and produce multi channel surround sound audio. An overview of the architectural and acoustical design challenges encountered for each sophisticated listening, recording, viewing, editing, and sound critical environment will be discussed.

  2. Temporal structure and complexity affect audio-visual correspondence detection

    Directory of Open Access Journals (Sweden)

    Rachel N Denison

    2013-01-01

    Full Text Available Synchrony between events in different senses has long been considered the critical temporal cue for multisensory integration. Here, using rapid streams of auditory and visual events, we demonstrate how humans can use temporal structure (rather than mere temporal coincidence to detect multisensory relatedness. We find psychophysically that participants can detect matching auditory and visual streams via shared temporal structure for crossmodal lags of up to 200 ms. Performance on this task reproduced features of past findings based on explicit timing judgments but did not show any special advantage for perfectly synchronous streams. Importantly, the complexity of temporal patterns influences sensitivity to correspondence. Stochastic, irregular streams – with richer temporal pattern information – led to higher audio-visual matching sensitivity than predictable, rhythmic streams. Our results reveal that temporal structure and its complexity are key determinants for human detection of audio-visual correspondence. The distinctive emphasis of our new paradigms on temporal patterning could be useful for studying special populations with suspected abnormalities in audio-visual temporal perception and multisensory integration.

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

    Science.gov (United States)

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

    2002-05-01

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

  4. Omnidirectional Audio-Visual Talker Localization Based on Dynamic Fusion of Audio-Visual Features Using Validity and Reliability Criteria

    Science.gov (United States)

    Denda, Yuki; Nishiura, Takanobu; Yamashita, Yoichi

    This paper proposes a robust omnidirectional audio-visual (AV) talker localizer for AV applications. The proposed localizer consists of two innovations. One of them is robust omnidirectional audio and visual features. The direction of arrival (DOA) estimation using an equilateral triangular microphone array, and human position estimation using an omnidirectional video camera extract the AV features. The other is a dynamic fusion of the AV features. The validity criterion, called the audioor visual-localization counter, validates each audio- or visual-feature. The reliability criterion, called the speech arriving evaluator, acts as a dynamic weight to eliminate any prior statistical properties from its fusion procedure. The proposed localizer can compatibly achieve talker localization in a speech activity and user localization in a non-speech activity under the identical fusion rule. Talker localization experiments were conducted in an actual room to evaluate the effectiveness of the proposed localizer. The results confirmed that the talker localization performance of the proposed AV localizer using the validity and reliability criteria is superior to that of conventional localizers.

  5. Design and realization of digital audio equalizer based on MCU and FPAA

    Institute of Scientific and Technical Information of China (English)

    Zhou Ping; Liu Zhuo; Xia Liang

    2008-01-01

    In analog audio equalizer, the filters are constructed by op-amplifiers and discrete components. Being influenced by its discrete capabilities, audio equalizer has many disadvantages. Meanwhile, pure digital audio equalizer has got better performance and stability, but its cost and price are too high. So digital audio equalizer only has its application in upscale domain. A new design method for audio equalizer is proposed, which attempts to design and realize a high precision and high SNR (signal noise ratio) digital audio equalizer system based on field programmable analog array (FPAA) and micro-controller unit. This design confirms that design speed and performance will be greatly enhanced when FPAA technology is applied to analog design domain.

  6. MP3 audio-editing software for the department of radiology

    International Nuclear Information System (INIS)

    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)

  7. Improving the perceptual quality of single-channel blind audio source separation.

    OpenAIRE

    Stokes, Tobias W.

    2015-01-01

    Given a mixture of audio sources, a blind audio source separation (BASS) tool is required to extract audio relating to one specific source whilst attenuating that related to all others. This thesis answers the question “How can the perceptual quality of BASS be improved for broadcasting applications?” The most common source separation scenario, particularly in the field of broadcasting, is single channel, and this is particularly challenging as a limited set of cues are available. Broadcas...

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

    OpenAIRE

    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 capacit...

  9. Follow the Sound : Design of mobile spatial audio applications for pedestrian navigation

    OpenAIRE

    2012-01-01

    Auditory displays are slower than graphical user interfaces. We believe spatial audio can change that. Human perception can localize the position of sound sources due to psychoacoustical cues. Spatial audio reproduces these cues to produce virtual sound source position by headphones. The spatial attribute of sound can be used to produce richer and more effective auditory displays. In this work, there is proposed a set of interaction design guidelines for the use of spatial audio displays i...

  10. The Language of Filmic Audio Description: a Corpus-Based Analysis of Adjectives

    OpenAIRE

    Arma, Saveria

    2011-01-01

    Audio description is a recent technique that allows primarily blind and low-sighted people to gain adequate access to audiovisual products such as movies, theatre performances, sport matches, museums, archives, web resources and sculpture works. An additional audio track describes the visual elements that would otherwise remain inaccessible to the visually impaired, such as settings, costumes, atmospheres and written titles. Though very recent, audio description has rapidly expande...

  11. A Virtual Audio Guidance and Alert System for Commercial Aircraft Operations

    Science.gov (United States)

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

    1996-01-01

    Our work in virtual reality systems at NASA Ames Research Center includes the area of aurally-guided visual search, using specially-designed audio cues and spatial audio processing (also known as virtual or "3-D audio") techniques (Begault, 1994). Previous studies at Ames had revealed that use of 3-D audio for Traffic Collision Avoidance System (TCAS) advisories significantly reduced head-down time, compared to a head-down map display (0.5 sec advantage) or no display at all (2.2 sec advantage) (Begault, 1993, 1995; Begault & Pittman, 1994; see Wenzel, 1994, for an audio demo). Since the crew must keep their head up and looking out the window as much as possible when taxiing under low-visibility conditions, and the potential for "blunder" is increased under such conditions, it was sensible to evaluate the audio spatial cueing for a prototype audio ground collision avoidance warning (GCAW) system, and a 3-D audio guidance system. Results were favorable for GCAW, but not for the audio guidance system.

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

    Science.gov (United States)

    1974-01-01

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

  13. Realization of guitar audio effects using methods of digital signal processing

    Science.gov (United States)

    Buś, Szymon; Jedrzejewski, Konrad

    2015-09-01

    The paper is devoted to studies on possibilities of realization of guitar audio effects by means of methods of digital signal processing. As a result of research, some selected audio effects corresponding to the specifics of guitar sound were realized as the real-time system called Digital Guitar Multi-effect. Before implementation in the system, the selected effects were investigated using the dedicated application with a graphical user interface created in Matlab environment. In the second stage, the real-time system based on a microcontroller and an audio codec was designed and realized. The system is designed to perform audio effects on the output signal of an electric guitar.

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

    Directory of Open Access Journals (Sweden)

    Aleksic Petar S

    2002-01-01

    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.

  15. Music and audio - oh how they can stress your network

    Science.gov (United States)

    Fletcher, R.

    Nearly ten years ago a paper written by the Audio Engineering Society (AES)[1] made a number of interesting statements: 1. 2. The current Internet is inadequate for transmitting music and professional audio. Performance and collaboration across a distance stress beyond acceptable bounds the quality of service Audio and music provide test cases in which the bounds of the network are quickly reached and through which the defects in a network are readily perceived. Given these key points, where are we now? Have we started to solve any of the problems from the musician's point of view? What is it that musician would like to do that can cause the network so many problems? To understand this we need to appreciate that a trained musician's ears are extremely sensitive to very subtle shifts in temporal materials and localisation information. A shift of a few milliseconds can cause difficulties. So, can modern networks provide the temporal accuracy demanded at this level? The sample and bit rates needed to represent music in the digital domain is still contentious, but a general consensus in the professional world is for 96 KHz and IEEE 64-bit floating point. If this was to be run between two points on the network across 24 channels in near real time to allow for collaborative composition/production/performance, with QOS settings to allow as near to zero latency and jitter, it can be seen that the network indeed has to perform very well. Lighting the Blue Touchpaper for UK e-Science - Closing Conference of ESLEA Project The George Hotel, Edinburgh, UK 26-28 March, 200

  16. Informed spectral analysis: audio signal parameter estimation using side information

    Science.gov (United States)

    Fourer, Dominique; Marchand, Sylvain

    2013-12-01

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

  17. A digital audio playback system with USB interface

    OpenAIRE

    Karlsen, Espen; Tørresen, Magne

    2009-01-01

    A high performance sound card is designed and implemented using a USB enabled microcontroller and an external dataconverter. Data is retrieved either via USB or S/PDIF. The sampling clock is generated by a precision clock synthesizer. This is programmable and can be adapted to different sampling rates of USB data. The system supports 24 bit, 192 kHz audio. Signal attenuation is performed through a relay based stepped voltage divider with constant output impedance. 64 dB attenuation in steps...

  18. 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

  19. 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 in...... order to augment the perception of the 3D space. We compare different interaction paradigms implemented using these interfaces, aiming to increase speed and accuracy and reduce the need for constant visual feedback. While the LEAP Motion relies upon visual perception and proprioception, users can forego...

  20. Lost Audio Packets Steganography: The First Practical Evaluation

    CERN Document Server

    Mazurczyk, Wojciech

    2011-01-01

    This paper presents first experimental results for an IP telephony-based steganographic method called LACK (Lost Audio PaCKets steganography). This method utilizes the fact that in typical multimedia communication protocols like RTP (Real-Time Transport Protocol), excessively delayed packets are not used for the reconstruction of transmitted data at the receiver, i.e. these packets are considered useless and discarded. The results presented in this paper were obtained basing on a functional LACK prototype and show the method's impact on the quality of voice transmission. Achievable steganographic bandwidth for the different IP telephony codecs is also calculated.

  1. 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

  2. DOA Estimation of Audio Sources in Reverberant Environments

    DEFF Research Database (Denmark)

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

    2016-01-01

    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...... bias. Our simulation results show that we can estimate the DOA of the desired signal more accurately with this procedure compared to state-of-theart estimator in both synthetic and real data experiments with reverberation....

  3. Frequency Response Of Series And Parallel Combination Of Two Single Feedback Class D Amplifier

    OpenAIRE

    R. Sivarajan; K. R. Padmavathy; A.kasthoory; N. Santha; S. Subashri

    2015-01-01

    Amplifier is generally used to increase the amplitude of the signal. Consider the audio signal, if the amplitude of the audio signal is increased, then automatically loudness of the signal is increased. Class D Amplifier (CDA) consists of integrator, PWM modulator and output stage. This system can be defined as a open loop CDA system. The main factor to be considered for any amplifier circuit is the gain which is obtained by frequency response. To improve the gain of the amplifier, ...

  4. Audio-vestibulary changes following radiation treatment of the epipharynx

    International Nuclear Information System (INIS)

    Audio-vestibular changes are described following radiation treatment of the epipharynx. A total of 83 patients were observed prior to and following radiotherapy. In 70 of them hearing power was diminished: sound perceiving decrease was recorded in 43 patients (51,8 per cent), mixed type of diminished hearing power in 16 patients (19,2 per cent) and sound conduction decrease in 11 patients (13,2 per cent). Apart from the clinical method for testing the power of hearing (whispering and conversation) tonal threshold audiometry is also employed. The patients were asked for vestibulary complaints: the latter were established in seven patients. The studies reveal that radiation therapy affects in a lesser degree the middle ear and more substantially the inner ear with an unpairment of the functions of the labyrinth. These first investigations warrant a more specialized observation (by means of more numerous audiometric tests) and more continuous period of observation with frequent control audiograms. The observed more severe decrease of the power of hearing long after completion of radiation therapy are indicative as regards the process in the organ of Corti. High significance is attributed by the author to these observations who assumes that along with correct planning of radiation therapy rehabilitation of the power of hearing and stimulation therapy of the audio-vestibulary apparatus should simultaneously be initiated. (author)

  5. 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; ·     �...

  6. Delivering Audio and Video with Rich Site Summary Enclosures

    Directory of Open Access Journals (Sweden)

    Kevin Curran

    2006-04-01

    Full Text Available Rich Site Summary (RSS technology is a web content syndication format commonly used to organise news and the content of news-like sites. Indeed any information that can be broken down into ‘discrete items’ can be syndicated via RSS. RSS has been extended to include tags known as RSS enclosures which allow audio and video to be described in terms of location, duration and size. Once data about each element is established in an RSS format, a program called an aggregator can check the feed frequently for changes and updates and act accordingly. This paper documents a scheduled RSS Multimedia prototype which utilises idle computer time (at night to subscribe to media RSS channels in order to download audio and video content. Thus when the user arrives in the morning, there are fresh clips ‘sitting on the desktop’. Now when a video link is now clicked, it starts playing immediately, as it is already on the local disk thus the wait is zero.

  7. Noise Adaptive Stream Weighting in Audio-Visual Speech Recognition

    Directory of Open Access Journals (Sweden)

    Martin Heckmann

    2002-11-01

    Full Text Available It has been shown that integration of acoustic and visual information especially in noisy conditions yields improved speech recognition results. This raises the question of how to weight the two modalities in different noise conditions. Throughout this paper we develop a weighting process adaptive to various background noise situations. In the presented recognition system, audio and video data are combined following a Separate Integration (SI architecture. A hybrid Artificial Neural Network/Hidden Markov Model (ANN/HMM system is used for the experiments. The neural networks were in all cases trained on clean data. Firstly, we evaluate the performance of different weighting schemes in a manually controlled recognition task with different types of noise. Next, we compare different criteria to estimate the reliability of the audio stream. Based on this, a mapping between the measurements and the free parameter of the fusion process is derived and its applicability is demonstrated. Finally, the possibilities and limitations of adaptive weighting are compared and discussed.

  8. 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 be demonstr...

  9. A comparative study on automatic audio-visual fusion for aggression detection using meta-information

    NARCIS (Netherlands)

    Lefter, I.; Rothkrantz, L.J.M.; Burghouts, G.J.

    2013-01-01

    Multimodal fusion is a complex topic. For surveillance applications audio-visual fusion is very promising given the complementary nature of the two streams. However, drawing the correct conclusion from multi-sensor data is not straightforward. In previous work we have analysed a database with audio-

  10. 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 ...

  11. The SWRL Audio Laboratory System (ALS): An Integrated Configuration for Psychomusicology Research. Technical Report 51.

    Science.gov (United States)

    Williams, David Brian; Hoskin, Richard K.

    This report describes features of the Audio Laboratory System (ALS), a device which supports research activities of the Southwest Regional Laboratory's Music Program. The ALS is used primarily to generate recorded audio tapes for psychomusicology research related to children's perception and learning of music concepts such as pitch, loudness,…

  12. 78 FR 18416 - Sixth Meeting: RTCA Special Committee 226, Audio Systems and Equipment

    Science.gov (United States)

    2013-03-26

    ... Federal Aviation Administration Sixth Meeting: RTCA Special Committee 226, Audio Systems and Equipment... Notice of RTCA Special Committee 226, Audio Systems and Equipment. SUMMARY: The FAA is issuing this... Equipment. DATES: The meeting will be held April 15-17, 2013 from 9:00 a.m.-5:00 p.m. ADDRESSES: The...

  13. 77 FR 37733 - Third Meeting: RTCA Special Committee 226, Audio Systems and Equipment

    Science.gov (United States)

    2012-06-22

    ... Federal Aviation Administration Third Meeting: RTCA Special Committee 226, Audio Systems and Equipment... Notice of RTCA Special Committee 226, Audio Systems and Equipment. SUMMARY: The FAA is issuing this... Equipment. DATES: The meeting will be held July 10-12, 2012, from 9 a.m.-5 p.m. ADDRESSES: The meeting...

  14. 78 FR 57673 - Eighth Meeting: RTCA Special Committee 226, Audio Systems and Equipment

    Science.gov (United States)

    2013-09-19

    ... Federal Aviation Administration Eighth Meeting: RTCA Special Committee 226, Audio Systems and Equipment... Notice of RTCA Special Committee 226, Audio Systems and Equipment. SUMMARY: The FAA is issuing this... Equipment. DATES: The meeting will be held October 8-10, 2012 from 9:00 a.m.-5:00 p.m. ADDRESSES:...

  15. 78 FR 38093 - Seventh Meeting: RTCA Special Committee 226, Audio Systems and Equipment

    Science.gov (United States)

    2013-06-25

    ... Federal Aviation Administration Seventh Meeting: RTCA Special Committee 226, Audio Systems and Equipment... Notice of RTCA Special Committee 226, Audio Systems and Equipment. SUMMARY: The FAA is issuing this... Equipment ] DATES: The meeting will be held July 15-19, 2013 from 9:00 a.m.-5:00 p.m. ADDRESSES: The...

  16. 77 FR 58209 - Fourth Meeting: RTCA Special Committee 226, Audio Systems and Equipment

    Science.gov (United States)

    2012-09-19

    ... Federal Aviation Administration Fourth Meeting: RTCA Special Committee 226, Audio Systems and Equipment... notice of RTCA Special Committee 226, Audio Systems and Equipment. SUMMARY: The FAA is issuing this... Equipment. DATES: The meeting will be held October 16-18, 2012 from 9 a.m.-5 p.m. ADDRESSES: The...

  17. Adaptive Quantization Index Modulation Audio Watermarking based on Fuzzy Inference System

    Directory of Open Access Journals (Sweden)

    Sunita V. Dhavale

    2014-02-01

    Full Text Available Many of the adaptive watermarking schemes reported in the literature consider only local audio signal properties. Many schemes require complex computation along with manual parameter settings. In this paper, we propose a novel, fuzzy, adaptive audio watermarking algorithm based on both global and local audio signal properties. The algorithm performs well for dynamic range of audio signals without requiring manual initial parameter selection. Here, mean value of energy (MVE and variance of spectral flux (VSF of a given audio signal constitutes global components, while the energy of each audio frame acts as local component. The Quantization Index Modulation (QIM step size Δ is made adaptive to both the global and local features. The global component automates the initial selection of Δ using the fuzzy inference system while the local component controls the variation in it based on the energy of individual audio frame. Hence Δ adaptively controls the strength of watermark to meet both the robustness and inaudibility requirements, making the system independent of audio nature. Experimental results reveal that our adaptive scheme outperforms other fixed step sized QIM schemes and adaptive schemes and is highly robust against general attacks.

  18. The Language System of Audio Description: An Investigation as a Discursive Process

    Science.gov (United States)

    Piety, Philip J.

    2004-01-01

    This study investigated the language used in a selection of films containing audio description and developed a set of definitions that allow productions containing it to be more fully defined, measured, and compared. It also highlights some challenging questions related to audio description as a discursive practice and provides a basis for future…

  19. 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 codi

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

    NARCIS (Netherlands)

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

    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 benchmark

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

    Science.gov (United States)

    Merry, Stephen; Orsmond, Paul

    2008-01-01

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

  2. 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…

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

    Science.gov (United States)

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

    2013-01-01

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

  4. Investigating Expectations and Experiences of Audio and Written Assignment Feedback in First-Year Undergraduate Students

    Science.gov (United States)

    Fawcett, Hannah; Oldfield, Jeremy

    2016-01-01

    Previous research suggests that audio feedback may be an important mechanism for facilitating effective and timely assignment feedback. The present study examined expectations and experiences of audio and written feedback provided through "turnitin for iPad®" from students within the same cohort and assignment. The results showed that…

  5. 76 FR 591 - Determination of Rates and Terms for Preexisting Subscription and Satellite Digital Audio Radio...

    Science.gov (United States)

    2011-01-05

    ..., respectively. 72 FR 71795 (December 19, 2007), 73 FR 4080 (January 24, 2008). Section 804(b)(3)(B) of the... Audio Radio Services AGENCY: Copyright Royalty Board, Library of Congress. ACTION: Notice announcing... subscription and satellite digital audio radio services for the digital performance of sound recordings and...

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

    Science.gov (United States)

    Sha, Guoquan

    2010-01-01

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

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

    Science.gov (United States)

    Weld, Christopher

    2014-01-01

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

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

    Science.gov (United States)

    Bauer, Sara

    2011-01-01

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

  9. Audio Environment Recognition using Zero Crossing Features and MPEG-7 Descriptors

    Directory of Open Access Journals (Sweden)

    Saleh Al-Zhrani

    2010-01-01

    Full Text Available Problem statement: This study investigated zero crossing features and selected MPEG-7 audio descriptors for environment sound recognition applications such as audio forensics. Approach: The study implemented several experiments focusing on the problems of environment recognition from audio particularly for forensic applications. Results: It was investigated the effect of the temporal zero crossing feature as well as selected MPEG-7 audio low level descriptors on environment sound recognition. The performance was evaluated against a varying number of training sounds and samples per training file. Conclusion/Recommendations: Experimental results showed that higher recognition accuracy is achieved by increasing the number of training files and by decreasing the number of samples per training file. This study presented an audio environment recognition using zero crossing features and MPEG-7 Descriptors.

  10. 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 effic...... 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.......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...

  11. 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.

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

    International Nuclear Information System (INIS)

    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.

  13. 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.

  14. Increasing observer objectivity with audio-visual technology: the Sphygmocorder.

    Science.gov (United States)

    Atkins; O'Brien; Wesseling; Guelen

    1997-10-01

    The most fallible component of blood pressure measurement is the human observer. The traditional technique of measuring blood pressure does not allow the result of the measurement to be checked by independent observers, thereby leaving the method open to bias. In the Sphygmocorder, several components used to measure blood pressure have been combined innovatively with audio-visual recording technology to produce a system consisting of a mercury sphygmomanometer, an occluding cuff, an automatic inflation-deflation source, a stethoscope, a microphone capable of detecting Korotkoff sounds, a camcorder and a display screen. The accuracy of the Sphygmocorder against the trained human observer has been confirmed previously using the protocol of the British Hypertension Society and in this article the updated system incorporating a number of innovations is described. PMID:10234128

  15. Real-time Covert Communications Channel for Audio Signals

    Directory of Open Access Journals (Sweden)

    Ashraf Seleym

    2012-09-01

    Full Text Available Covert communications channel is considered as a type of secure communications that creates capability to transfer information between entities while hiding the contents of the channel. Multimedia data hiding techniques can be used to establish a covert channel for secret communications within a media carrier. In this paper, a high-rate covert communications channel is developed to exploit an audio stream as a carrier signal using multiple embedding in the Quantization Index Modulation framework. The proposed approach uses multi quantization vectors to increase data transmission rate. The embedding algorithms consider the embedding process as a communications problem, that it uses structured scheme of Multiple Trellis-Coded Quantization jointed with Multiple Trellis-Coded Modulation. Using convolution codes based trellis coding returns a real-time communications, because it can be continuously encoded and decoded. The proposed approach exhibits a high channel capacity due to the increase in data embedding rate without severely increasing in embedding distortion.

  16. Direct observation of magnetic patterns in audio tapes

    International Nuclear Information System (INIS)

    Magneto-optical imaging has become a powerful technique for the measurement of local magnetic fields. The technique consists in measuring the rotation in the light polarization plane when light travels through a transparent sensitive garnet (Ytrium Iron Garnet, YIG). The rotation angle is a function of the magnetic field at the YIG location. We have studied commercial audio tapes in which computer generated functions were recorded. We present a study of the stray field of periodically magnetized tapes with square and sawtooth waveforms. Observations are described modeling the magnetized tapes as an array of coaxial circular coils where the current in each coil reproduces the recorded functions. The effect of the magnetic field components, normal and parallel to the YIG surface, is discussed

  17. An Audio Haptic Tool for Visually Impaired Web Users

    Directory of Open Access Journals (Sweden)

    Catherine A. Todd

    2012-08-01

    Full Text Available An application that functions as a browser has been created for visually impaired Web users. It is distinctive from popular browsers such as Internet Explorer, Firefox and Google Chrome as it is designed specifically for partially sighted and blind Web users. The custom browser provides the feel of website elements via sense of touch (Haptic feedback, combined with speech synthesis. Force and sound feedback are provided through a virtual reality environment containing representations of 3D Web elements from the accessed website. This paper presents preliminary findings from the browser design and implementation of Haptic and audio Web element representation. Future work aims at creating a complete 3D Web browser with an interactive GUI that supports various features of existing browsers such as History, Favorites, Help and Image data.

  18. 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.

  19. 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...... moods in 22 separate users and examining the influence of applying affective offset to the users' sessions. Results show that, in the case when affective offset was applied, better user satisfaction was achieved: the average ratings went from 7.80 up to 8.65, with an average decrease in the number of...

  20. 77 FR 42764 - Distribution of the 2005, 2006, 2007 and 2008 Digital Audio Recording Technology Royalty Funds...

    Science.gov (United States)

    2012-07-20

    ... Copyright Royalty Board Distribution of the 2005, 2006, 2007 and 2008 Digital Audio Recording Technology... the digital audio recording technology royalty fees in the 2005, 2006, 2007 and 2008 Musical Works... royalties on digital audio recording devices and media that are distributed in the United States. 17...

  1. Comparing the Effects of Classroom Audio-Recording and Video-Recording on Preservice Teachers' Reflection of Practice

    Science.gov (United States)

    Bergman, Daniel

    2015-01-01

    This study examined the effects of audio and video self-recording on preservice teachers' written reflections. Participants (n = 201) came from a secondary teaching methods course and its school-based (clinical) fieldwork. The audio group (n[subscript A] = 106) used audio recorders to monitor their teaching in fieldwork placements; the video group…

  2. Something for Everyone? An Evaluation of the Use of Audio-Visual Resources in Geographical Learning in the UK.

    Science.gov (United States)

    McKendrick, John H.; Bowden, Annabel

    1999-01-01

    Reports from a survey of geographers that canvassed experiences using audio-visual resources to support teaching. Suggests that geographical learning has embraced audio-visual resources and that they are employed effectively. Concludes that integration of audio-visual resources into mainstream curriculum is essential to ensure effective and…

  3. 47 CFR 25.144 - Licensing provisions for the 2.3 GHz satellite digital audio radio service.

    Science.gov (United States)

    2010-10-01

    ... digital audio radio service. 25.144 Section 25.144 Telecommunication FEDERAL COMMUNICATIONS COMMISSION....144 Licensing provisions for the 2.3 GHz satellite digital audio radio service. (a) Qualification... digital audio radio service in the 2310-2360 MHz band shall describe in detail the proposed...

  4. Audio-Visual and Autogenic Relaxation Alter Amplitude of Alpha EEG Band, Causing Improvements in Mental Work Performance in Athletes.

    Science.gov (United States)

    Mikicin, Mirosław; Kowalczyk, Marek

    2015-09-01

    The aim of the present study was to investigate the effect of regular audio-visual relaxation combined with Schultz's autogenic training on: (1) the results of behavioral tests that evaluate work performance during burdensome cognitive tasks (Kraepelin test), (2) changes in classical EEG alpha frequency band, neocortex (frontal, temporal, occipital, parietal), hemisphere (left, right) versus condition (only relaxation 7-12 Hz). Both experimental (EG) and age-and skill-matched control group (CG) consisted of eighteen athletes (ten males and eight females). After 7-month training EG demonstrated changes in the amplitude of mean electrical activity of the EEG alpha bend at rest and an improvement was significantly changing and an improvement in almost all components of Kraepelin test. The same examined variables in CG were unchanged following the period without the intervention. Summing up, combining audio-visual relaxation with autogenic training significantly improves athlete's ability to perform a prolonged mental effort. These changes are accompanied by greater amplitude of waves in alpha band in the state of relax. The results suggest usefulness of relaxation techniques during performance of mentally difficult sports tasks (sports based on speed and stamina, sports games, combat sports) and during relax of athletes. PMID:26016588

  5. An audio-magnetotelluric investigation of the eastern margin of the Mamfe Basin, Cameroon

    International Nuclear Information System (INIS)

    Audio-magnetotelluric (AMT) data has been used to study the eastern margin of the Mamfe sedimentary basin along two profiles. Both profiles run across the sedimentary-metamorphic transition zone in this part of the basin. A 1-D interpretation of these data has been carried out using frequency profiling, pseudosections and geoelectric sections. Studying the propagation of the electric field at each station also gives an initial qualitative understanding of the possible layering of the subsurface at the station. A dioritic basement intrusion into the sediments has been identified along one of these profiles and a granitic intrusion under the other. Faults have been identified along both profiles marking the transition from sedimentary to metamorphic rocks at the eastern edge of the basin. However, this transition is complex and not smooth. This complexity can probably be explained by the fact that regional lithospheric stretching must have been responsible for the formation of this basin resulting in faulting in the eastern margin, thus strengthening the link between this basin and the Benue Trough of Nigeria. (author)

  6. A High Performance Sigma-Delta ADC for Audio Decoder Chip

    Directory of Open Access Journals (Sweden)

    Yu Fan

    2013-11-01

    Full Text Available This paper gives a high performance sigma delta Analog to Digital Converter (ADC applied in computer audio decoder chip. In this design, a 3rd-order single-loop CIFF topology is chosen to achieve the high performance ADC. Its signal bandwidth is 20KHz, sampling frequency is 10.24MHz and oversampling ratio is 256. Local feedback coefficient is used to reduce quantization noise. The non-linear model of modulator is given and the stability is analyzed. It is got that when quantizer gain is bigger than 0.322 the system is stable. According to simulation, Signal to Noise Ratio (SNR is 123.1dB and Effective Number of Bits (ENOB is 20.15bits. When input level is bigger than -3dBFs, the modulator is overload and becomes unstable. Then the integrator, quantizer and feed forward summation in ADC circuit are designed.  Then the ADC is implemented in 0.6um CMOS process, and the test result shows that its performance is 99.28dB.  

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

    OpenAIRE

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

    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 mecha...

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

    Directory of Open Access Journals (Sweden)

    Luiz M. G. Gonçalves

    2012-02-01

    Full Text Available 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.

  9. When the third party observer of a neuropsychological evaluation is an audio-recorder.

    Science.gov (United States)

    Constantinou, Marios; Ashendorf, Lee; McCaffrey, Robert J

    2002-08-01

    The presence of third parties during neuropsychological evaluations is an issue of concern for contemporary neuropsychologists. Previous studies have reported that the presence of an observer during neuropsychological testing alters the performance of individuals under evaluation. The present study sought to investigate whether audio-recording affects the neuropsychological test performance of individuals in the same way that third party observation does. In the presence of an audio-recorder the performance of the participants on memory tests declined. Performance on motor tests, on the other hand, was not affected by the presence of an audio-recorder. The implications of these findings in forensic neuropsychological evaluations are discussed. PMID:12607152

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

    DEFF Research Database (Denmark)

    Christensen, Mads Græsbøll; Sturm, Bob L.

    In this paper, we consider the problem of finding sparse representations of audio signals for coding purposes. In doing so, it is of utmost importance that when only a subset of the present components of an audio signal are extracted, it is the perceptually most important ones. To this end, we...... using standard software. A prominent feature of the new method is that it solves a problem that is closely related to the objective of coding, namely rate-distortion optimization. In computer simulations, we demonstrate the properties of the algorithm and its application to real audio signals....

  11. Audio watermarking based on psychoacoustic model and critical band wavelet transform

    Institute of Scientific and Technical Information of China (English)

    TAO Zhi; ZHAO Heming; GU Jihua; WU Di

    2007-01-01

    Watermark embedding algorithm based on critical band wavelet transform of digital audio signal is proposed in this paper. The masking threshold for each audio signal segment was calculated on the basic of psychoacoustic model. According to the similarity between critical band of human auditory system and critical band wavelet transform, a watermark was embedded into the low-band and mid-band coefficients of digital wavelet. The embedding strength was adaptively controlled by the masking threshold. The experiment results show that the embedded watermark signal is inaudible, and the watermarked audio signal has good robustness against many attacks such as compression, noise, re-sampling, low-pass filtering.

  12. 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 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 experimental results from prototype amplifier...

  13. Deus ex Machina – The audio production for Adel Abidin's “Blueprint”

    OpenAIRE

    Jokinen, Kalle

    2014-01-01

    Deus Ex Machina is a generative audio system produced for Finnish-Iraqi artist Adel Abidin's mixed-media installation ”Blueprint”. The work was exhibited in Sharjah, United Arab Emirates between 11th March and 16th June 2013. I was commissioned to produce and manage the audio part of the installation including the audio content and the implementation at the location. The written part of my thesis is a report of this adventurous and demanding process, which follows the project from the ear...

  14. 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)

  15. 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...

  16. 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.

  17. Emerging magnetic technologies for consumer audio/video (invited)

    Science.gov (United States)

    Fujiwara, Hideo

    1993-05-01

    In the field of consumer audio/video, digital technology is the natural path for advancement. In audio systems, it has just been introduced in the form of digital compact cassette tape recorder and mini disk system in which magneto-optical recording is used. Therefore, the digital video instruments, such as small cassette digital video tape recorders (VTR) for high-definition television and static or short moving image devices will be the next attractive goals to be achieved. Technology analysis shows that the basic techniques of signal processing for small cassette VTR, in which the recording density of 1 μm2/bit and the data rate of 130-160 Mb/s are required, are almost at hand. For the media, improved metal-evaporated tape including oblique incident Co-CoO will be the most favorable candidate, although particulate media of metal powder tape (including Co or N modified forms) or Ba-ferrite tape coated by using the newly developed ultrathin coating techniques may take its position with the help of enhanced head sensitivity, because of their better durability. The basic demand for increasing saturation induction Bs of the head materials will be fulfilled by fine grain Fe-based materials, the available Bs being as high as 2 T. A dramatic increase in magnetoresistive (MR) head sensitivity is expected by using the giant MR effect, although a breakthrough is required to extract its full potential. On the other hand, varieties of static (including short moving) image devices will be developed by use of rapidly progressing magneto-optical recording technology. Preliminary techniques are now ready to meet the demand for the required increase in bit rate and recording density. A bit rate of 64 Mbits/s (8 Mbytes/s) has already been attained. A bit density of 107 bits/mm2 (6.5 Gb/in.2) is likely to be realized, and a factor of 2 or 3 more enhancement can be counted on. Rigid disk and perpendicular recording will have a chance to participate in the field if their techniques

  18. Stream Weight Training Based on MCE for Audio-Visual LVCSR

    Institute of Scientific and Technical Information of China (English)

    LIU Peng; WANG Zuoying

    2005-01-01

    In this paper we address the problem of audio-visual speech recognition in the framework of the multi-stream hidden Markov model. Stream weight training based on minimum classification error criterion is discussed for use in large vocabulary continuous speech recognition (LVCSR). We present the lattice re-scoring and Viterbi approaches for calculating the loss function of continuous speech. The experimental results show that in the case of clean audio, the system performance can be improved by 36.1% in relative word error rate reduction when using state-based stream weights trained by a Viterbi approach, compared to an audio only speech recognition system. Further experimental results demonstrate that our audio-visual LVCSR system provides significant enhancement of robustness in noisy environments.

  19. Automatic Segmentation of News Items Based on Video and Audio Features

    Institute of Scientific and Technical Information of China (English)

    王伟强; 高文

    2002-01-01

    The automatic segmentation of news items is a key for implementing the automatic cataloging system of news video. This paper presents an approach which manages audio and video feature information to automatically segment news items. The integration of audio and visual analyses can overcome the weakness of the approach using only image analysis techniques. It makes the approach more adaptable to various situations of news items. The proposed approach detects silence segments in accompanying audio, and integrates them with shot segmentation results, as well as anchor shot detection results, to determine the boundaries among news items. Experimental results show that the integration of audio and video features is an effective approach to solving the problem of automatic segmentation of news items.

  20. 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.

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

    DEFF Research Database (Denmark)

    Dimitrov, Smilen; Serafin, Stefania

    2014-01-01

    This article investigates how lacking suitable platforms for laboratory exercises becomes a learning problem, limiting the practical experience students gain. In engineering education, laboratory demonstration difficulty of issues like real-time streaming in digital signal and audio processing is...

  2. How to gain gain a reference book on triodes in audio pre-amps

    CERN Document Server

    Vogel, Burkhard

    2008-01-01

    This book gives a detailed insight into the most common gain producing and constant current generating possibilities (28) of triodes for audio pre-amplifier purposes. It shows how to calculate first and spend money for expensive components later.

  3. 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.

  4. Learning one-to-many mapping functions for audio-visual integrated perception

    Science.gov (United States)

    Lim, Jung-Hui; Oh, Do-Kwan; Lee, Soo-Young

    2010-04-01

    In noisy environment the human speech perception utilizes visual lip-reading as well as audio phonetic classification. This audio-visual integration may be done by combining the two sensory features at the early stage. Also, the top-down attention may integrate the two modalities. For the sensory feature fusion we introduce mapping functions between the audio and visual manifolds. Especially, we present an algorithm to provide one-to-many mapping function for the videoto- audio mapping. The top-down attention is also presented to integrate both the sensory features and classification results of both modalities, which is able to explain McGurk effect. Each classifier is separately implemented by the Hidden-Markov Model (HMM), but the two classifiers are combined at the top level and interact by the top-down attention.

  5. 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

    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-emotion...... 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........ 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...

  6. 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.

  7. Joint Audio-Visual Tracking Using Particle Filters

    Directory of Open Access Journals (Sweden)

    Dmitry N. Zotkin

    2002-11-01

    Full Text Available It is often advantageous to track objects in a scene using multimodal information when such information is available. We use audio as a complementary modality to video data, which, in comparison to vision, can provide faster localization over a wider field of view. We present a particle-filter based tracking framework for performing multimodal sensor fusion for tracking people in a videoconferencing environment using multiple cameras and multiple microphone arrays. One advantage of our proposed tracker is its ability to seamlessly handle temporary absence of some measurements (e.g., camera occlusion or silence. Another advantage is the possibility of self-calibration of the joint system to compensate for imprecision in the knowledge of array or camera parameters by treating them as containing an unknown statistical component that can be determined using the particle filter framework during tracking. We implement the algorithm in the context of a videoconferencing and meeting recording system. The system also performs high-level semantic analysis of the scene by keeping participant tracks, recognizing turn-taking events and recording an annotated transcript of the meeting. Experimental results are presented. Our system operates in real-time and is shown to be robust and reliable.

  8. 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.

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

    Science.gov (United States)

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

    2016-05-01

    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 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. PMID:26625860

  10. User requirements for multimedia indexing and retrieval of unedited audio-visual footage - RUSHES

    OpenAIRE

    Schreer, O; Fuentes Ardeo, L; Sotiriou, D.; Sadka, A.H.; Izquierdo, E

    2008-01-01

    Multimedia analysis and reuse of raw un-edited audio visual content known as rushes is gaining acceptance by a large number of research labs and companies. A set of research projects are considering multimedia indexing, annotation, search and retrieval in the context of European funded research, but only the FP6 project RUSHES is focusing on automatic semantic annotation, indexing and retrieval of raw and un-edited audio-visual content. Even professional content creators and providers as well...

  11. 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...

  12. Listen to the Music: Audio Preview Cues for Exploration of Online Music.

    OpenAIRE

    Schraefel, M. C.; Karam, Maria; Zhao, Shengdong

    2003-01-01

    This paper presents a novel mechanism that seeks to allow people to explore large collections of loosely structured audio. The approach provides a lightweight preview mechanism that allows people to explore the audio collection by providing supporting information (analogous to the use of tooltips in visual interfaces) We present an evaluation of these “preview cues” towards developing a design heuristics for their deployment.

  13. Collusion-Resistant Audio Fingerprinting System in the Modulated Complex Lapped Transform Domain

    OpenAIRE

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

    2013-01-01

    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 par...

  14. Audio textbook of Spanish for children afflicted with specific learning disability{--} with dyslexia

    OpenAIRE

    Lacinová, Lenka

    2008-01-01

    This work dedicates to education of foreign languages of children afflicted with specific learning disability, with dyslexia. Theoretic part is occupied by specific learning disability, above all by its reasons, tokens, diagnostic and reeducation. The second part can be used as a help in education of Spanish of children afflicted with specific learning disability, with dyslexia. It is audio textbook of Spanish which contains also CD that serves as a audio help for reading texts and for learni...

  15. 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)

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

    OpenAIRE

    John Mourjopoulos; Andreas Floros

    2007-01-01

    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 syste...

  17. An Artistic Technique for Audio-to-Video Translation on a Music Perception Study

    Directory of Open Access Journals (Sweden)

    Eugene Mikyung Kim

    2012-07-01

    Full Text Available The paper presents an audio-to-visual instrument that allows sound-to-image transformation based on anempirical investigation of the relationship between four auditory parameters – pitch, amplitude, timbre, and duration - and four visual parameters – color, location, shape, and size - in the multimedia context. Implementing the audio-to-visual instruments involves real-time sound analysis by using a constant-Q transform and image generation in a Max/MSP/Jitter environment.

  18. An Artistic Technique for Audio-to-Video Translation on a Music Perception Study

    Directory of Open Access Journals (Sweden)

    Eugene Mikyung Kim

    2012-06-01

    Full Text Available The paper presents an audio-to-visual instrument that allows sound-to-image transformation based on an empirical investigation of the relationship between four auditory parameters – pitch, amplitude, timbre, and duration - and four visual parameters – color, location, shape, and size - in the multimedia context. Implementing the audio-to-visual instruments involves real-time sound analysis by using a constant-Q transform and image generation in a Max/MSP/Jitter environment.

  19. 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...... 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....

  20. 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 ...

  1. Comparing Learning Gains: Audio Versus Text-based Instructor Communication in a Blended Online Learning Environment

    Science.gov (United States)

    Shimizu, Dominique

    Though blended course audio feedback has been associated with several measures of course satisfaction at the postsecondary and graduate levels compared to text feedback, it may take longer to prepare and positive results are largely unverified in K-12 literature. The purpose of this quantitative study was to investigate the time investment and learning impact of audio communications with 228 secondary students in a blended online learning biology unit at a central Florida public high school. A short, individualized audio message regarding the student's progress was given to each student in the audio group; similar text-based messages were given to each student in the text-based group on the same schedule; a control got no feedback. A pretest and posttest were employed to measure learning gains in the three groups. To compare the learning gains in two types of feedback with each other and to no feedback, a controlled, randomized, experimental design was implemented. In addition, the creation and posting of audio and text feedback communications were timed in order to assess whether audio feedback took longer to produce than text only feedback. While audio feedback communications did take longer to create and post, there was no difference between learning gains as measured by posttest scores when student received audio, text-based, or no feedback. Future studies using a similar randomized, controlled experimental design are recommended to verify these results and test whether the trend holds in a broader range of subjects, over different time frames, and using a variety of assessment types to measure student learning.

  2. 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...

  3. Stress Reduction through Audio Distraction in Anxious Pediatric Dental Patients: An Adjunctive Clinical Study

    OpenAIRE

    Singh, Divya; Samadi, Firoza; Jaiswal, JN; Tripathi, Abhay Mani

    2015-01-01

    ABSTRACT Aim: The purpose of the present study was to evaluate the eff-cacy of ‘audio distraction’ in anxious pediatric dental patients. Materials and methods: Sixty children were randomly selected and equally divided into two groups of thirty each. The first group was control group (group A) and the second group was music group (group B). The dental procedure employed was extraction for both the groups. The children included in music group were allowed to hear audio presentation throughout t...

  4. MINDING OUR WORDS: AUDIO RESPONSIBILITIES IN ENDANGERED LANGUAGES DOCUMENTATION AND ARCHIVING

    OpenAIRE

    David Nathan

    2008-01-01

    Linguists are addressing the predicted the loss of many of the world's languages through an emerging discipline called Language Documentation, which focuses not on theory but on data, and how the data is acquired, represented, presented, and preserved. For most endangered languages, which are not written, much of this data is audio, and unlike many corpora it is likely to be local, particular, opportunistic, and uneven. New questions are raised, such as: what audio data counts as a record of ...

  5. 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

    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...

  6. 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...

  7. Audio Structure Based on Android System%基于Android系统的音频架构

    Institute of Scientific and Technical Information of China (English)

    王峻

    2012-01-01

    介绍了Android编程模型和媒体处理的Audio部分,包括OPENCORE概念、音频系统在Java,JNI,NDK层的主要的类如AudioTrack和AudioFlinger.最后对其在数字电视机顶盒中的应用、与中间件的关系做了分析.

  8. Contributions à la séparation de sources et à la description des contenus audio

    OpenAIRE

    Vincent, Emmanuel

    2012-01-01

    Revised version including a bugfix in Figure 4.1. Audio data occupy a central position in our life, whether it is for spoken communication, personal videos, radio and television, music, cinema, video games, or live entertainment. This raises a range of application needs from signal enhancement to information retrieval, including content repurposing and interactive manipulation. Real-world audio data exhibit a complex structure due to the superposition of several sound sources and the coexi...

  9. Class D audio amplifier with 4th order output filter and self-oscillating full-state hysteresis based feedback driving capacitive transducers

    DEFF Research Database (Denmark)

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

    2014-01-01

    A practical solution is presented for the design of a non-isolated high voltage DC/AC power converter. The converter is intended to be used as a class D audio amplifier for a Dielectric Electro Active Polymer (DEAP) transducer. A simple and effective hysteretic control scheme for the converter (b...... (buck with fourth- order output filter) is developed and analyzed. The proposed design is verified experimentally by a 125 VAR prototype amplifier, capable of delivering a peak output voltage of 240 V within the frequency range of 100 Hz – 3.5 kHz. A peak efficiency of 87 % is reported....

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

    International Nuclear Information System (INIS)

    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

  11. 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.

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

    Science.gov (United States)

    Krebs, Eric M.

    2002-09-01

    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 were aimed at implementing acoustical models, such as reverberation and occlusion, and live audio streaming to any simulation employing this architecture, Several free or open-source sound APIs were evaluated, and DirectSound3DTM was selected as the core component of the audio architecture, Creative Technology Ltd, Environmental Audio Extensions (EAXTM 3,0) were integrated into the architecture to provide environmental effects such as reverberation, occlusion, obstruction, and exclusion, Voice over IP (VoIP) technology was evaluated to provide live, streaming voice to any virtual environment DirectVoice was selected as the voice component of the VoIP architecture due to its integration with DirectSound3DTM, However, extremely high latency considerations with DirectVoice, and any other VoIP application or software, required further research into alternative live voice architectures for inclusion in virtual environments Ausim3D's GoldServe Audio System was evaluated and integrated into the hardware component of the audio architecture to provide an extremely low-latency, live, streaming voice capability.

  13. Using virtual 3D audio in multispeech channel and multimedia environments

    Science.gov (United States)

    Orosz, Michael D.; Karplus, Walter J.; Balakrishnan, Jerry D.

    2000-08-01

    The advantages and disadvantages of using virtual 3-D audio in mission-critical, multimedia display interfaces were evaluated. The 3D audio platform seems to be an especially promising candidate for aircraft cockpits, flight control rooms, and other command and control environments in which operators must make mission-critical decisions while handling demanding and routine tasks. Virtual audio signal processing creates the illusion for a listener wearing conventional earphones that each of a multiplicity of simultaneous speech or audio channels is originating from a different, program- specified location in virtual space. To explore the possible uses of this new, readily available technology, a test bed simulating some of the conditions experienced by the chief flight test coordinator at NASA's Dryden Flight Research Center was designed and implemented. Thirty test subjects simultaneously performed routine tasks requiring constant hand-eye coordination, while monitoring four speech channels, each generating continuous speech signals, for the occurrence of pre-specified keywords. Performance measures included accuracy in identifying the keywords, accuracy in identifying the speaker of the keyword, and response time. We found substantial improvements on all of these measures when comparing virtual audio with conventional, monaural transmissions. We also explored the effect on operator performance of different spatial configurations of the audio sources in 3-D space, simulated movement (dither) in the source locations, and of providing graphical redundancy. Some of these manipulations were less effective and may even decrease performance efficiency, even though they improve some aspects of the virtual space simulation.

  14. 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.

  15. 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.

  16. Biochemistry on the Media: daily science in audio and video

    Directory of Open Access Journals (Sweden)

    B. P. Melo et al

    2014-08-01

    Full Text Available Biochemistry on the Media: daily science in audio and video Melo,B. P1; Henriques, L. R1; Júnior, H. G2; Galvão, G. R2; Costa, M. M2; Silva, A. S3; Costa, M. P3; Barreto, L. P3; Almeida, A. A3; Fontes, P. P3; Meireles, L. M3; Costa, P. A3; Costa, C. B3; Monteiro, L. M. O3 Konig, I. M3; Dias, B. K. M1; Santos, R. C. V1; Bagno, F. F1; Fernandes, L1; Alves, P. R1; Sales, F. M1; Martins, T. C. N1; Moreira, V. J. V1; Marchiori, J. M1; Medeiros, L.4; Leite, J. P. V5; Moraes, G. H. K6.   1 Members of ETP-Biochemistry UFV; 2 Students of program Jovens Talentos para a Ciência UFV; 3 Graduating Students of ETP; 4 Coordinator in Espaço Ciência UFV; 5 Pharmaceutical, professor at Molecular Biology and Biochemistry Department (BBD UFV, ETP’s tutor; 6 Agronomist, professor at BDD, work’s advisor.   INTRODUCTION: The Educational Tutorial Program in Biochemistry (ETP from UFV have worked in qualification of basic science teachers, offering courses about Biochemistry. In courses, was detected the necessity of a personal material to inspire them. To do it, ETP compiled some media spots in a box and have used it in qualification courses. OBJECTIVES: The objective of this work was construct a part of a permanent material to be used in courses to qualifications high school's teachers and evaluate it. METODOLOGY: Applying questionnaires to high school students, ETP's members had detected that these students don't have a solid idea about how is Biochemistry. Thus, themes about common Biochemistry daily things were elected to be transformed in spots to radio and television. Texts about shampoo composition, vegetable’s darkening, bread’s fermentation, etc, were written and a script done by Journalism’s students of Espaço Ciência(*. Finally, the spots were recorded and vehiculated on universitary channel. In 2013, the spots were compiled in a media box. It has been included in a permanent material used in qualification courses. According to ALBAGLI

  17. 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.

  18. The development and use of audio-visual technology in terms of economy and socio-economic trends in society

    OpenAIRE

    Mikšík, Jan

    2014-01-01

    The aim of this work is to describe history of audio-visual technology and to analyse the influence of digitalization. The text describes the history of cinematography, television and also the introduction of audio-visual technology to people's homes. It contains information on present situation as well as new trends and the influence of the Internet on audio-visual making. There is a comparison of past and present technologies. The new technologies are accessible even for amateur creators wh...

  19. From vibration to perception: using Large Multi-Actuator Panels (LaMAPs) to create coherent audio-visual environments

    OpenAIRE

    Rébillat, Marc; Corteel, Etienne; Katz, Brian,; Boutillon, Xavier

    2012-01-01

    International audience Virtual reality aims at providing users with audio-visual worlds where they will behave and learn as if they were in the real world. In this context, specific acoustic transducers are needed to fulfill simultaneous spatial requirements on visual and audio rendering in order to make them coherent. Large multi-actuator panels (LaMAPs) allow for the combined construction of a projection screen and loudspeaker array, and thus allows for the coherent creation of an audio ...

  20. A Conceptual Framework for the Analysis of First-Person Shooter Audio and its Potential Use for Game Engines

    OpenAIRE

    Mark Grimshaw; Gareth Schott

    2008-01-01

    We introduce and describe a new conceptual framework for the design and analysis of audio for immersive first-person shooter games, and discuss its potential implications for the development of the audio component of game engines. The framework was created in order to illustrate and acknowledge the direct role of in-game audio in shaping player-player interactions and in creating a sense of immersion in the game world. Furthermore, it is argued that the relationship between player and sou...