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Sample records for audio frequency

  1. Audio Classification from Time-Frequency Texture

    CERN Document Server

    Yu, Guoshen

    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.

  2. 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 u...

  3. 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.

  4. Audio frequency in vivo optical coherence elastography

    Energy Technology Data Exchange (ETDEWEB)

    Adie, Steven G; Kennedy, Brendan F; Armstrong, Julian J; Alexandrov, Sergey A; Sampson, David D [Optical-Biomedical Engineering Laboratory (OBEL), School of Electrical, Electronic and Computer Engineering, University of Western Australia, 35 Stirling Highway, Crawley, Western Australia 6009 (Australia)], E-mail: dsampson@ee.uwa.edu.au

    2009-05-21

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

  5. Extraction of ions and electrons from audio frequency plasma source

    Directory of Open Access Journals (Sweden)

    N. A. Haleem

    2016-09-01

    Full Text Available Herein, the extraction of high ion / electron current from an audio frequency (AF nitrogen gas discharge (10 – 100 kHz is studied and investigated. This system is featured by its small size (L= 20 cm and inner diameter = 3.4 cm and its capacitive discharge electrodes inside the tube and its high discharge pressure ∼ 0.3 Torr, without the need of high vacuum system or magnetic fields. The extraction system of ion/electron current from the plasma is a very simple electrode that allows self-beam focusing by adjusting its position from the source exit. The working discharge conditions were applied at a frequency from 10 to 100 kHz, power from 50 – 500 W and the gap distance between the plasma meniscus surface and the extractor electrode extending from 3 to 13 mm. The extracted ion/ electron current is found mainly dependent on the discharge power, the extraction gap width and the frequency of the audio supply. SIMION 3D program version 7.0 package is used to generate a simulation of ion trajectories as a reference to compare and to optimize the experimental extraction beam from the present audio frequency plasma source using identical operational conditions. The focal point as well the beam diameter at the collector area is deduced. The simulations showed a respectable agreement with the experimental results all together provide the optimizing basis of the extraction electrode construction and its parameters for beam production.

  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. Comparisons of Audio and Audiovisual Measures of Stuttering Frequency and Severity in Preschool-Age Children

    Science.gov (United States)

    Rousseau, Isabelle; Onslow, Mark; Packman, Ann; Jones, Mark

    2008-01-01

    Purpose: To determine whether measures of stuttering frequency and measures of overall stuttering severity in preschoolers differ when made from audio-only recordings compared with audiovisual recordings. Method: Four blinded speech-language pathologists who had extensive experience with preschoolers who stutter measured stuttering frequency and…

  8. Frequency allocations for a new satellite service - Digital audio broadcasting

    Science.gov (United States)

    Reinhart, Edward E.

    1992-03-01

    The allocation in the range 500-3000 MHz for digital audio broadcasting (DAB) is described in terms of key issues such as the transmission-system architectures. Attention is given to the optimal amount of spectrum for allocation and the technological considerations relevant to downlink bands for satellite and terrestrial transmissions. Proposals for DAB allocations are compared, and reference is made to factors impinging on the provision of ground/satellite feeder links. The allocation proposals describe the implementation of 50-60-MHz bandwidths for broadcasting in the ranges near 800 MHz, below 1525 MHz, near 2350 MHz, and near 2600 MHz. Three specific proposals are examined in terms of characteristics such as service areas, coverage/beam, channels/satellite beam, and FCC license status. Several existing problems are identified including existing services crowded with systems, the need for new bands in the 1000-3000-MHz range, and variations in the nature and intensity of implementations of existing allocations that vary from country to country.

  9. 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...

  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. Mount St. Helens: Controlled-source audio-frequency magnetotelluric (CSAMT) data and inversions

    Science.gov (United States)

    Wynn, Jeff; Pierce, Herbert A.

    2015-01-01

    This report describes a series of geoelectrical soundings carried out on and near Mount St. Helens volcano, Washington, in 2010–2011. These soundings used a controlled-source audio-frequency magnetotelluric (CSAMT) approach (Zonge and Hughes, 1991; Simpson and Bahr, 2005). We chose CSAMT for logistical reasons: It can be deployed by helicopter, has an effective depth of penetration of as much as 1 kilometer, and requires less wire than a Schlumberger sounding.

  12. Perceptual Coding of Audio Signals Using Adaptive Time-Frequency Transform

    Directory of Open Access Journals (Sweden)

    Karthikeyan Umapathy

    2007-08-01

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

  13. Digital audio broadcasting: Comparison of coverage at different frequencies and with different bandwidths

    Science.gov (United States)

    Maddocks, M. C. D.; Pullen, I. R.

    A Digital Audio Broadcasting (DAB) system capable of reliable reception in vehicles and portables has been developed by the Eureka 147 project. This Report describes a set of experiments performed to investigate the effect on the coverage area by changing the bandwidth of the DAB signal and its transmit frequency band. It is concluded that the choice of a bandwidth for the DAB signal of approximately 1.5 MHz is suitable. This is because it is sufficiently wideband to provide a significant benefit in reducing the location variation of the total received signal power, while being narrow enough to allow suitable channelization within the existing frequency bands. It is also concluded that a frequency allocation below Band IV would be more suitable in order to provide satisfactory coverage for all types of reception from terrestrial DAB transmitters. Above this frequency, the effects of clutter and terrain undulations appear to significantly increase the problems of providing uniform coverage at low antenna heights.

  14. Structure-borne sound structural vibrations and sound radiation at audio frequencies

    CERN Document Server

    Cremer, L; Petersson, Björn AT

    2005-01-01

    Structure-Borne Sound"" is a thorough introduction to structural vibrations with emphasis on audio frequencies and the associated radiation of sound. The book presents in-depth discussions of fundamental principles and basic problems, in order to enable the reader to understand and solve his own problems. It includes chapters dealing with measurement and generation of vibrations and sound, various types of structural wave motion, structural damping and its effects, impedances and vibration responses of the important types of structures, as well as with attenuation of vibrations, and sound radi

  15. 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

    2016-01-29

    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.

  16. Digital audio broadcasting: Measuring techniques and coverage performance for a medium power VHF single frequency network

    Science.gov (United States)

    Maddocks, M. C. D.; Eng, C.; Pullen, I. R.; Green, J. A.

    1995-02-01

    The advent of digital formats such as CD has created demand for uniformly high audio quality from radio. In order to provide such high-quality stereo reception, a Digital Audio Broadcasting (DAB) system capable of reliable reception in vehicles and on portables has been developed by the European EUREKA 147 Project. As a VHF frequency allocation would appear most suitable for the introduction of terrestrial broadcasting of DAB in the United Kingdom, the BBC is undertaking a major experiment to test the EUREKA DAB system and to generate data to allow efficient planning of its transmitter network. A network of four, 1 kW e.r.p., VHF transmitters has been installed to cover the London area in England. This Report describes the experimental program and the rationale and measurement techniques behind it. The results show a wide-area coverage from the transmitter network which is in reasonable agreement with computer predictions. This indicates that the current transmitting and receiving equipment (built to the EUREKA specification) is operating in the way that would be expected from theoretical studies and simulation. The results also provide quantitative values which can be used for coverage prediction and for international co-ordination of services. Finally, the performance of the system demonstrates a number of the benefits of the EUREKA DAB system for mobile and portable reception.

  17. High-Resolution Audio with Inaudible High-Frequency Components Induces a Relaxed Attentional State without Conscious Awareness

    Science.gov (United States)

    Kuribayashi, Ryuma; Nittono, Hiroshi

    2017-01-01

    High-resolution audio has a higher sampling frequency and a greater bit depth than conventional low-resolution audio such as compact disks. The higher sampling frequency enables inaudible sound components (above 20 kHz) that are cut off in low-resolution audio to be reproduced. Previous studies of high-resolution audio have mainly focused on the effect of such high-frequency components. It is known that alpha-band power in a human electroencephalogram (EEG) is larger when the inaudible high-frequency components are present than when they are absent. Traditionally, alpha-band EEG activity has been associated with arousal level. However, no previous studies have explored whether sound sources with high-frequency components affect the arousal level of listeners. The present study examined this possibility by having 22 participants listen to two types of a 400-s musical excerpt of French Suite No. 5 by J. S. Bach (on cembalo, 24-bit quantization, 192 kHz A/D sampling), with or without inaudible high-frequency components, while performing a visual vigilance task. High-alpha (10.5–13 Hz) and low-beta (13–20 Hz) EEG powers were larger for the excerpt with high-frequency components than for the excerpt without them. Reaction times and error rates did not change during the task and were not different between the excerpts. The amplitude of the P3 component elicited by target stimuli in the vigilance task increased in the second half of the listening period for the excerpt with high-frequency components, whereas no such P3 amplitude change was observed for the other excerpt without them. The participants did not distinguish between these excerpts in terms of sound quality. Only a subjective rating of inactive pleasantness after listening was higher for the excerpt with high-frequency components than for the other excerpt. The present study shows that high-resolution audio that retains high-frequency components has an advantage over similar and indistinguishable digital

  18. High-Resolution Audio with Inaudible High-Frequency Components Induces a Relaxed Attentional State without Conscious Awareness.

    Science.gov (United States)

    Kuribayashi, Ryuma; Nittono, Hiroshi

    2017-01-01

    High-resolution audio has a higher sampling frequency and a greater bit depth than conventional low-resolution audio such as compact disks. The higher sampling frequency enables inaudible sound components (above 20 kHz) that are cut off in low-resolution audio to be reproduced. Previous studies of high-resolution audio have mainly focused on the effect of such high-frequency components. It is known that alpha-band power in a human electroencephalogram (EEG) is larger when the inaudible high-frequency components are present than when they are absent. Traditionally, alpha-band EEG activity has been associated with arousal level. However, no previous studies have explored whether sound sources with high-frequency components affect the arousal level of listeners. The present study examined this possibility by having 22 participants listen to two types of a 400-s musical excerpt of French Suite No. 5 by J. S. Bach (on cembalo, 24-bit quantization, 192 kHz A/D sampling), with or without inaudible high-frequency components, while performing a visual vigilance task. High-alpha (10.5-13 Hz) and low-beta (13-20 Hz) EEG powers were larger for the excerpt with high-frequency components than for the excerpt without them. Reaction times and error rates did not change during the task and were not different between the excerpts. The amplitude of the P3 component elicited by target stimuli in the vigilance task increased in the second half of the listening period for the excerpt with high-frequency components, whereas no such P3 amplitude change was observed for the other excerpt without them. The participants did not distinguish between these excerpts in terms of sound quality. Only a subjective rating of inactive pleasantness after listening was higher for the excerpt with high-frequency components than for the other excerpt. The present study shows that high-resolution audio that retains high-frequency components has an advantage over similar and indistinguishable digital sound

  19. High-performance combination method of electric network frequency and phase for audio forgery detection in battery-powered devices.

    Science.gov (United States)

    Savari, Maryam; Abdul Wahab, Ainuddin Wahid; Anuar, Nor Badrul

    2016-09-01

    Audio forgery is any act of tampering, illegal copy and fake quality in the audio in a criminal way. In the last decade, there has been increasing attention to the audio forgery detection due to a significant increase in the number of forge in different type of audio. There are a number of methods for forgery detection, which electric network frequency (ENF) is one of the powerful methods in this area for forgery detection in terms of accuracy. In spite of suitable accuracy of ENF in a majority of plug-in powered devices, the weak accuracy of ENF in audio forgery detection for battery-powered devices, especially in laptop and mobile phone, can be consider as one of the main obstacles of the ENF. To solve the ENF problem in terms of accuracy in battery-powered devices, a combination method of ENF and phase feature is proposed. From experiment conducted, ENF alone give 50% and 60% accuracy for forgery detection in mobile phone and laptop respectively, while the proposed method shows 88% and 92% accuracy respectively, for forgery detection in battery-powered devices. The results lead to higher accuracy for forgery detection with the combination of ENF and phase feature.

  20. Coverage aspects of a single frequency network designed for digital audio broadcasting

    Science.gov (United States)

    Bell, C. P.; Williams, W. F.

    Since the early 1960s, the VHF/FM pilot tone system has been used by broadcasters to supply high quality stereophonic programs to listeners equipped with fixed receivers and modest gain directional antennas mounted externally at a height of about ten meters above ground levels. More recently, with the advent of improved radio receivers and digital technology in the reproduction industry, the broadcasters' target audience has changed and listeners now expect high fidelity reception, comparable to CD quality, on their portable and mobile receivers. In many areas, the demand for high quality can be supplied using the existing VHF/FM networks. There are, however, areas where good portable and mobile reception cannot be obtained, due to shadowing and multipath. This report discusses the coded orthogonal frequency division multiplex (COFDM) channel coding and modulation system developed within the Eureka 147 digital audio broadcasting (DAB) project, which is able to overcome the problems of multipath. Indeed, it can make use of multipath signals. This fact leads to the concept of a Single Frequency Network (SFN) where a single DAB COFDM frequency block can be used to supply five or six programs to a whole country. The derivation of minimum required field strength levels for DAB COFDM signals, and the allowances required for high percentage location coverage, necessary for digital systems, are presented together with a spectrum efficiency comparison between SFN and VHF/FM networks. Theoretical lattice planning, using the CCIR Rec. 370 prediction method, is used initially to illustrate the SFN concept and suggest required effective radiated powers. The results of SFN coverage studies for the UK, using existing broadcast transmitter sites and utilizing both the Rec. 370 and the BBC terrain data based prediction methods, are given. In addition, a possible four frequency block SFN plan for Europe is discussed. The Report concludes that, in comparison, networks using the

  1. 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.;

    2007-01-01

    -frequency domain. Pure-tone carrier amplitude-modulation (AM) depth-discrimination thresholds were found to be similar using both traditional gated stimuli and using a temporally modulated fringe for a fixed standard depth (ms=0.25) and a range of AM frequencies (4-64 Hz). In a second experiment, masked sinusoidal......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...

  2. 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.;

    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...... were found to be the same in conditions with a continuous (modulated) carrier and with traditional gated stimuli for AM frequencies ranging from 4 –64 Hz. The second set of experiments compared the amount of CMR in a tone-in-noise detection task when slow, regular fluctuations were imposed...

  3. 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

  4. Audio Papers

    DEFF Research Database (Denmark)

    Groth, Sanne Krogh; Samson, Kristine

    2016-01-01

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

  5. Balancing Audio

    DEFF Research Database (Denmark)

    Walther-Hansen, Mads

    2016-01-01

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

  6. Relative-intensity squeezing at audio frequencies using four-wave mixing in an atomic vapor

    CERN Document Server

    McCormick, C F; Lett, P D; Marino, A M

    2007-01-01

    We demonstrate the use of four-wave mixing in hot atomic vapor to generate up to -7.1 dB of measured relative-intensity squeezing. Due to its intrinsic simplicity, our system is strongly decoupled from environmental noise, and we observe more than -4 dB of squeezing down to frequencies as low as 5 kHz. This robust source of narrowband squeezed light may be useful for a variety of applications, such as coupling to atomic ensembles and enhancing the sensitivity of photothermal spectroscopy.

  7. 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.

  8. 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....

  9. Three-dimensional audio-frequency magnetotelluric imaging of Akebasitao granitic intrusions in Western Junggar, NW China

    Science.gov (United States)

    Yang, Bo; Zhang, Anqi; Zhang, Sheng; Liu, Ying; Zhang, Shengye; Li, Yongtao; Xu, Yixian; Wang, Qinyan

    2016-12-01

    An audio-frequency magnetotelluric (AMT) array was deployed here to map the resistivity structure of the Akebasitao intrusions and the surrounding area in Western Junggar, China. High quality AMT data were acquired at 463 sites that covered the whole Akebasitao intrusions. The latest 3D electromagnetic inversion system, ModEM, was employed to invert the AMT dataset. A high resolution resistivity model was recovered by using a nonlinear conjugate-gradient scheme. The Akebasitao intrusions, the most pronounced resistive features in the preferred model, are clearly imaged from shallow depths to more than 10 km, with some conductive zones and spots in and surrounding it. The shape of the Akebasitao pluton is asymmetrical mushroom-like, suggesting an extension stress environment at its forming age (the Late Carboniferous to the Early Permian) in Western Junggar. Our explanation is consistent with the subducting spreading ridge tectonic model for Western Junggar in the Late Carboniferous, with a paleo-Asia ocean plate subducting northwestward beneath Karamay arc, as former studies proposed. The upwelling channel of the Akebasitao pluton seems to be located at its center. The Darbut Fault has been clearly imaged as a subvertical narrow conductive zone extending from the surface to 3-5 km or deeper. The most prominent conductors are two elongate zones in depths deeper than 500 m. Other conductive zones can also be identified surrounding the boundaries of the Akebasitao pluton, which can be interpreted as pyrometasomatic metamorphism relating to magmatic activities. Because the ophiolitic rocks outcrop as some lens along these two boundaries, and as previous studies showed that the serpentine metamorphosed from peridotite with well-connected magnetite possessing high conductivity, the conductors may therefore also represent their sources in depth.

  10. Audio Watermarking Using Lsb With Adjustment Method

    Directory of Open Access Journals (Sweden)

    Ansith.S, Priyanka Udayabhanu

    2013-05-01

    Full Text Available In this paper we are discussing watermarking on audio signals. In this method the recorded audio data is first sampled using a sampling frequency of 22050 Hz. Then the watermark message is watermarked into the sampled data of the audio signal. In this method the adjustment is done to increase the accuracy of the watermarked signal. Finally we extract the message from the audio data.

  11. Authentication of digital audio recording based on power system frequency%基于电网频率的数字录音真伪鉴别研究

    Institute of Scientific and Technical Information of China (English)

    刘育明; 姚陈果; 孙才新; 袁智勇; Liu Yilu

    2013-01-01

    采用电网频率鉴别数字录音真伪是司法科学领域近年来的研究热点.在北美电网频率监测系统(FNET)的基础上,构建用于数字录音信号真伪鉴别的专用标准电网频率数据库,提出一种完整的真伪鉴别算法.采用数字滤波技术和短时傅里叶变换估计蕴含在数字信号中的电网频率分量,通过补零操作及三点样条插值提高频率估计的准确度;首次考虑数字采集系统振荡器对频率数据库匹配的影响,提出一种迭代式振荡器误差校正方法;分析短时傅里叶变换的参数选择,如窗函数、补零系数对频率估计准确度的影响,最后实现了该套数字录音真伪鉴别算法.在实测数字语音信号上的实验结果表明,该方法能有效地鉴别数字信号的真伪并确定其原创时间.%Power system frequency has recently been explored for digital audio authentication in the field of forensic science.This paper firstly establishes an ad hoc power system frequency reference database according to the North American frequency monitoring system (FNET),and then proposes a thorough algorithm for the authentication problem.Digital filtering is incorporated with short-time Fourier transform (STFT) to estimate the coarse frequency embedded in a questioned digital audio recording,then zero-padding operation and three-point spline interpolation are adopted to improve the accuracy of frequency estimation.An iterative method for oscillator error correction is put forward for the first time considering the effect of digital acquisition system oscillator on frequency database matching.Moreover,the selection of the STFT parameters,such as window functions and zero-padding coefficient,are also discussed.Finally,the digital audio recording authentication algorithm is implemented;the experiment result for real tested digital audio recording signal indicates that the proposed algorithm can effectively realize digital audio recording signal

  12. The Audio Frequency Conductance Study of Some Metal Succinate Salts in Aqueous Medium at Different Temperatures (Part I: Magnesium, Manganese (II, Barium and Copper Succinates

    Directory of Open Access Journals (Sweden)

    Kosrat N. Kaka

    2013-01-01

    Full Text Available The audio electrical conductances of aqueous solutions of magnesium, manganese II, barium, and copper succinates have been measured at various temperatures in the range of 298.15 K to 313.15 K, using an audio frequency conductance bridge. The evaluation of conductance data was carried out by minimisation technique using the theoretical equations of the complete and modified forms of Pitts (P and Fuoss-Hsia (F-H, each a three-parameter equation, association constant (KA, molar conductance (Λm, and distance parameter (a. Quantitative results showed that these salts do not behave as “strong” electrolytes, and that their dissociations are far from complete. The abnormally low conductances of these electrolytes are not due to the presence of electrically neutral molecules but to the ion-pair formation. The Walden product values, as well as the standard thermodynamics functions (ΔH∘, ΔG∘, ΔS∘ for the association reaction at the four temperatures studied, have been evaluated.

  13. AC-3 audio coder

    Science.gov (United States)

    Todd, Craig

    1995-12-01

    AC-3 is a system for coding up to 5.1 channels of audio into a low bit-rate data stream. High quality may be obtained with compression ratios approaching 12-1 for multichannel audio programs. The high compression ratio is achieved by methods which do not increase decoder memory, and thus cost. The methods employed include: the transmission of a high frequency resolution spectral envelope; and a novel forward/backward adaptive bit allocation algorithm. In order to satisfy practical requirements of an emissions coder, the AC-3 syntax includes a number of features useful to broadcasters and consumers. These features include: loudness uniformity between programs; dynamic range control; and broadcaster control of downmix coefficients. The AC-3 coder has been formally selected for inclusion of the U.S. HDTV broadcast standard, and has been informally selected for several additional applications.

  14. A Forced Vibration Non-Resonant Method for the Determination of Complex Modulus in the Audio Frequency Range

    Science.gov (United States)

    1992-01-01

    water -borne sound when used in anechoic coatings1 . The frequency and temperature dependence of the viscoelastic properties of polymers is essential...strain on the sample, 8: E t11=tan 8- E’ - tanq ) [8] In this case the measured phase angle, p, is identical with 8, the angle between stress and strain

  15. 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.

  16. 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 ...

  17. Audio Indexing for Efficiency

    Science.gov (United States)

    Rahnlom, Harold F.; Pedrick, Lillian

    1978-01-01

    This article describes Zimdex, an audio indexing system developed to solve the problem of indexing audio materials for individual instruction in the content area of the mathematics of life insurance. (Author)

  18. Video salient event classification using audio features

    Science.gov (United States)

    Corchs, Silvia; Ciocca, Gianluigi; Fiori, Massimiliano; Gasparini, Francesca

    2014-03-01

    The aim of this work is to detect the events in video sequences that are salient with respect to the audio signal. In particular, we focus on the audio analysis of a video, with the goal of finding which are the significant features to detect audio-salient events. In our work we have extracted the audio tracks from videos of different sport events. For each video, we have manually labeled the salient audio-events using the binary markings. On each frame, features in both time and frequency domains have been considered. These features have been used to train different classifiers: Classification and Regression Trees, Support Vector Machine, and k-Nearest Neighbor. The classification performances are reported in terms of confusion matrices.

  19. 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.

  20. 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

  1. 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...

  2. Quantization of wavelet packet audio coding

    Institute of Scientific and Technical Information of China (English)

    Tan Jianguo; Zhang Wenjun; Liu Peilin

    2006-01-01

    The method of quantization noise control of audio coding in the wavelet domain is proposed. Using the inverse Discrete Fourier Transform (DFT), it converts the masking threshold coming from MPEG psycho-acoustic model in the frequency domain to the signal in the time domain; the Discrete Wavelet Packet Transform (DWPT) is performed; the energy in each subband is regarded as the maximum allowed quantization noise energy. The experimental result shows that the proposed method can attain the nearly transparent audio quality below 64kbps for the most testing audio signals.

  3. 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.

  4. Audio Papers - A Manifesto

    DEFF Research Database (Denmark)

    Krogh Groth, Sanne; Samson, Kristine

    2016-01-01

    Audio papers resemble the regular essay or the academic text in that they deal with a certain topic of interest, but presented in the form of an audio production. The audio paper is an extension of the written paper through its specific use of media, a sonic awareness of aesthetics and materiality......, and creative approach towards communication. The audio paper is a performative format working together with an affective and elaborate understanding of language. It is an experiment embracing intellectual arguments and creative work, papers and performances, written scholarship and sonic aesthetics....

  5. Digital Audio Legal Recorder

    Data.gov (United States)

    Department of Transportation — The Digital Audio Legal Recorder (DALR) provides the legal recording capability between air traffic controllers, pilots and ground-based air traffic control TRACONs...

  6. Robust audio hashing for audio authentication watermarking

    Science.gov (United States)

    Zmudzinski, Sascha; Steinebach, Martin

    2008-02-01

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

  7. The Study of Audio Watermarking

    Institute of Scientific and Technical Information of China (English)

    王景; 唐晟

    2011-01-01

    This paper mainly introduced the basic knowledge of the digital watermarking and digital audio watermarking, including the definition of digital watermarking and digital audio watermarking, the embedding algorithm of digital audio watermarking and the com

  8. Structure Learning in Audio

    DEFF Research Database (Denmark)

    Nielsen, Andreas Brinch

    By having information about the setting a user is in, a computer is able to make decisions proactively to facilitate tasks for the user. Two approaches are taken in this thesis to achieve more information about an audio environment. One approach is that of classifying audio, and a new approach...

  9. 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.

  10. 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.

  11. 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.

  12. 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.

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

    DEFF Research Database (Denmark)

    Knott, Arnold; Ploug, Rasmus Overgaard

    2012-01-01

    The varying switching frequencies of self oscillating switch mode audio amplifiers have been known to cause interchannel intermodulation disturbances in multi channel configurations. This crosstalk phenomenon has a negative impact on the audio performance. The goal of this paper is to present a m...

  14. 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.

  15. Fall Detection Using Smartphone Audio Features.

    Science.gov (United States)

    Cheffena, Michael

    2016-07-01

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

  16. 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.

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

    OpenAIRE

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

    2014-01-01

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

  18. Embedded Audio Without Beeps

    DEFF Research Database (Denmark)

    Overholt, Daniel; Møbius, Nikolaj Friis

    2014-01-01

    software environments for audio processing) via innovative interfaces that send real-time inputs to such software running on a laptop, mobile device, or small Linux board (e.g., Raspberry Pi or Beagleboard). Basic hardware will be provided, but participants are also encouraged to bring related equipment...

  19. Editing Audio with Audacity

    Directory of Open Access Journals (Sweden)

    Brandon Walsh

    2016-08-01

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

  20. Multipurpose audio watermarking algorithm

    Institute of Scientific and Technical Information of China (English)

    Ning CHEN; Jie ZHU

    2008-01-01

    To make audio watermarking accomplish both copyright protection and content authentication with localization, a novel multipurpose audio watermarking scheme is proposed in this paper. The zero-watermarking idea is introduced into the design of robust watermarking algorithm to ensure the transparency and to avoid the interference between the robust watermark and the semi-fragile watermark. The property of natural audio that the VQ indices of DWT-DCT coefficients among neighboring frames tend to be very similar is utilized to extract essential feature from the host audio, which is then used for watermark extraction. And, the chaotic mapping based semi-fragile watermark is embedded in the detail wavelet coefficients based on the instantaneous mixing model of the independent component analysis (ICA) system. Both the robust and semi-fragile watermarks can be extracted blindly and the semi-fragile watermarking algorithm can localize the tampering accurately. Simulation results demonstrate the effectiveness of our algorithm in terms of transparency, security, robustness and tampering localization ability.

  1. 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…

  2. 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 ...

  3. 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

  4. A Dither Modulation Audio Watermarking Algorithm Based on HAS

    Directory of Open Access Journals (Sweden)

    Yi-bo Huang

    2012-11-01

    Full Text Available In this study, we propose a dither modulation audio watermarking algorithm based on human auditory system which applied the theory of dither modulation. The algorithm made the two-value image watermarking to one-dimensional digital sequence firstly and used the Fibonacci to transform one-dimensional digital sequence. Then divide the audio into audio data segment and made discrete wavelet transform with audio data segment, every segment can adaptive choose quantization step. Finally put low frequency coefficients transformed embedding the watermarking which applied the dither modulation. When extract the watermark with no original audio, they realized blind extraction. The experimental results show that this algorithm has preferable robustness to against the attack from noise addition, compression, low pass filtering and re-sampling.

  5. Research on Application of Natural Source Audio Frequency Magnetotellurics in Structure Exploration%天然源大地电磁法在构造勘察中的应用研究

    Institute of Scientific and Technical Information of China (English)

    祁晓雨; 许广春; 李志华

    2011-01-01

    研究目的:近年来,随着铁路的建设,铁路线位深入到丘陵山区的情况越来越多,在铁路断层构造勘察中,传统物探方法的局限性越来越明显.针对断层构造勘察,对天然源(音频)大地电磁法(AMT)的处理和解释等方面进行深入的研究,目的是提高物探对断层构造勘察的精度及使断层构造勘察基本不受限制.研究结论:(1)在铁路断层构造勘查中,天然源(音频)大地电磁法(AMT)与传统物探方法相比,具有受地形影响小,快捷,勘探深度大,成本低等优势;(2)自主研发的数据处理方法,将天然源(音频)大地电磁法(AMT)与人工源(可控源音频)大地电磁法(CSAMT)使用同一平台进行数据处理,解决了以往存在反演过于平滑,异常均一化的缺陷,断层构造勘察效果更加明显,查明了断层构造分布格局;(3)为对铁路选线提供了有力的依据,对类似情况的铁路断层构造勘察具有很强的指导意义.%Research purposes; With the rapid railway construction, more and more railway lines pass through the hill and mountain areas in recent years, the limitations of the conventional geophysical methods are more and more obvious when they are used for the fault structure exploration for railway. The research is done on the data processing and interpretation of the natural source audio frequency magnetollurics for the fault structure exploration for the purpose of improving the accuracy of fault structure survey and basically having no limitation in the faut structure survey. Research conclusions; In the fault structure exploration for railway, compared with the conventional geophysical methods, the natural source audio frequency magnetotellurics has the features of the rapid survey with less topographical limitation, big survey depth and low cost. With the self - developed data processing methods, the same platform can be used for processing the data of the natural source audio frequency magnetollurics

  6. 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.

  7. Transparency benchmarking on audio watermarks and steganography

    Science.gov (United States)

    Kraetzer, Christian; Dittmann, Jana; Lang, Andreas

    2006-02-01

    The evaluation of transparency plays an important role in the context of watermarking and steganography algorithms. This paper introduces a general definition of the term transparency in the context of steganography, digital watermarking and attack based evaluation of digital watermarking algorithms. For this purpose the term transparency is first considered individually for each of the three application fields (steganography, digital watermarking and watermarking algorithm evaluation). From the three results a general definition for the overall context is derived in a second step. The relevance and applicability of the definition given is evaluated in practise using existing audio watermarking and steganography algorithms (which work in time, frequency and wavelet domain) as well as an attack based evaluation suite for audio watermarking benchmarking - StirMark for Audio (SMBA). For this purpose selected attacks from the SMBA suite are modified by adding transparency enhancing measures using a psychoacoustic model. The transparency and robustness of the evaluated audio watermarking algorithms by using the original and modifid attacks are compared. The results of this paper show hat transparency benchmarking will lead to new information regarding the algorithms under observation and their usage. This information can result in concrete recommendations for modification, like the ones resulting from the tests performed here.

  8. 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.

  9. Portable Audio Design

    DEFF Research Database (Denmark)

    Groth, Sanne Krogh

    2014-01-01

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

  10. 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

  11. 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

  12. 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)

  13. Indirectly exploit buried deposits of rich iron by audio-frequency magnetotelluric method%采用音频大地电磁法间接探测深埋富集铁矿床

    Institute of Scientific and Technical Information of China (English)

    席振铢; 朱伟国; 张道军; 张良六; 冯万杰; 邓志刚

    2012-01-01

    由于深埋富集铁矿矿体的埋深超过500 m,直接运用音频大地电磁法(AMT法)圈定深埋富集铁矿矿体异常的纵向分辨率难以实现,为此,根据矿床成因、成矿环境以及矿体赋存空间特征等地质因素,建立地质模型,在地质模型的基础上构建地电模型;在地电模型的基础上,应用AMT法达到间接找矿的目的.綦江式铁矿和宁乡式铁矿的勘查实践结果表明,应用AMT法能够确定找矿标志、含矿层次和赋矿构造等,从而间接实现探测深埋富集铁矿床.%When the enrichment iron ore is buried more than 500 m, it is difficult to achieve the longitudinal resolution of iron ore abnormalities by using audio-frequency magnetotelluric (AMT) method directly. For this reason, based on the geological factors of the ore genesis, metallogenic environment, ore controlling space, and so on, the geological model was made and then the electrical model was set up, which was used to achieve the aim of indirect ore prospection by AMT method. The prospecting tests results in Qijiang-type iron deposit and Ningxiang-type iron deposit prove that the AMT method can deterimine prespecting mark, ore-bearing strata, ore-bearing structure and so on. So, indirectly detect deep-buried rich iron ore.

  14. 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.

  15. 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.

  16. 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.

  17. 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

  18. 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

  19. 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,

  20. 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.

  1. 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.

  2. 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

  3. A novel audio watermarking scheme using multiscale wavelet modulation

    Institute of Scientific and Technical Information of China (English)

    JI Bing; ZHANG De; JI Xiaoyong

    2004-01-01

    A novel audio watermarking scheme to embed robust and inaudible watermarks for the purpose of copyright protection is proposed. The key innovation is to add time-frequency redundancy into watermark signals by multiscale wavelet modulation. In order to maximize the watermarking strength within perceptual constraints, the signals synthesized from different scales are masked using a frequency auditory model, respectively, and then intergrated to form the final watermark signal. The detection structure is built using the redundancy in watermark signals, and the performance is further enhanced by modeling the statistical behaviors of wavelet coefficients as generalized Gaussian distribution. The use of original audio signal is not required in watermark detection. The experimental results show that our approach can achieve not only good transparency but also satisfying robustness to common audio manipulations.

  4. 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...

  5. DOUBLE-BOOST DC-AC CONVERTER WITH SLIDING-MODE CONTROL FOR PORTABLE AUDIO

    DEFF Research Database (Denmark)

    Bolten Maizonave, Gert; Andersen, Michael Andreas E.; Kjærgaard, Claus;

    2009-01-01

    The double-boost topology is studied for operation as a dc-ac converter and single stage audio amplifier. A sliding-mode controller is designed in order to achieve fast enough response for the whole audio frequency range. Symmetric, asymmetric and interleaved operation modes are analyzed....

  6. 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....

  7. 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...

  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 unde...

  9. 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.

  10. 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

  11. 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

  12. Evaluation of Audio Compression Artifacts

    Directory of Open Access Journals (Sweden)

    M. Herrera Martinez

    2007-01-01

    Full Text Available This paper deals with subjective evaluation of audio-coding systems. From this evaluation, it is found that, depending on the type of signal and the algorithm of the audio-coding system, different types of audible errors arise. These errors are called coding artifacts. Although three kinds of artifacts are perceivable in the auditory domain, the author proposes that in the coding domain there is only one common cause for the appearance of the artifact, inefficient tracking of transient-stochastic signals. For this purpose, state-of-the art audio coding systems use a wide range of signal processing techniques, including application of the wavelet transform, which is described here. 

  13. An Efficient Audio Classification Approach Based on Support Vector Machines

    Directory of Open Access Journals (Sweden)

    Lhoucine Bahatti

    2016-05-01

    Full Text Available In order to achieve an audio classification aimed to identify the composer, the use of adequate and relevant features is important to improve performance especially when the classification algorithm is based on support vector machines. As opposed to conventional approaches that often use timbral features based on a time-frequency representation of the musical signal using constant window, this paper deals with a new audio classification method which improves the features extraction according the Constant Q Transform (CQT approach and includes original audio features related to the musical context in which the notes appear. The enhancement done by this work is also lay on the proposal of an optimal features selection procedure which combines filter and wrapper strategies. Experimental results show the accuracy and efficiency of the adopted approach in the binary classification as well as in the multi-class classification.

  14. 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.

  15. Amplificador de audio Clase D

    OpenAIRE

    2012-01-01

    El presente proyecto lleva a cabo el desarrollo de un amplificador de audio tipo D basado en dos tipos de modulación, modulación PWM y modulación Sigma-Delta ambos con puente inversor en H. Tanto el modulador PWM como el modulador Sigma-Delta se desarrollaran mediante circuitos digitales implementados en una FPGA. La señal de audio de entrada se digitalizará mediante un convertidor analógico–digital (ADC) que también estará controlado mediante una circuitería digital implementada en la misma ...

  16. 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

  17. 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...

  18. Audio-magnetotelluric methods in reconnaissance geothermal exploration

    Science.gov (United States)

    Hoover, D.B.; Long, C.L.

    1976-01-01

    An audio-magnetotelluric (AMT) system has been developed by the U.S. Geological Survey for low-cost reconnaissance exploration of geothermal regions. This is an electromagnetic sounding technique in which the scalar or Cagniard resistivity is computed at 12 frequencies logarithmically spaced from 7.5 to 18 600 Hz. Our system uses natural source fields except at the two upper frequencies of 10 200

  19. 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

  20. 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…

  1. 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...

  2. The Audio-Visual Man.

    Science.gov (United States)

    Babin, Pierre, Ed.

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

  3. 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…

  4. [Audio-visual aids and tropical medicine].

    Science.gov (United States)

    Morand, J J

    1989-01-01

    The author presents a list of the audio-visual productions about Tropical Medicine, as well as of their main characteristics. He thinks that the audio-visual educational productions are often dissociated from their promotion; therefore, he invites the future creator to forward his work to the Audio-Visual Health Committee.

  5. Spatial audio quality perception (part 1)

    DEFF Research Database (Denmark)

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

    2015-01-01

    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 resu...

  6. 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......There are many situations in which multiple audio programs are replayed over loudspeakers in the same acoustic environment, allowing listeners to focus on their desired target program. Where this situation is deliberately created and the different program items are centrally controlled, each...... separation, and frequency content of the interferer. The model was found to predict accurately for the training and validation datasets....

  7. Optimization of audio - ultrasonic plasma system parameters

    Science.gov (United States)

    Haleem, N. A.; Abdelrahman, M. M.; Ragheb, M. S.

    2016-10-01

    The present plasma is a special glow plasma type generated by an audio ultrasonic discharge voltage. A definite discharge frequency using a gas at a narrow band pressure creates and stabilizes this plasma type. The plasma cell is a self-extracted ion beam; it is featured with its high output intensity and its small size. The influence of the plasma column length on the output beam due to the variation of both the audio discharge frequency and the power applied to the plasma electrodes is investigated. In consequence, the aim of the present work is to put in evidence the parameters that influence the self-extracted collected ion beam and to optimize the conditions that enhance the collected ion beam. The experimental parameters studied are the nitrogen gas, the applied frequency from 10 to 100 kHz, the plasma length that varies from 8 to 14 cm, at a gas pressure of ≈ 0.25 Torr and finally the discharge power from 50 to 500 Watt. A sheet of polyethylene of 5 micrometer covers the collector electrode in order to confirm how much ions from the beam can go through the polymer and reach the collector. To diagnose the occurring events of the beam on the collector, the polymer used is analyzed by means of the FTIR and the XRF techniques. Optimization of the plasma cell parameters succeeded to enhance and to identify the parameters that influence the output ion beam and proved that its particles attaining the collector are multi-energetic.

  8. Possible technical solutions to reduce energy consumption in audio products

    Energy Technology Data Exchange (ETDEWEB)

    Nielsen, K.; Andersen, M.A.E.

    1999-07-01

    In common audio products nearly all the supplied power is dissipated as heat. The major consumers are with almost no exception the power supply and the audio amplifier. This paper is divided in two parts, concentrating on typical efficiency measures for the concepts of today and the possibly technical solutions, by which the overall efficiency can be considerably improved in the future. Traditional power supplies are made using a transformer operating on the mains frequency followed by a linear regulator. These are bulky and the efficiency is only around 40%. Using high frequency switch mode power supplies the size of the power supply can be reduced and the efficiency can be increased to 80-90%. Construction of optimal amplifiers in regard to total energy consumption over life time, can only be accomplished by considering both the general volume control distribution, and the general spectral amplitude distribution of audio signals. The traditional efficiency measure specified at the maximum efficiency level says only very little about the real energy consumption of the audio amplifier. As an example, the theoretical efficiency for at traditional class B amplifier is 78%. Using a new efficiency measure defined on the basis of the approximate volume control distribution, an 50W amplifier example shows an overall efficiency of only 1%. In the paper possible solutions and guidelines to increase the real amplifier efficiency are given. (au)

  9. Worldwide survey of direct-to-listener digital audio delivery systems development since WARC-1992

    Science.gov (United States)

    Messer, Dion D.

    Each country was allocated frequency band(s) for direct-to-listener digital audio broadcasting at WARC-92. These allocations were near 1500, 2300, and 2600 MHz. In addition, some countries are encouraging the development of digital audio broadcasting services for terrestrial delivery only in the VHF bands (at frequencies from roughly 50 to 300 MHz) and in the medium-wave broadcasting band (AM band) (from roughly 0.5 to 1.7 MHz). The development activity increase was explosive. Current development, as of February 1993, as it is known to the author is summarized. The information given includes the following characteristics, as appropriate, for each planned system: coverage areas, audio quality, number of audio channels, delivery via satellite/terrestrial or both, carrier frequency bands, modulation methods, source coding, and channel coding. Most proponents claim that they will be operational in 3 or 4 years.

  10. Design of an Audio Interface for Patmos

    OpenAIRE

    Ausin, Daniel Sanz; Goerge, Fabian

    2017-01-01

    This paper describes the design and implementation of an audio interface for the Patmos processor, which runs on an Altera DE2-115 FPGA board. This board has an audio codec included, the WM8731. The interface described in this work allows to receive and send audio from and to the WM8731, and to synthesize, store or manipulate audio signals writing C programs for Patmos. The audio interface described in this paper is intended to be used with the Patmos processor. Patmos is an open source RISC ...

  11. Two-dimensional audio watermark for MPEG AAC audio

    Science.gov (United States)

    Tachibana, Ryuki

    2004-06-01

    Since digital music is often stored in a compressed file, it is desirable that an audio watermarking method in a content management system handles compressed files. Using an audio watermarking method that directly manipulates compressed files makes it unnecessary to decompress the files before embedding or detection, so more files can be processed per unit time. However, it is difficult to detect a watermark in a compressed file that has been compressed after the file was watermarked. This paper proposes an MPEG Advanced Audio Coding (AAC) bitstream watermarking method using a two-dimensional pseudo-random array. Detection is done by correlating the absolute values of the recovered MDCT coefficients and the pseudo-random array. Since the embedding algorithm uses the same pseudo-random values for two adjacent overlapping frames and the detection algorithm selects the better frame in the two by comparing detected watermark strengths, it is possible to detect a watermark from a compressed file that was compressed after the watermark was embedded in the original uncompressed file. Though the watermark is not detected as clearly in this case, the watermark can still be detected even when the watermark was embedded in a compressed file and the file was then decompressed, trimmed, and compressed again.

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

    Directory of Open Access Journals (Sweden)

    Nitin Arora

    2010-06-01

    Full Text Available 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 transparent manner that leaves the archival audio file unaltered. Doing so has alleviated many of the tensions of cumbersome workflows, complicated documentation, preservation concerns, and reliance on expensive closed-source GUI audio applications.

  13. 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...... are organized in topical sections on multimodal integration, tactile and sonic explorations, walking and navigation interfaces, prototype design and evaluation, and gestures and emotions.......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...

  14. Comparison of Linear Prediction Models for Audio Signals

    Directory of Open Access Journals (Sweden)

    van Waterschoot Toon

    2008-01-01

    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.

  15. 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.

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

    Directory of Open Access Journals (Sweden)

    Jussi Kuutti

    2014-02-01

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

  17. 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.

  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.

  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. Review of AVS Audio Coding Standard

    Institute of Scientific and Technical Information of China (English)

    ZHANG Tao; ZHANG Caixia; ZHAO Xin

    2016-01-01

    Audio Video Coding Standard (AVS) is a second⁃generation source coding standard and the first standard for audio and video coding in China with independent intellectual property rights. Its performance has reached the international standard. Its coding efficiency is 2 to 3 times greater than that of MPEG⁃2. This technical solution is more simple, and it can greatly save channel resource. After more than ten years ’develop⁃ment, AVS has achieved great success. The latest version of the AVS audio coding standard is ongoing and mainly aims at the increasing demand for low bitrate and high quality audio services. The paper reviews the history and recent develop⁃ment of AVS audio coding standard in terms of basic fea⁃tures, key techniques and performance. Finally, the future de⁃velopment of AVS audio coding standard is discussed.

  2. Digital audio watermarking fundamentals, techniques and challenges

    CERN Document Server

    Xiang, Yong; Yan, Bin

    2017-01-01

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

  3. Distortion Estimation in Compressed Music Using Only Audio Fingerprints

    NARCIS (Netherlands)

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

    2008-01-01

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

  4. 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

  5. Inexpensive Audio Activities: Earbud-based Sound Experiments

    Science.gov (United States)

    Allen, Joshua; Boucher, Alex; Meggison, Dean; Hruby, Kate; Vesenka, James

    2016-11-01

    Inexpensive alternatives to a number of classic introductory physics sound laboratories are presented including interference phenomena, resonance conditions, and frequency shifts. These can be created using earbuds, economical supplies such as Giant Pixie Stix® wrappers, and free software available for PCs and mobile devices. We describe two interference laboratories (beat frequency and two-speaker interference) and two resonance laboratories (quarter- and half-wavelength). Lastly, a Doppler laboratory using rotating earbuds is explained. The audio signal captured by all experiments is analyzed on free spectral analysis software and many of the experiments incorporate the unifying theme of measuring the speed of sound in air.

  6. Exploiting Acoustic Similarity of Propagating Paths for Audio Signal Separation

    Directory of Open Access Journals (Sweden)

    Yin Bin

    2003-01-01

    Full Text Available Blind signal separation can easily find its position in audio applications where mutually independent sources need to be separated from their microphone mixtures while both room acoustics and sources are unknown. However, the conventional separation algorithms can hardly be implemented in real time due to the high computational complexity. The computational load is mainly caused by either direct or indirect estimation of thousands of acoustic parameters. Aiming at the complexity reduction, in this paper, the acoustic paths are investigated through an acoustic similarity index (ASI. Then a new mixing model is proposed. With closely spaced microphones (5–10 cm apart, the model relieves the computational load of the separation algorithm by reducing the number and length of the filters to be adjusted. To cope with real situations, a blind audio signal separation algorithm (BLASS is developed on the proposed model. BLASS only uses the second-order statistics (SOS and performs efficiently in frequency domain.

  7. 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.    

  8. Presence and the utility of audio spatialization

    DEFF Research Database (Denmark)

    Bormann, Karsten

    2005-01-01

    or not, while the presence questionnaire used by Slater and coworkers (see Tromp et al., 1998) was more sensitive to whether audio was fully spatialized or not. Finally, having the sound source active positively impacts the assessment of the audio while negatively impacting subjects' assessment...

  9. 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...

  10. 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)

  11. Digital Advances in Contemporary Audio Production.

    Science.gov (United States)

    Shields, Steven O.

    Noting that a revolution in sonic high fidelity occurred during the 1980s as digital-based audio production methods began to replace traditional analog modes, this paper offers both an overview of digital audio theory and descriptions of some of the related digital production technologies that have begun to emerge from the mating of the computer…

  12. Construction of 3-D Audio Systems: Background, Research, and General Requirements

    Science.gov (United States)

    2008-10-01

    frequency content to a sound was to brighten the timbre , it should be possible to add high frequency content using the same temporal envelope. Patterson...Hammershoi, D. (1996). Binaural technique: Do we need individual recordings? Journal of the Audio Engineering Society., 44 (6), 451- 469. Musicant

  13. 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.

  14. Detecting double compression of audio signal

    Science.gov (United States)

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

    2010-01-01

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

  15. The HDTV digital audio matrix

    Science.gov (United States)

    Mason, A. J.

    Multichannel sound systems are being studied as part of the Eureka 95 and Radio-communication Bureau TG10-1 investigations into high definition television. One emerging sound system has five channels; three at the front and two at the back. This raises some compatibility issues. The listener might have only, say, two loudspeakers or the material to be broadcast may have fewer than five channels. The problem is how best to produce a set of signals to be broadcast, which is suitable for all listeners, from those that are available. To investigate this area, a device has been designed and built which has six input channels and six output channels. Each output signal is a linear combination of the input signals. The inputs and outputs are in AES/EBU digital audio format using BBC-designed AESIC chips. The matrix operation, to produce the six outputs from the six inputs, is performed by a Motorola DSP56001. The user interface and 'housekeeping' is managed by a T222 transputer. The operator of the matrix uses a VDU to enter sets of coefficients and a rotary switch to select which set to use. A set of analog controls is also available and is used to control operations other than the simple compatibility matrixing. The matrix has been very useful for simple tasks: mixing a stereo signal into mono, creating a stereo signal from a mono signal, applying a fixed gain or attenuation to a signal, exchanging the A and B channels of an AES/EBU bitstream, and so on. These are readily achieved using simple sets of coefficients. Additions to the user interface software have led to several more sophisticated applications which still consist of a matrix operation. Different multichannel panning laws have been evaluated. The analog controls adjust the panning; the audio signals are processed digitally using a matrix operation. A digital SoundField microphone decoder has also been implemented. digital audio matrix is such that it can be applied to a wide variety of signal processing

  16. 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.

  17. DEVELOPMENT OF HYDROLOGICAL CABLEWAY SAND CONTROL MEASUREMENT INSTRUMENT BASED ON THE DOUBLE AUDIO FREQUENCY SIGNAL CONTROL%基于双音频信号控制的水文缆道测沙自控仪的研制

    Institute of Scientific and Technical Information of China (English)

    康修洪; 李先哲; 肖英; 朱兵

    2015-01-01

    According to the automatic cable measurement instruments widely used in hydrologic station, which has some problems such as cannot measurement and sampling sediment normally, we carried out the development of hydrological cableway sand control measurement instrument. The automatic control instrument will organically combine double tube double steady water sample collected valves and lead fish, reasonable decorate into the drainage pipeline and communication lines, the horizontal and vertical of hydrological cableway operate accurate positions, with double audio signal to transmit reliable control subsea sampler valve switch, acquisition section of each location, different depth of water samples, and send back the valve switch status signal in time. Practice has proved that the adoption of double tube double steady water samples collected valves can solve the problem of inlet pipe alluvial sediment under closed pressure, at the same time valve no need to seal. It also implements the hydrological monitoring data two-way transmission control of the bank to the water and the water to the shore.%针对水文站中普遍使用的自动缆道测流仪器,均存在进水阀密封、信号控制、与缆道测流系统整合等诸多问题,研制了一种基于嵌入式的水文缆道测沙自控仪。该自控仪将双管双稳态水样采集阀门与铅鱼有机结合,合理布置进排水管线和通讯线路,水文缆道的水平和垂直运行准确定位,采用双音频信号传输可靠的控制水下采样器阀门的开关,测取断面各位置,不同水深的水样,并及时发回阀门开关状况信号。实践证明,由于采用了双管双稳态水样采集阀门解决了关闭状态下进水管淤积泥沙问题,同时阀门不必进行全密封,还实现了水文监测数据岸上至水中和水中至岸上的双向控制传输,研制的仪器获得成功应用。

  18. 基于情绪对自主神经功能调节作用的恐怖音频对心率和心率变异性的影响研究%Influence of Frightening Audio Frequency based on Regulation of ANS by Emotion on Heart Rate and Heart Rate Variability

    Institute of Scientific and Technical Information of China (English)

    王济舟; 陈俊琦; 黄焕琳; 岑曦; 李雨捷; 潘洪权; 黄泳; Gustav Wik

    2013-01-01

    Objective To investigate the impact of frightening audio frequency on heart rate and its variability. Methods Seventy healthy volunteers were recruited and divided into two groups with 35 in each according to random number table. Volunteers in group 1 received neutral audio stimulation and those in group 2 received frightening audio stimulation. The heart rate and its variability of each subject were recorded during the stimulation through Actiheart device. Results Compared with group 1, in group 2 the root mean square of successive differences ( RMSSD ) was lower [ ( 37. 5 ± 14. 7 ) ms vs. ( 53. 8 ± 30. 3 ) ms ] , the heart rate was higher [ ( 70. 2 ± 8. 1 ) /min vs. ( 65. 2 ± 8. 2 ) /min ], and the ratio of low frequency over high frequency ( LF/HF ) [ ( 1. 52 ±0. 96 ) vs. ( 1. 03 ±0. 58 ) ] increased , the differences were statistically significant ( P <0. 05 ). Conclusion The fright emotion could result in the inhibition of vagal activity and produce a sympathetic dominance, and thus reduce RMSSD, increase heart rate and LF/HF.%目的 探讨恐怖音频对心率和心率变异性的影响.方法 招募健康试验者70例,根据随机数字表随机分为恐怖音频组和中性音频组,各35例,分别给予恐怖音频和中性音频刺激,并同时用Actiheart心电记录仪记录音频刺激时的心率和心率变异性指标的变化.结果 与接受中性音频刺激相比,恐怖音频组相邻NN间期差的均方根(RMSSD)降低[(37.5±14.7) ms vs.(53.8±30.3) ms],心率[(70.2±8.1) 次/min vs.(65.2±8.2) 次/min]和低频功率与高频功率之比(LF/HF)[(1.52±0.96)vs.(1.03±0.58)]升高,差异均有统计学意义(P<0.05).结论 在交感-迷走平衡中,恐惧情绪可以导致迷走神经兴奋性受到抑制,交感神经功能占主导地位,从而降低RMSSD以及升高心率和LF/HF.

  19. 音频电20%碘化钾透入为主治疗慢性软组织损伤的临床观察%Clinical Observation of Treatment on Chronic Soft Tissue Injury with Audio Frequency Electrotherapy of 20% Potassium Iodide

    Institute of Scientific and Technical Information of China (English)

    黄烈弥; 杨秋萍

    2011-01-01

    目的:研究以音频电20%碘化钾透入为主治疗慢性软组织损伤的临床疗效.方法:将确诊的156例慢性软组织损伤患者随机分为两组,均在超短波治疗基础上,治疗组(84例)加用20%碘化钾音频电透入;对照组(72例)加用20%碘化钾直流电导入,在治疗前及治疗3周后均采用健康调查简表SF-36进行评定,所有数据用SPSS16.0进行统计学处理.结果:2组治疗后与治疗前比较差异有统计学意义(P〈0.05),治疗后两组间比较差异有统计学意义(P〈0.01).结论:采用音频电20%碘化钾透入为主治疗慢性软组织损伤临床效果显著,操作简单,能有效减轻患者痛苦,提高生活质量,值得临床推广.%Objective: To observe the clinical effect of treatment on chronic tissue injury with audio frequency electrotherapy of 20% potassium iodide.Methods: Randomly divided the confirmed 156 cases of chronic soft tissue injury into two groups.On the basis of ultra-short wave therapy,the treatment group(84 cases) added audio frequency electrotherapy of 20% potassium iodide,and the control group(72 cases) added direct current electrotherapy of 20% potassium iodide.Before and three weeks after the therapy,evaluated the physical health and mental health of patients with SF-36 healthy measuring scale,all the datas were dealt with SPSS16.0.Results: In both groups,there was a significant difference before and after the therapy(P 0.05.And after the therapy,there was a significant difference between two groups(P 0.01).Conclusion: The clinical effect of treat-ment on chronic soft tissue injury with audio frequency injury of 20% potassium iodide is obvious,and the operation is simplified.The therapy can relieve the pain of patients and improve their living quality,so it deserves the clinical expansion.

  20. Implementing Audio-CASI on Windows' Platforms.

    Science.gov (United States)

    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 control they provide for manipulating the video presentation of survey questions. This article reports our experiences adapting Audio-CASI to Microsoft Windows 3.1 and Windows 95 platforms. Overall, our Windows-based system provided the desired control over video presentation and afforded other advantages including compatibility with a much wider array of audio devices than our DOS-based Audio-CASI technologies. These advantages came at the cost of increased system requirements --including the need for both more RAM and larger hard disks. While these costs will be an issue for organizations converting large inventories of PCS to Windows Audio-CASI today, this will not be a serious constraint for organizations and individuals with small inventories of machines to upgrade or those purchasing new machines today.

  1. Implementing Audio-CASI on Windows’ Platforms

    Science.gov (United States)

    Cooley, Philip C.; Turner, Charles F.

    2011-01-01

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

  2. 间歇性导尿配合音频电疗、微波治疗脊髓损伤后痉挛性膀胱的效果研究%Research in intermittent urethral catheterization combined with audio frequency electrotherapy and microwave therapy in the treatment of patients with spastic bladder after spinal cord injury

    Institute of Scientific and Technical Information of China (English)

    应燕萍

    2008-01-01

    目的 探讨脊髓损伤(SCI)后痉挛性膀胱患者恢复膀胱功能和自主排尿的方法.方法 对实验组30例患者间歇导尿6次/d,音频电疗、微波治疗各2次/d;对照组30例患者采用传统的留置尿管方法,观察2组患者膀胱功能恢复的情况.结果 实验组患者恢复自行排尿成功率显著高于对照组,差异有统计学意义(P<0.01).实验组膀胱容量增加、残余尿量减少优于对照组(P<0.05).结论 间歇性导尿结合音频电疗、微波进行膀胱功能训练能有效地促进SCI后痉挛性膀胱患者膀胱功能的恢复.%Objective To discuss methods for recovery of bladder function and autonomous urination for patients with spastic bladder after spinal cord injury. Methods Patients in the experimental group(30 cases)received intermittent urethral catheterization 6 times per day,audio frequency electrotherapy and microwave therapy 2 times per day.The control group(30 cases)adopted routine urethral catheterization.The recovery of bladder function of the two groups Was observed.Results The rate of autonomous urination in the experimental group was higher than that of the control group(P<0.01).Increase of bladder capacity and decrease of remnant urine volume in the experimental group was better than that of the control group(P<0.05).Conclusions Intermittent urethral catheterization combined with audio frequency electrotherapy and microwave therapy could effectively promote the recovery of bladder function for patients with spastic bladder after spinal cord injury.

  3. Effect of Six Different Audio Frequencies on Growth of Cowpea ( Vigna unguiculata ) during Seedling Stage%6种不同声频对豇豆苗期生长影响的研究

    Institute of Scientific and Technical Information of China (English)

    姜仕仁; 黄俊

    2011-01-01

    [目的]研究不同类型、频率特征的声波对豇豆苗期生长的影响.[方法]以古典音乐与昆虫鸣声的混合声波(MI)、杜鹃鸣声、蟋蟀鸣声、单一频率的400 Hz声波和多种频率合成的F5,Fn声波进行试验比较,对豇豆苗高、苗重进行测定,采用EXCEL进行统计分析和多重比较.[结果]经6种不同类型和频率声频处理后,豇豆苗期生长状况均显著优于对照组,表明声波能显著促进株高生长,对豇豆苗期助长效果较好的是400Hz、杜鹃鸣声和蟋蟀鸣声,其次是MI,Fn和F5;与对照组相比,杜鹃鸣声和蟋蟀鸣声处理组在生长期间能显著促进苗株增重.[结论]不同类型、频率特征的声波对豇豆苗期生长都有明显的助长作用,但作用效果有所差异.%[ Objective] The aim was to study the effects of acoustic waves with different types and frequency characteristics on the growth of cowpea (Vigna unguiculata) during seedling stage. [ Method] The insect-music mixed sound (MS), cuckoo acoustic song, cricket acoustic song, single 400 Hz frequency acoustic wave, F5 and Fn acoustic waves composed of different frequencies were designed to investigate the effects on height and weight of cowpea seedling, and experimental data were statistically analyzed and multiple-compared by EXCEL. [ Result ] After treatment by six different types and frequencies of acoustic waves, the growth situations of cowpea were better than control. This indicated that acoustic waves could significantly promote height growth of plant. The treatments with good growth promotion effect included 400 Hz frequency acoustic wave, cuckoo acoustic song and cricket acoustic song, followed by MI, Fn and F5. Cuckoo and cricket acoustic song treatment could promote the weight of cowpea seedling during growth stage. [ Conclusion ] Acoustic waves with different types and frequency characteristics had significant growth-promotion effect on growth of cowpea during seedling stage, but

  4. 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.

  5. 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.

  6. Audio Indexing on the Web: a Preliminary Study of Some Audio Descriptors

    OpenAIRE

    Parlangeau-Vallès, Nathalie; Farinas, Jérôme; Fohr, Dominique; Illina, Irina; Magrin-Chagnolleau, Ivan; Mella, Odile; PINQUIER, Julien; Rouas, Jean-Luc; Sénac, Christine

    2003-01-01

    Colloque avec actes et comité de lecture. internationale.; International audience; The "Invisible Web" is composed of documents which can not be currently accessed by Web search engines, because they have a dynamic URL or are not textual, like video or audio documents. For audio documents, one solution is automatic indexing. It consists in finding good descriptors of audio documents which can be used as indexes for archiving and search. This paper presents an overview and recent results of th...

  7. 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.

  8. Virtual Microphones for Multichannel Audio Resynthesis

    Directory of Open Access Journals (Sweden)

    Athanasios Mouchtaris

    2003-09-01

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

  9. 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.

  10. Spatial audio reproduction with primary ambient extraction

    CERN Document Server

    He, JianJun

    2017-01-01

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

  11. Definición de audio

    OpenAIRE

    Montañez, Luis A.; Cabrera, Juan G.

    2015-01-01

    Descripción del significado de Audio como objeto de estudio por distintos autores, y su diferenciación con el significado de Sonido. De esta forma se define Audio como una señal eléctrica con características similares en su forma de onda en comparación a la de una señal sonora, teniendo en cuenta la señal sonora corresponde a presión en u medio físico, mientras que la señal de Audio es una tensión o voltaje definida como señal análoga. En este orden de ideas, el Audio se concibe como una seña...

  12. Post-Production: "Sweeting" the Final Audio.

    Science.gov (United States)

    Beasley, Augie

    1995-01-01

    Knowing how to use audio mixers in the postproduction of student videos is necessary for high-quality sound. Equipment and techniques are described, and the use of background sound, sound effects, and music is described. (AEF)

  13. 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...

  14. Web Audio/Video Streaming Tool

    Science.gov (United States)

    Guruvadoo, Eranna K.

    2003-01-01

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

  15. 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.

  16. Audio description as an accessibility enhancer

    OpenAIRE

    Martins, Cláudia Susana Nunes

    2012-01-01

    Audio description for the blind and visually-impaired has been around since people have described what is seen. Throughout time, it has evolved and developed in different contexts, starting with daily life, moving into the cinema and television, then across other performing arts, museums and galleries, historical sites and public places. Audio description is above all an issue of accessibility and of providing visually-impaired people with the same rights to have access to culture, e...

  17. Watermarking-Based Digital Audio Data Authentication

    Directory of Open Access Journals (Sweden)

    Jana Dittmann

    2003-09-01

    Full Text Available Digital watermarking has become an accepted technology for enabling multimedia protection schemes. While most efforts concentrate on user authentication, recently interest in data authentication to ensure data integrity has been increasing. Existing concepts address mainly image data. Depending on the necessary security level and the sensitivity to detect changes in the media, we differentiate between fragile, semifragile, and content-fragile watermarking approaches for media authentication. Furthermore, invertible watermarking schemes exist while each bit change can be recognized by the watermark which can be extracted and the original data can be reproduced for high-security applications. Later approaches can be extended with cryptographic approaches like digital signatures. As we see from the literature, only few audio approaches exist and the audio domain requires additional strategies for time flow protection and resynchronization. To allow different security levels, we have to identify relevant audio features that can be used to determine content manipulations. Furthermore, in the field of invertible schemes, there are a bunch of publications for image and video data but no approaches for digital audio to ensure data authentication for high-security applications. In this paper, we introduce and evaluate two watermarking algorithms for digital audio data, addressing content integrity protection. In our first approach, we discuss possible features for a content-fragile watermarking scheme to allow several postproduction modifications. The second approach is designed for high-security applications to detect each bit change and reconstruct the original audio by introducing an invertible audio watermarking concept. Based on the invertible audio scheme, we combine digital signature schemes and digital watermarking to provide a public verifiable data authentication and a reproduction of the original, protected with a secret key.

  18. 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

  19. Audio Video Compression Stream Synthesis and Implementation

    Institute of Scientific and Technical Information of China (English)

    徐燕凌; 方向忠; 周源华

    2004-01-01

    Multiplex of digital streams is one of the key technologies in audio video communication, and determines audio-video quality. A design scheme for an MPEG2 compliant digital television system including audio-video encoding and multiplexing was implemented. The principles and elements of system layer stream synthesis were analyzed. The key technologies of video and audio PES packetization were discussed, such as stream structure,scheduling matching, audio-video synchronization, data flow and buffering. DSP and FPGA are combined to construct header information and packet structure. The substitution of traditional RAM or PLD results in high operational efficiency and saves memory space. A scheduling algorithm was introduced for PES coding, using the monitor information of PES buffers. DTS is generated by multiplexer to guarantee synchronization. The system is not only simple but also stable, and maintains synchronization constraints of the standard. It supports both analogy and digital audio-video source input, and provides real-time MPEG2 compliant TS/PS output. It has perfect performance and meets the national broadcasting requirements.

  20. 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

  1. Optimized Audio Classification and Segmentation Algorithm by Using Ensemble Methods

    OpenAIRE

    Saadia Zahid; Fawad Hussain; Muhammad Rashid; Muhammad Haroon Yousaf; Hafiz Adnan Habib

    2015-01-01

    Audio segmentation is a basis for multimedia content analysis which is the most important and widely used application nowadays. An optimized audio classification and segmentation algorithm is presented in this paper that segments a superimposed audio stream on the basis of its content into four main audio types: pure-speech, music, environment sound, and silence. An algorithm is proposed that preserves important audio content and reduces the misclassification rate without using large amount o...

  2. Audio Journal in an ELT Context

    Directory of Open Access Journals (Sweden)

    Neşe Aysin Siyli

    2012-09-01

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

  3. Near-field Localization of Audio

    DEFF Research Database (Denmark)

    Jensen, Jesper Rindom; Christensen, Mads Græsbøll

    2014-01-01

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

  4. The roar of Yasur: Handheld audio recorder monitoring of Vanuatu volcanic vent activity

    Science.gov (United States)

    Lorenz, Ralph D.; Turtle, Elizabeth P.; Howell, Robert; Radebaugh, Jani; Lopes, Rosaly M. C.

    2016-08-01

    We describe how near-field audio recording using a pocket digital sound recorder can usefully document volcanic activity, demonstrating the approach at Yasur, Vanuatu in May 2014. Prominent emissions peak at 263 Hz, interpreted as an organ-pipe mode. High-pass filtering was found to usefully discriminate volcano vent noise from wind noise, and autocorrelation of the high pass acoustic power reveals a prominent peak in exhalation intervals of 2.5, 4 and 8 s, with a number of larger explosive events at 200 s intervals. We suggest that this compact and inexpensive audio instrumentation can usefully supplement other field monitoring such as seismic or infrasound. A simple estimate of acoustic power interpreted with a dipole jet noise model yielded vent velocities too low to be compatible with pyroclast emission, suggesting difficulties with this approach at audio frequencies (perhaps due to acoustic absorption by volcanic gases).

  5. Research Progress on Key Technologies of Audio Forensics%音频取证若干关键技术研究进展

    Institute of Scientific and Technical Information of China (English)

    包永强; 梁瑞宇; 丛韫; 高冲红; 王青云

    2016-01-01

    The latest research progress in audio forensics is introduced with the emphasis on audio authen‐ticity .First ,the history of audio forensics research is reviewed .The classification of audio forensics is discussed .Then ,the framework of audio forensics is designed .Several key technologies of audio foren‐sics are summarized including audio active forensics technology ,audio tamper technology based on electri‐cal network frequency (ENF) ,audio tamper detection technology with different sampling rates and audio tamper detection technology with the same sampling rates under the passive power grid frequency compo‐nents ,the characteristic parameters of recording equipment ,pattern recognition ,situation of database construction ,recording environment identification and so on .Finally ,the prospective of audio forensics technology is presented .%介绍了音频取证领域的最新研究进展、音频真实性的研究状况。对音频取证研究领域的历史进行了回顾,探讨了音频取证的分类,构建了音频取证框架。对音频取证的若干个关键技术进行了总结,包括音频主动取证技术、基于电网频率特征的音频篡改技术、无电网频率成分下的音频篡改检测技术、录音设备的特征参数、模式识别、数据库建设情况以及录音场合识别等。最后对音频取证技术进行了总结和展望。

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

    Institute of Scientific and Technical Information of China (English)

    朱樟明; 刘帘曦; 杨银堂; 雷晗

    2009-01-01

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

  7. 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.

  8. 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.

  9. 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 preprocessing step to select and synchronize recordings before further processing. The two methods perform similarly in classification, but fingerprinting scales better with the number of recordings, while cross-correlation can offer sample resolution synchronization. We propose and investigate the benefits...

  10. 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.

  11. Audio Mining with emphasis on Music Genre Classification

    DEFF Research Database (Denmark)

    Meng, Anders

    2004-01-01

    Audio is an important part of our daily life, basically it increases our impression of the world around us whether this is communication, music, danger detection etc. Currently the field of Audio Mining, which here includes areas of music genre, music recognition / retrieval, playlist generation...... in searching / retrieving audio effectively is needed. Currently, search engines such as e.g. Google, AltaVista etc. do not search into audio files, but uses either the textual information attached to the audio file or the textual information around the audio. Also in the hearing aid industries around...... 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...

  12. 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...

  13. Cross-modal retrieval of scripted speech audio

    Science.gov (United States)

    Owen, Charles B.; Makedon, Fillia

    1997-12-01

    This paper describes an approach to the problem of searching speech-based digital audio using cross-modal information retrieval. Audio containing speech (speech-based audio) is difficult to search. Open vocabulary speech recognition is advancing rapidly, but cannot yield high accuracy in either search or transcription modalities. However, text can be searched quickly and efficiently with high accuracy. Script- light digital audio is audio that has an available transcription. This is a surprisingly large class of content including legal testimony, broadcasting, dramatic productions and political meetings and speeches. An automatic mechanism for deriving the synchronization between the transcription and the audio allows for very accurate retrieval of segments of that audio. The mechanism described in this paper is based on building a transcription graph from the text and computing biphone probabilities for the audio. A modified beam search algorithm is presented to compute the alignment.

  14. 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.

  15. Relevant Research on Audio-Tutorial Methods

    Science.gov (United States)

    Novak, Joseph D.

    1970-01-01

    Reviews two aspects of research related to audio-tutorial instructional methods. First, the learning theory of David P. Ausebel is summarized and applied to instructional procedures. Secondly, learning time for attainment of concept and knowledge levels is discussed. Concludes that studies are needed on designs based on Ausebel's theory,…

  16. 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.

  17. 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...

  18. 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…

  19. 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.

  20. Structuring Broadcast Audio for Information Access

    Directory of Open Access Journals (Sweden)

    Gauvain Jean-Luc

    2003-01-01

    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.

  1. Agency Video, Audio and Imagery Library

    Science.gov (United States)

    Grubbs, Rodney

    2015-01-01

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

  2. Audio-visual integration in schizophrenia

    NARCIS (Netherlands)

    Gelder, B.L.M.F. de; Vroomen, J.; Annen, L.; Masthoff, E.D.M.; Hodiamont, P.P.G.

    2003-01-01

    Integration of information provided simultaneously by audition and vision was studied in a group of 18 schizophrenic patients. They were compared to a control group, consisting of 12 normal adults of comparable age and education. By administering two tasks, each focusing on one aspect of audio-visua

  3. Audio-visual integration in schizophrenia.

    NARCIS (Netherlands)

    Gelder, B. de; Vroomen, J.; Annen, L.; Masthof, E.; Hodiamont, P.P.G.

    2003-01-01

    Integration of information provided simultaneously by audition and vision was studied in a group of 18 schizophrenic patients. They were compared to a control group, consisting of 12 normal adults of comparable age and education. By administering two tasks, each focusing on one aspect of audio-visua

  4. 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…

  5. 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 measu...

  6. Progressive Syntax-Rich Coding of Multichannel Audio Sources

    Directory of Open Access Journals (Sweden)

    Dai Yang

    2003-09-01

    Full Text Available 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 and stereo audio material. Little work has been done on progressive coding of multichannel audio sources. MPEG advanced audio coding (AAC is one of the most distinguished multichannel digital audio compression systems. Based on AAC, we develop in this work a progressive syntax-rich multichannel audio codec (PSMAC. It not only supports fine grain bit rate scalability for the multichannel audio bitstream but also provides several other desirable functionalities. A formal subjective listening test shows that the proposed algorithm achieves an excellent performance at several different bit rates when compared with MPEG AAC.

  7. Enhancement of LSB based Steganography for Hiding Image in Audio

    OpenAIRE

    Pradeep Kumar Singh; R.K.Aggrawal

    2010-01-01

    In this paper we will take an in-depth look on steganography by proposing a new method of Audio Steganography. Emphasize will be on the proposed scheme of image hiding in audio and its comparison with simple Least Significant Bit insertion method for data hiding in audio.

  8. 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...

  9. 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...

  10. 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...

  11. 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.

  12. Audio watermarking technologies for automatic cue sheet generation systems

    Science.gov (United States)

    Caccia, Giuseppe; Lancini, Rosa C.; Pascarella, Annalisa; Tubaro, Stefano; Vicario, Elena

    2001-08-01

    Usually watermark is used as a way for hiding information on digital media. The watermarked information may be used to allow copyright protection or user and media identification. In this paper we propose a watermarking scheme for digital audio signals that allow automatic identification of musical pieces transmitted in TV broadcasting programs. In our application the watermark must be, obviously, imperceptible to the users, should be robust to standard TV and radio editing and have a very low complexity. This last item is essential to allow a software real-time implementation of the insertion and detection of watermarks using only a minimum amount of the computation power of a modern PC. In the proposed method the input audio sequence is subdivided in frames. For each frame a watermark spread spectrum sequence is added to the original data. A two steps filtering procedure is used to generate the watermark from a Pseudo-Noise (PN) sequence. The filters approximate respectively the threshold and the frequency masking of the Human Auditory System (HAS). In the paper we discuss first the watermark embedding system then the detection approach. The results of a large set of subjective tests are also presented to demonstrate the quality and robustness of the proposed approach.

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

    DEFF Research Database (Denmark)

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

    2014-01-01

    , 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 accounted for 89% of the variance; the second dimension, accounting for 10% of the variance, was labeled “balance and blend.” © 2014 Acoustical Society of America...

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

    Directory of Open Access Journals (Sweden)

    Valenzise G

    2009-01-01

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

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

    Directory of Open Access Journals (Sweden)

    G. Valenzise

    2009-02-01

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

  16. 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.

  17. Practical Design of Delta-Sigma Multiple Description Audio Coding

    DEFF Research Database (Denmark)

    Leegaard, Jack Højholt; Østergaard, Jan; Jensen, Søren Holdt;

    2014-01-01

    framework is suitable for practical low-delay MD audio coding. In particular, we design a practical MD audio coder with two descriptions and provide simulations on real audio data. The simulations demonstrate that even when using low-dimensional noise-shaping, prediction, and resampling filters......, it is possible to obtain good quality audio in the presence of packet losses. Simulations on real audio reveal that, contrary to existing designs, it is straightforward to obtain a large number of trade-off points between side distortion and central distortion, which makes the proposed coder suitable for a wide...

  18. 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......Guided by the idea that musical human-computer interaction may become more effective, intuitive, and creative when basing its computer part on cognitively more plausible learning principles, we employ unsupervised incremental online learning (i.e. clustering) to build a system that predicts.......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...

  19. 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.

  20. 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.

  1. 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.

  2. 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...

  3. PDE-SVD Based Audio Denoising

    OpenAIRE

    Baravdish, George; Evangelista, Gianpaolo; Svensson, Olof; Sofya, Faten

    2012-01-01

    In this paper we present a new method for denoising audio signals. The method is based on the Singular Value Decomposition (SVD) of the frame matrix representing the signal inthe Overlap Add decomposition. Denoising is performed by modifying both the singular values, using a tapering model, and the singular vectors of the representation, using a nonlinear PDE method. The performance of the method is evaluated and compared with denoising obtained by filtering.

  4. Indexing spoken audio by LSA and SOMs

    OpenAIRE

    2000-01-01

    This paper presents an indexing system for spoken audio documents. The framework is indexing and retrieval of broadcast news. The proposed indexing system applies latent semantic analysis (LSA) and self-organizing maps (SOM) to map the documents into a semantic vector space and to display the semantic structures of the document collection. The SOM is also used to enhance the indexing of the documents that are difficult to decode. Relevant index terms and suitable index weights are computed by...

  5. 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.

  6. 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.

  7. 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.

  8. 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.

  9. Design of a New Audio Watermarking System Based on Human Auditory System

    Energy Technology Data Exchange (ETDEWEB)

    Shin, D.H. [Maqtech Co., Ltd., (Korea); Shin, S.W.; Kim, J.W.; Choi, J.U. [Markany Co., Ltd., (Korea); Kim, D.Y. [Bucheon College, Bucheon (Korea); Kim, S.H. [The University of Seoul, Seoul (Korea)

    2002-07-01

    In this paper, we propose a robust digital copyright-protection technique based on the concept of human auditory system. First, we propose a watermarking technique that accepts the various attacks such as, time scaling, pitch shift, add noise and a lot of lossy compression such as MP3, AAC, WMA. Second, we implement audio PD(portable device) for copyright protection using proposed method. The proposed watermarking technique is developed using digital filtering technique. Being designed according to critical band of HAS(human auditory system), the digital filters embed watermark without nearly affecting audio quality. Before processing of digital filtering, wavelet transform decomposes the input audio signal into several signals that are composed of specific frequencies. Then, we embed watermark in the decomposed signal (0kHz-11kHz) by designed band-stop digital filter. Watermarking detection algorithm is implemented on audio PD(portable device). Proposed watermarking technology embeds 2bits information per 15 seconds. If PD detects watermark '11', which means illegal song, PD displays 'Illegal Song' message on LCD, skips the song and plays the next song. The implemented detection algorithm in PD requires 19 MHz computational power, 7.9kBytes ROM and 10kBytes RAM. The suggested technique satisfies SDMI(secure digital music initiative) requirements of platform3 based on ARM9E core. (author). 9 refs., 8 figs.

  10. 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

    -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.......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...

  11. Performance Improvement of Threshold based Audio Steganography using Parallel Computation

    OpenAIRE

    Muhammad Shoaib; Zakir Khan; Danish Shehzad; Tamer Dag; Arif Iqbal Umar; Noor Ul Amin

    2016-01-01

    Audio steganography is used to hide secret information inside audio signal for the secure and reliable transfer of information. Various steganography techniques have been proposed and implemented to ensure adequate security level. The existing techniques either focus on the payload or security, but none of them has ensured both security and payload at same time. Data Dependency in existing solution was reluctant for the execution of steganography mechanism serially. The audio data and secret ...

  12. Stuttering and speech naturalness: audio and audiovisual judgments.

    Science.gov (United States)

    Martin, R R; Haroldson, S K

    1992-06-01

    Unsophisticated raters, using 9-point interval scales, judged speech naturalness and stuttering severity of recorded stutterer and nonstutterer speech samples. Raters judged separately the audio-only and audiovisual presentations of each sample. For speech naturalness judgments of stutterer samples, raters invariably judged the audiovisual presentation more unnatural than the audio presentation of the same sample; but for the nonstutterer samples, there was no difference between audio and audiovisual naturalness ratings. Stuttering severity ratings did not differ significantly between audio and audiovisual presentations of the same samples. Rater reliability, interrater agreement, and intrarater agreement for speech naturalness judgments were assessed.

  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. Design of audio amplifier%音响放大器设计

    Institute of Scientific and Technical Information of China (English)

    张文娟

    2015-01-01

    According to traditional audio amplifier design shortcomings, combined with modern electronic technology, using audio amplification IC, an approach of the audio amplifier is designed. Firstly,analysis the principle and circuit of the audio amplifier.then introduces the power circuit design, power rectifier, power filtering in detail;preamplifier design;PA design.At last, debugging the major technical indicators such as the effective frequency , THD, the rated output power, input impedance, output impedance etc. Debugging results showed that the main technical indicators have reached the design requirements, which has a good value for money, ideal for audio output.%针对传统音响放大的设计缺点,结合现代电子技术,采用音响放大集成电路,设计了一种音响放大器。首先分析了音响放大器的工作原理及电路组成,然后详细介绍了电源电路设计、电源整流、电源滤波;前置放大器设计;功率放大器设计;最后,分别对有效频率、总谐波失真、额定输出功率、输入阻抗、输出阻抗等主要技术指标进行了调试。调试结果表明,主要技术指标均达到设计要求,该设计具有较好的性价比,音响输出非常理想。

  16. 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.

  17. 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...

  18. 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

  19. 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...

  20. 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

  1. 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.

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

    Directory of Open Access Journals (Sweden)

    Carlos E. Galván-Tejada

    2016-01-01

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

  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. Audio spectrum analysis of umbilical artery Doppler ultrasound signals applied to a clinical material.

    Science.gov (United States)

    Thuring, Ann; Brännström, K Jonas; Jansson, Tomas; Maršál, Karel

    2014-12-01

    Analysis of umbilical artery flow velocity waveforms characterized by pulsatility index (PI) is used to evaluate fetoplacental circulation in high-risk pregnancies. However, an experienced sonographer may be able to further differentiate between various timbres of Doppler audio signals. Recently, we have developed a method for objective audio signal characterization; the method has been tested in an animal model. In the present pilot study, the method was for the first time applied to human pregnancies. Doppler umbilical artery velocimetry was performed in 13 preterm fetuses before and after two doses of 12 mg betamethasone. The auditory measure defined by the frequency band where the spectral energy had dropped 15 dB from its maximum level (MAXpeak-15 dB ), increased two days after betamethasone administration (p = 0.001) parallel with a less pronounced decrease in PI (p = 0.04). The new auditory parameter MAXpeak-15 dB reflected the changes more sensitively than the PI did.

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

    Science.gov (United States)

    Chuan, Ching-Hua; Chew, Elaine

    2006-12-01

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

  7. Effect of downsampling and compressive sensing on audio-based continuous cough monitoring.

    Science.gov (United States)

    Casaseca-de-la-Higuera, Pablo; Lesso, Paul; McKinstry, Brian; Pinnock, Hilary; Rabinovich, Roberto; McCloughan, Lucy; Monge-Álvarez, Jesús

    2015-01-01

    This paper presents an efficient cough detection system based on simple decision-tree classification of spectral features from a smartphone audio signal. Preliminary evaluation on voluntary coughs shows that the system can achieve 98% sensitivity and 97.13% specificity when the audio signal is sampled at full rate. With this baseline system, we study possible efficiency optimisations by evaluating the effect of downsampling below the Nyquist rate and how the system performance at low sampling frequencies can be improved by incorporating compressive sensing reconstruction schemes. Our results show that undersampling down to 400 Hz can still keep sensitivity and specificity values above 90% despite of aliasing. Furthermore, the sparsity of cough signals in the time domain allows keeping performance figures close to 90% when sampling at 100 Hz using compressive sensing schemes.

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

    Directory of Open Access Journals (Sweden)

    Elaine Chew

    2007-01-01

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

  9. Technical Evaluation Report. 65. Video-Conferencing with Audio Software

    Science.gov (United States)

    Baggaley, Jon; Klaas, Jim

    2006-01-01

    An online conference is illustrated using the format of a TV talk show. The conference combined live audio discussion with visual images spontaneously selected by the moderator in the manner of a TV control-room director. A combination of inexpensive online collaborative tools was used for the event, based on the browser-based audio-conferencing…

  10. 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...

  11. 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...

  12. DOA Estimation of Audio Sources in Reverberant Environments

    DEFF Research Database (Denmark)

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

    2016-01-01

    Reverberation is well-known to have a detrimental impact on many localization methods for audio sources. We address this problem by imposing a model for the early reflections as well as a model for the audio source itself. Using these models, we propose two iterative localization methods that est...

  13. Improving audio chord transcription by exploiting harmonic and metric knowledge

    NARCIS (Netherlands)

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

    2012-01-01

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

  14. 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

  15. 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-deci...

  16. 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…

  17. Performance Analysis of Data Hiding in MPEG-4 AAC Audio

    Institute of Scientific and Technical Information of China (English)

    XU Shuzheng; ZHANG Peng; WANG Pengjun; YANG Huazhong

    2009-01-01

    A high capacity data hiding technique was developed for compressed digital audio.As perceptual audio coding has become the accepted technology for storage and transmission of audio signals,compressed audio information hiding enables robust,imperceptible transmission of data within audio signals,thus allowing valuable information to be attached to the content,such as the song title,lyrics,composer's name,and artist or property rights related data.This paper describes simultaneous low bitrate encoding and information hiding for highly compressed audio signals.The information hiding is implemented in the quantization process of the audio content which improves robustness,signal quality,and security.The imperceptibility of the embedded data is ensured based on the masking property of the human auditory system (HAS).The robustness and security are evaluated by various attacking algorithms.Tests with an extended MPEG4 advanced audio coding (AAC) encoder confirm that the method is robust to the regular and singular groups method (RS) and sample pair analysis (SPA) attacks as well as other statistical steganalysis method attacks.

  18. Multi Carrier Modulation Audio Power Amplifier with Programmable Logic

    DEFF Research Database (Denmark)

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

    2009-01-01

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

  19. Decision-level fusion for audio-visual laughter detection

    NARCIS (Netherlands)

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

    2008-01-01

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

  20. An Audio Stream Redirector for the Ethernet Speaker

    Science.gov (United States)

    Mandrekar, Ishan; Prevelakis, Vassilis; Turner, David Michael

    2004-01-01

    The authors have developed the "Ethernet Speaker" (ES), a network-enabled single board computer embedded into a conventional audio speaker. Audio streams are transmitted in the local area network using multicast packets, and the ES can select any one of them and play it back. A key requirement for the ES is that it must be capable of playing any…

  1. Circular microphone array for multi channel audio recording

    NARCIS (Netherlands)

    Hulsebos, E.M.; De Vries, D.; Boone, M.M.; Schuurmans, T.J.G.

    2004-01-01

    An audio system has a circular microphone array with a number of microphones arranged on a circle for receiving a sound field. A digital signal processor is provided for processing output signals from these microphones. To establish well controlled and sharp directivity patterns the audio system per

  2. Decision-Level Fusion for Audio-Visual Laughter Detection

    NARCIS (Netherlands)

    Reuderink, Boris; Poel, Mannes; Truong, Khiet; Poppe, Ronald; Pantic, Maja; Popescu-Belis, Andrei; Stiefelhagen, Rainer

    2008-01-01

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

  3. Dynamic Bayesian Networks for Audio-Visual Speech Recognition

    Directory of Open Access Journals (Sweden)

    Liang Luhong

    2002-01-01

    Full Text Available The use of visual features in audio-visual speech recognition (AVSR is justified by both the speech generation mechanism, which is essentially bimodal in audio and visual representation, and by the need for features that are invariant to acoustic noise perturbation. As a result, current AVSR systems demonstrate significant accuracy improvements in environments affected by acoustic noise. In this paper, we describe the use of two statistical models for audio-visual integration, the coupled HMM (CHMM and the factorial HMM (FHMM, and compare the performance of these models with the existing models used in speaker dependent audio-visual isolated word recognition. The statistical properties of both the CHMM and FHMM allow to model the state asynchrony of the audio and visual observation sequences while preserving their natural correlation over time. In our experiments, the CHMM performs best overall, outperforming all the existing models and the FHMM.

  4. High Capacity and Resistance to Additive Noise Audio Steganography Algorithm

    Directory of Open Access Journals (Sweden)

    Haider Ismael Shahadi

    2011-09-01

    Full Text Available Steganography is the art of message hiding in a cover signal without attracting attention. The requirements of the good steganography algorithm are security, capacity, robustness and imperceptibility, all them are contradictory, therefore, satisfying all together is not easy especially in audio cover signal because human auditory system (HAS has high sensitivity to audio modification. In this paper, we proposed a high capacity audio steganography algorithm with good resistance to additive noise. The proposed algorithm is based on wavelet packet transform and blocks matching. It has capacity above 35% of the input audio file size with acceptable signal to noise ratio. Also, it is resistance to additive Gaussian noise to about 25 db. Furthermore, the reconstruction of actual secret messages does not require the original cover audio signal.

  5. 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…

  6. 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

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

    Directory of Open Access Journals (Sweden)

    Smita Paira

    2016-09-01

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

  8. 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.

  9. Enhanced Audio LSB Steganography for Secure Communication

    Directory of Open Access Journals (Sweden)

    Muhammad Junaid Hussain

    2016-01-01

    Full Text Available The ease with which data can be remitted across the globe via Internet has made it an obvious (as medium choice for on line data transmission and communication. This salient trait, however, is constraint with akin issues of privacy, veracity of the information being exchanged over it, and legitimacy of its sender together with its availability when needed. Although cryptography is being used to confront confidentiality concern yet for many is slightly limited in scope because of discernibility of encrypted information. Further, due to restrictions imposed on the use of cryptography by its citizens for personal doings, various Governments have also coxswained the research arena to explore another discipline of information hiding called steganography – whose sole purpose is to make the information being exchanged inaudible. This research is focused on evolution of model based secure LSB Steganographic scheme for digital audio wave file format to withstand passive attack by Warden Wendy.

  10. Particle Filtering on the Audio Localization Manifold

    CERN Document Server

    Ettinger, Evan

    2010-01-01

    We present a novel particle filtering algorithm for tracking a moving sound source using a microphone array. If there are N microphones in the array, we track all $N \\choose 2$ delays with a single particle filter over time. Since it is known that tracking in high dimensions is rife with difficulties, we instead integrate into our particle filter a model of the low dimensional manifold that these delays lie on. Our manifold model is based off of work on modeling low dimensional manifolds via random projection trees [1]. In addition, we also introduce a new weighting scheme to our particle filtering algorithm based on recent advancements in online learning. We show that our novel TDOA tracking algorithm that integrates a manifold model can greatly outperform standard particle filters on this audio tracking task.

  11. A direct broadcast satellite-audio experiment

    Science.gov (United States)

    Vaisnys, Arvydas; Abbe, Brian; Motamedi, Masoud

    1992-03-01

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

  12. Time-Scale Invariant Audio Data Embedding

    Directory of Open Access Journals (Sweden)

    Mansour Mohamed F

    2003-01-01

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

  13. 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.

  14. Quality Enhancement of Compressed Audio Based on Statistical Conversion

    Directory of Open Access Journals (Sweden)

    Mouchtaris Athanasios

    2008-01-01

    Full Text Available Most audio compression formats are based on the idea of low bit rate transparent encoding. As these types of audio signals are starting to migrate from portable players with inexpensive headphones to higher quality home audio systems, it is becoming evident that higher bit rates may be required to maintain transparency. We propose a novel method that enhances low bit rate encoded audio segments by applying multiband audio resynthesis methods in a postprocessing stage. Our algorithm employs the highly flexible Generalized Gaussian mixture model which offers a more accurate representation of audio features than the Gaussian mixture model. A novel residual conversion technique is applied which proves to significantly improve the enhancement performance without excessive overhead. In addition, both cepstral and residual errors are dramatically decreased by a feature-alignment scheme that employs a sorting transformation. Some improvements regarding the quantization step are also described that enable us to further reduce the algorithm overhead. Signal enhancement examples are presented and the results show that the overhead size incurred by the algorithm is a fraction of the uncompressed signal size. Our results show that the resulting audio quality is comparable to that of a standard perceptual codec operating at approximately the same bit rate.

  15. Quality Enhancement of Compressed Audio Based on Statistical Conversion

    Directory of Open Access Journals (Sweden)

    Chris Kyriakakis

    2008-07-01

    Full Text Available Most audio compression formats are based on the idea of low bit rate transparent encoding. As these types of audio signals are starting to migrate from portable players with inexpensive headphones to higher quality home audio systems, it is becoming evident that higher bit rates may be required to maintain transparency. We propose a novel method that enhances low bit rate encoded audio segments by applying multiband audio resynthesis methods in a postprocessing stage. Our algorithm employs the highly flexible Generalized Gaussian mixture model which offers a more accurate representation of audio features than the Gaussian mixture model. A novel residual conversion technique is applied which proves to significantly improve the enhancement performance without excessive overhead. In addition, both cepstral and residual errors are dramatically decreased by a feature-alignment scheme that employs a sorting transformation. Some improvements regarding the quantization step are also described that enable us to further reduce the algorithm overhead. Signal enhancement examples are presented and the results show that the overhead size incurred by the algorithm is a fraction of the uncompressed signal size. Our results show that the resulting audio quality is comparable to that of a standard perceptual codec operating at approximately the same bit rate.

  16. Sampling Function of Degree 2 for DVD-Audio

    Science.gov (United States)

    Toraichi, Kazuo; Nakamura, Koji

    Authors have been studying Fluency Information Theory that generalizes Shannon’s sampling theorem and its applications. Among the practical application of the research, the Fluency DAC that is developed as the Digital-to-analog converter for CD audio could have received objective valuation including receipt Golden Sound Award in 1988. In recent, DVD-Audio that deal with maximum sampling rate of 192 kHz has appeared. Due to the introduction of DVD audio that requires four times the sampling rate of nowadays CD audio, the request for developing a new Fluency DAC for DVD audio was initiated. From such requirements, the research for developing the Fluency DAC for DVD-Audio has been started. The result of the research could revive awards in local contest in Japan audio apparatus at 2000 and 2001. As the initial report on our project in developing the Fluency DAC that is capable of dealing with a maximum sampling rate of 192kHz, in this paper we aimed to derive the sampling function that acts as the impulse response for such a D/A converter.

  17. Optimized Audio Classification and Segmentation Algorithm by Using Ensemble Methods

    Directory of Open Access Journals (Sweden)

    Saadia Zahid

    2015-01-01

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

  18. 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.

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

    Directory of Open Access Journals (Sweden)

    Xiang Shijun

    2011-01-01

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

  20. Lattice Vector Quantization Applied to Speech and Audio Coding

    Institute of Scientific and Technical Information of China (English)

    Minjie Xie

    2012-01-01

    Lattice vector quantization (LVQ) has been used for real-time speech and audio coding systems. Compared with conventional vector quantization, LVQ has two main advantages: It has a simple and fast encoding process, and it significantly reduces the amount of memory required. Therefore, LVQ is suitable for use in low-complexity speech and audio coding. In this paper, we describe the basic concepts of LVQ and its advantages over conventional vector quantization. We also describe some LVQ techniques that have been used in speech and audio coding standards of international standards developing organizations (SDOs).

  1. A Review on Audio-visual Translation Studies

    Institute of Scientific and Technical Information of China (English)

    李瑶

    2008-01-01

    <正>This paper is dedicated to a thorough review on the audio-visual related translations from both home and abroad.In reviewing the foreign achievements on this specific field of translation studies it can shed some lights on our national audio-visual practice and research.The review on the Chinese scholars’ audio-visual translation studies is to offer the potential developing direction and guidelines to the studies and aspects neglected as well.Based on the summary of relevant studies,possible topics for further studies are proposed.

  2. 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...... 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....

  3. A New Steganographic Method for Embedded Image In Audio File

    Directory of Open Access Journals (Sweden)

    Mohammed S. Altaei

    2012-04-01

    Full Text Available Because secure transaction of information is increasing day by day therefore Steganography hasbecome very important and used modern strategies. Steganography is a strategy in whichrequired information is concealment in any other information such that the second informationdoes not change significantly and it appears the same as original. This work presents a newapproach of concealment encrypted mobile image in a audio file.The proposed work is replacingtwo LSB of each byte in audio file and these bytes are choices as randomly location. It becomesvery difficult for intruder to guess that an image is hidden in the audio.

  4. Robust message authentication code algorithm for digital audio recordings

    Science.gov (United States)

    Zmudzinski, Sascha; Steinebach, Martin

    2007-02-01

    Current systems and protocols for integrity and authenticity verification of media data do not distinguish between legitimate signal transformation and malicious tampering that manipulates the content. Furthermore, they usually provide no localization or assessment of the relevance of such manipulations with respect to human perception or semantics. We present an algorithm for a robust message authentication code (RMAC) to verify the integrity of audio recodings by means of robust audio fingerprinting and robust perceptual hashing. Experimental results show that the proposed algorithm provides both a high level of distinction between perceptually different audio data and a high robustness against signal transformations that do not change the perceived information.

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

    DEFF Research Database (Denmark)

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

    2016-01-01

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

  6. 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 amplitude panning audio system (panning) in a virtual environment. We present a performance study involving 33 participants locating aurally-aided visual targets placed at fixed positions, under different audio conditions. A varying amount of visual distractors were present, represented as black circles...

  7. 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

  8. 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....

  9. A Robust Audio Watermarking Technique Operates in MDCT Domain based on Perceptual Measures

    Directory of Open Access Journals (Sweden)

    Maha Bellaaj

    2016-06-01

    Full Text Available the review presents a digital audio watermarking technique operating in the frequency domain with two variants. This technique uses the Modified Discrete Cosine Transform (MDCT to move to the frequency domain. To ensure more inaudibility, we exploited the proprieties of the psychoacoustic model 1 (PMH1 of MPEG1 encoder layer I in the first variant and those of psychoacoustic model 2 (PMH2 of MPEG1 encoder Layer III in the second alternative to search the places for insertion of the watermark. In both variants of the technique, the bits of the mark will be duplicated to increase the capacity of insertion then inserted into the least significant bit (LSB. For more reliability in the detection phase, we use an error correction code (Hamming on the mark. Next, to analyze the performance of the proposed technique, we perform two comparative studies. In the first, we compare the proposed digital audio watermarking technique with her two variants and those achieved by Luigi Rosa and Rolf Brigola, ‘which we download the M-files of each’. The technique developed by Luigi Rosa operates in the frequency domain but using the Discrete Cosine Transform (DCT as transformation and that proposed by Rolf Brigola uses the Fast Fourier Transform (FFT. We studied the robustness of each technique against different types of attacks such as compression / decompression MP3, stirmark audio attack and we evaluated the inaudibility by using an objective approach by calculating the SNR and the ODG notes given by PEAQ. The robustness of this technique is shown against different types of attacks. In the second, we prove the contribution of the proposed technique by comparing the payload data, imperceptibility and robustness against attack MP3 with others existing techniques in the literature.

  10. 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...

  11. 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.

  12. Perancangan Sistem Audio Mobil berbasiskan Sistem Pakar dan Web

    Directory of Open Access Journals (Sweden)

    Djunaidy Santoso

    2011-11-01

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

  13. Effectiveness of 3-D audio for warnings in the cockpit

    NARCIS (Netherlands)

    Oving, A.B.; Veltman, J.A.; Bronkhorst, A.W.

    2004-01-01

    Een tweetal vliegsimulator experimenten lieten zien dat piloten sneller reagereerden op de auditieve waarschuwingen van het TCAS systeem in de civiele cockpit, waneer deze waarschuwingen werden gepresenteerd met 3D-audio in vergelijking tot mono geluid.

  14. 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)

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

    DEFF Research Database (Denmark)

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

    Introduction In order to give patients possibility to listen to their consultation again, we have designed a system which gives the patients access to digital audio recordings of their consultations. An Interactive Voice Response platform enables the audio recording and gives the patients access...... to replay their consultation. The intervention is evaluated in a randomised controlled trial with 5.460 patients in order to determine whether providing patients with digital audio recording of the consultation affects the patients overall perception of their consultation. In addition to this primary...... 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...

  16. Audio-Visual Integration of Emotional Information

    Directory of Open Access Journals (Sweden)

    Penny Bergman

    2011-10-01

    Full Text Available Emotions are central to our perception of the environment surrounding us (Berlyne, 1971. An important aspect in the emotional response to a sound is dependent on the meaning of the sound, ie, it is not the physical parameter per se that determines our emotional response to the sound but rather the source of the sound (Genell, 2008, and the relevance it has to the self (Tajadura-Jiménez et al 2010. When exposed to sound together with visual information, the information from both modalities is integrated, altering the perception of each modality, in order to generate a coherent experience. In emotional information this integration is rapid and without requirements of attentional processes (De Gelder, 1999. The present experiment investigates perception of pink noise in two visual settings in a within-subjects design. Nineteen participants rated the same sound twice in terms of pleasantness and arousal in either a pleasant or an unpleasant visual setting. The results showed that pleasantness of the sound decreased in the negative visual setting, thus suggesting an audio-visual integration, where the affective information in the visual modality is translated to the auditory modality when information-markers are lacking in it. The results are discussed in relation to theories of emotion perception.

  17. 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.

  18. Multi-Level Audio Classification Architecture

    Directory of Open Access Journals (Sweden)

    Jozef Vavrek

    2015-01-01

    Full Text Available A multi-level classification architecture for solving binary discrimination problem is proposed in this paper. The main idea of proposed solution is derived from the fact that solving one binary discrimination problem multiple times can reduce the overall miss-classification error. We aimed our effort towards building the classification architecture employing the combination of multiple binary SVM (Support Vector Machine classifiers for solving two-class discrimination problem. Therefore, we developed a binary discrimination architecture employing the SVM classifier (BDASVM with intention to use it for classification of broadcast news (BN audio data. The fundamental element of BDASVM is the binary decision (BD algorithm that performs discrimination between each pair of acoustic classes utilizing decision function modeled by separating hyperplane. The overall classification accuracy is conditioned by finding the optimal parameters for discrimination function resulting in higher computational complexity. The final form of proposed BDASVM is created by combining four BDSVM discriminators supplemented by decision table. Experimental results show that the proposed classification architecture can decrease the overall classification error in comparison with binary decision trees SVM (BDTSVM architecture.

  19. IELTS speaking instruction through audio/voice conferencing

    Directory of Open Access Journals (Sweden)

    Hamed Ghaemi

    2012-02-01

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

  20. Virtual environment interaction through 3D audio by blind children.

    Science.gov (United States)

    Sánchez, J; Lumbreras, M

    1999-01-01

    Interactive software is actively used for learning, cognition, and entertainment purposes. Educational entertainment software is not very popular among blind children because most computer games and electronic toys have interfaces that are only accessible through visual cues. This work applies the concept of interactive hyperstories to blind children. Hyperstories are implemented in a 3D acoustic virtual world. In past studies we have conceptualized a model to design hyperstories. This study illustrates the feasibility of the model. It also provides an introduction to researchers to the field of entertainment software for blind children. As a result, we have designed and field tested AudioDoom, a virtual environment interacted through 3D Audio by blind children. AudioDoom is also a software that enables testing nontrivial interfaces and cognitive tasks with blind children. We explored the construction of cognitive spatial structures in the minds of blind children through audio-based entertainment and spatial sound navigable experiences. Children playing AudioDoom were exposed to first person experiences by exploring highly interactive virtual worlds through the use of 3D aural representations of the space. This experience was structured in several cognitive tasks where they had to build concrete models of their spatial representations constructed through the interaction with AudioDoom by using Legotrade mark blocks. We analyze our preliminary results after testing AudioDoom with Chilean children from a school for blind children. We discuss issues such as interactivity in software without visual cues, the representation of spatial sound navigable experiences, and entertainment software such as computer games for blind children. We also evaluate the feasibility to construct virtual environments through the design of dynamic learning materials with audio cues.

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

    DEFF Research Database (Denmark)

    Dimitrov, Smilen; Serafin, Stefania

    2011-01-01

    Technology Devices International Ltd [FTDI] company) can be demonstrated to behave as a full-duplex, mono, 8-bit 44.1 kHz soundcard, through an implementation of: a PC audio driver for ALSA (Advanced Linux Sound Architecture); a matching program for the Arduino's ATmega microcontroller - and nothing more...... than headphones (and a couple of capacitors). The main contribution of this paper is to bring a holistic aspect to the discussion on the topic of implementation of soundcards - also by referring to open-source driver, microcontroller code and test methods; and outline a complete implementation...

  2. Analysis of current-bidirectional buck-boost based switch-mode audio amplifier

    DEFF Research Database (Denmark)

    Bolten Maizonave, Gert; Andersen, Michael A. E.; Kjærgaard, Claus;

    2011-01-01

    in a differential mode buckbased amplifier with a boost converter as power supply. The averaged switch modelling of the differential mode current bidirectional topology is also used, in order to analyze the steady state and frequency-wise behaviour of this converter and parameterize it to meet the design criteria......The following studdy was carried out in order to assses quantitatively the performannce of the buck--boost converter whhen used as swiitch-mode audio amplifier. It comprises of, to beggin with, the de limitation of design criteria bassed on the state of-the-art solution, which is based...

  3. Automated processing of massive audio/video content using FFmpeg

    Directory of Open Access Journals (Sweden)

    Kia Siang Hock

    2014-01-01

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

  4. 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.

  5. Calibration of Frequency Data Collection Systems Using Shortwave Radio Signals

    Science.gov (United States)

    Estler, Ron

    2000-09-01

    The atomic-clock-derived audio tones broadcast on the National Institute of Standards and Technology (NIST) shortwave station WWV are used to calibrate computer frequency data collection systems via Fast Fourier Transforms (FFT). Once calibrated, the data collection system can be used to accurately determine the audio signals used in several instructional physical chemistry laboratory experiments. This method can be applied to virtually any hardware-software configuration that allows adjustment of the apparent time scale (digitizing rate) of the recorded audio file.

  6. 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.

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

    Institute of Scientific and Technical Information of China (English)

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

    2011-01-01

    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 0.18μm 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% @ lx 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.

  8. An Adaptive Robust Watermarking Algorithm for Audio Signals Using SVD

    Science.gov (United States)

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

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

  9. The Fungible Audio-Visual Mapping and its Experience

    Directory of Open Access Journals (Sweden)

    Adriana Sa

    2014-12-01

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

  10. Audio Watermarking Based on HAS and Neural Networks in DCT Domain

    Directory of Open Access Journals (Sweden)

    Cheng Ji-Shiung

    2003-01-01

    Full Text Available We propose a new intelligent audio watermarking method based on the characteristics of the HAS and the techniques of neural networks in the DCT domain. The method makes the watermark imperceptible by using the audio masking characteristics of the HAS. Moreover, the method exploits a neural network for memorizing the relationships between the original audio signals and the watermarked audio signals. Therefore, the method is capable of extracting watermarks without original audio signals. Finally, the experimental results are also included to illustrate that the method significantly possesses robustness to be immune against common attacks for the copyright protection of digital audio.

  11. Performance Improvement of Threshold based Audio Steganography using Parallel Computation

    Directory of Open Access Journals (Sweden)

    Muhammad Shoaib

    2016-10-01

    Full Text Available Audio steganography is used to hide secret information inside audio signal for the secure and reliable transfer of information. Various steganography techniques have been proposed and implemented to ensure adequate security level. The existing techniques either focus on the payload or security, but none of them has ensured both security and payload at same time. Data Dependency in existing solution was reluctant for the execution of steganography mechanism serially. The audio data and secret data pre-processing were done and existing techniques were experimentally tested in Matlab that ensured the existence of problem in efficient execution. The efficient least significant bit steganography scheme removed the pipelining hazard and calculated Steganography parallel on distributed memory systems. This scheme ensures security, focuses on payload along with provisioning of efficient solution. The result depicts that it not only ensures adequate security level but also provides better and efficient solution.

  12. Audio system using binaural synthesis for multimodal telepresence applications

    DEFF Research Database (Denmark)

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

    2013-01-01

    of 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......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...... a person (the visitor) to a different physical location (the destination). The goal is that both the visitor and the people physically at the destination (the locals) should be provided with a sensation that the visitor is really there. Both the general multimodal system and the auditory part...

  13. 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.

  14. Sistema de adquisición y procesamiento de audio

    OpenAIRE

    Pérez Segurado, Rubén

    2015-01-01

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

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

    DEFF Research Database (Denmark)

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

    different departments: Orthopedics, Urology, Internal Medicine and Pediatrics. A total of 5,460 patients will be included from the outpatient clinics. All patients randomized to an intervention group are offered audio recording of their consultation. An Interactive Voice Response platform enables an audio......Introduction Information provided in an outpatient consultation concerns medication, diagnostic tests, treatment and rehabilitation which is crucial knowledge in regards of patient compliance, decision making and general patient satisfaction. Despite good communication skills among clinicians...... the communication is challenged by the fact that patients tend to forget or misunderstand 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...

  16. Improving Security of Audio Watermarking in Image using Selector Keys

    Directory of Open Access Journals (Sweden)

    Amir Reza Fazli

    2012-06-01

    Full Text Available This study presents a novel watermarking algorithm for improving the security and robustness of hiding audio data in an image. Multi resolution discrete wavelet transform is used for embedding the audio watermark in an image. In this context, security is quantified from an information theoretic point of view by means of the equivocation and information leakage of the secret parameters. The selector keys are used as a criterion to determine the location of appropriate wavelet blocks and wavelet coefficients for embedding the watermark. Also, simulations assess the security levels derived in the theoretical part of the paper. The experimental results demonstrate that using the selector keys enhance the security level of the watermark embedding for a variety of scenarios. The level of the algorithm robustness is shown by considering Normalized Correlation (NC between the original audio watermark and extracted watermark.

  17. 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.

  18. 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.

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

    DEFF Research Database (Denmark)

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

    2016-01-01

    the interpolation filter and the SD modulator on the system level so that the switching frequency of the Class D PA - the main power consumer in the back end - is minimized. A figure-of-merit (FOM) which allows judging the power consumption of the digital part of the back end early in the design process is used...... to track the hardware and power demands as the tradeoffs of the system level parameters are investigated. The result is the digital part of the back end optimized with respect to power which provides audio performance comparable to state-of-theart. A combination of system level parameters leading...... to the lowest switching frequency of the Class D power amplifier reported in literature for the SD modulatorbased back end is derived using this approach....

  20. Digital audio broadcasting by satellite utilising Trellis-Coded Quasi-Orthogonal Code Division Multiplexing

    Science.gov (United States)

    de Gaudenzi, R.

    This paper introduces trellis-coded quasi-orthogonal code division multiplexing (TCQO-CDM) as a transmission technique for digital audio broadcasting. The proposed technique performs well over the satellite L-band fading channel and also in the terrestrial gap-filter type of transmission. Preliminary satellite link budgets based on extensive computer-simulation results are provided. The capacity achieved by the terrestrial single-frequency gap-filler network by using the same satellite frequency and user receiver is also discussed. Numerical results show that a remarkable overall capacity can be achieved by using HEO satellite orbits complemented by a terrestrial gap-filler. A variety of transmission rates and hence broadcasting services can be realized. It is shown that a geostationary satellite can provide limited service availability and limited capacity to mobile users, but can also be used for experimental purposes.

  1. Evaluation of robustness and transparency of multiple audio watermark embedding

    Science.gov (United States)

    Steinebach, Martin; Zmudzinski, Sascha

    2008-02-01

    As digital watermarking becomes an accepted and widely applied technology, a number of concerns regarding its reliability in typical application scenarios come up. One important and often discussed question is the robustness of digital watermarks against multiple embedding. This means that one cover is marked several times by various users with by same watermarking algorithm but with different keys and different watermark messages. In our paper we discuss the behavior of our PCM audio watermarking algorithm when applying multiple watermark embedding. This includes evaluation of robustness and transparency. Test results for multiple hours of audio content ranging from spoken words to music are provided.

  2. Technical Evaluation Report 52: Audio/ Videoconferencing Packages: High cost

    Directory of Open Access Journals (Sweden)

    Urel Sawyers

    2005-11-01

    Full Text Available This report compares two integrated course delivery packages: Centra 6 and WebEx. Both applications feature asynchronous and synchronous audio communications for online education and training. They are relatively costly products, and provide useful comparisons with the two less expensive products to be evaluated in the following report #53. The criteria used in the current evaluation include capacity, interactivity features, integration with learning management systems, technical specifications, and cost. The report ends with a short analysis of the currently emerging audio-conferencing software, Google Talk.

  3. 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 .

  4. 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

  5. DOA Estimation of Audio Sources in Reverberant Environments

    DEFF Research Database (Denmark)

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

    2016-01-01

    Reverberation is well-known to have a detrimental impact on many localization methods for audio sources. We address this problem by imposing a model for the early reflections as well as a model for the audio source itself. Using these models, we propose two iterative localization methods that est...... 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....

  6. Design of a WAV audio player based on K20

    Directory of Open Access Journals (Sweden)

    Xu Yu

    2016-01-01

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

  7. Technical Evaluation Report 56: Video-Conferencing with Audio Software

    Directory of Open Access Journals (Sweden)

    Jon Baggaley

    2006-06-01

    Full Text Available An online conference is illustrated using the format of a TV talk show. The conference combined live audio discussion with visual images spontaneously selected by the moderator in the manner of a TV control-room director. A combination of inexpensive online collaborative tools was used for the event, based on the browser-based audio-conferencing software, iVocalize. The exercise illustrates how an impression of a fully featured online video-conference can be created without the need for complex video-conferencing software and high bandwidth.

  8. 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.

  9. 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....

  10. SNR-adaptive stream weighting for audio-MES ASR.

    Science.gov (United States)

    Lee, Ki-Seung

    2008-08-01

    Myoelectric signals (MESs) from the speaker's mouth region have been successfully shown to improve the noise robustness of automatic speech recognizers (ASRs), thus promising to extend their usability in implementing noise-robust ASR. In the recognition system presented herein, extracted audio and facial MES features were integrated by a decision fusion method, where the likelihood score of the audio-MES observation vector was given by a linear combination of class-conditional observation log-likelihoods of two classifiers, using appropriate weights. We developed a weighting process adaptive to SNRs. The main objective of the paper involves determining the optimal SNR classification boundaries and constructing a set of optimum stream weights for each SNR class. These two parameters were determined by a method based on a maximum mutual information criterion. Acoustic and facial MES data were collected from five subjects, using a 60-word vocabulary. Four types of acoustic noise including babble, car, aircraft, and white noise were acoustically added to clean speech signals with SNR ranging from -14 to 31 dB. The classification accuracy of the audio ASR was as low as 25.5%. Whereas, the classification accuracy of the MES ASR was 85.2%. The classification accuracy could be further improved by employing the proposed audio-MES weighting method, which was as high as 89.4% in the case of babble noise. A similar result was also found for the other types of noise.

  11. 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…

  12. 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.

  13. 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…

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

    DEFF Research Database (Denmark)

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

    2008-01-01

    While switch-mode audio power amplifiers allow compact implementations and high output power levels due to their high power efficiency, they are very well known for creating electromagnetic interference (EMI) with other electronic equipment, in particular radio receivers. Lowering the EMI of swit...

  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 sw...

  16. 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...

  17. Objective assessment of speech and audio quality - Technology and applications

    NARCIS (Netherlands)

    Rix, A.W.; Beerends, J.G.; Kim, D.-S.; Kroon, P.; Ghitza, O.

    2006-01-01

    In the past few years, objective quality assessment models have become increasingly used for assessing or monitoring speech and audio quality. By measuring perceived quality on an easily-understood subjective scale, such as listening quality (excellent, good, fair, poor, bad), these methods provide

  18. 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.

  19. Output impedance and stability of audio power amplifiers

    NARCIS (Netherlands)

    Schaink, T.

    2006-01-01

    This report is about the design of an audio amplifier which is stable for all passive loads. If stability analysis of an opamp is done, the ‘classical’ approach is to derive its transfer function. Investigation of the open loop gain and a phase/gain margin determine the stability of the opamp. Desig

  20. 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.

  1. A Multimedia Application: Spatial Perceptual Entropy of Multichannel Audio Signals

    Directory of Open Access Journals (Sweden)

    Chen Shuixian

    2010-01-01

    Full Text Available Usually multimedia data have to be compressed before transmitting, and higher compression rate, or equivalently lower bitrate, relieves the load of communication channels but impacts negatively the quality. We investigate the bitrate lower bound for perceptually lossless compression of a major type of multimedia—multichannel audio signals. This bound equals to the perceptible information rate of the signals. Traditionally, Perceptual Entropy (PE, based primarily on monaural hearing measures the perceptual information rate of individual channels. But PE cannot measure the spatial information captured by binaural hearing, thus is not suitable for estimating Spatial Audio Coding (SAC bitrate bound. To measure this spatial information, we build a Binaural Cue Physiological Perception Model (BCPPM on the ground of binaural hearing, which represents spatial information in the physical and physiological layers. This model enables computing Spatial Perceptual Entropy (SPE, the lower bitrate bound for SAC. For real-world stereo audio signals of various types, our experiments indicate that SPE reliably estimates their spatial information rate. Therefore, "SPE plus PE" gives lower bitrate bounds for communicating multichannel audio signals with transparent quality.

  2. A Multimedia Application: Spatial Perceptual Entropy of Multichannel Audio Signals

    Directory of Open Access Journals (Sweden)

    Shuixian Chen

    2010-01-01

    Full Text Available Usually multimedia data have to be compressed before transmitting, and higher compression rate, or equivalently lower bitrate, relieves the load of communication channels but impacts negatively the quality. We investigate the bitrate lower bound for perceptually lossless compression of a major type of multimedia—multichannel audio signals. This bound equals to the perceptible information rate of the signals. Traditionally, Perceptual Entropy (PE, based primarily on monaural hearing measures the perceptual information rate of individual channels. But PE cannot measure the spatial information captured by binaural hearing, thus is not suitable for estimating Spatial Audio Coding (SAC bitrate bound. To measure this spatial information, we build a Binaural Cue Physiological Perception Model (BCPPM on the ground of binaural hearing, which represents spatial information in the physical and physiological layers. This model enables computing Spatial Perceptual Entropy (SPE, the lower bitrate bound for SAC. For real-world stereo audio signals of various types, our experiments indicate that SPE reliably estimates their spatial information rate. Therefore, “SPE plus PE” gives lower bitrate bounds for communicating multichannel audio signals with transparent quality.

  3. An Audio-Visual Lecture Course in Russian Culture

    Science.gov (United States)

    Leighton, Lauren G.

    1977-01-01

    An audio-visual course in Russian culture is given at Northern Illinois University. A collection of 4-5,000 color slides is the basis for the course, with lectures focussed on literature, philosophy, religion, politics, art and crafts. Acquisition, classification, storage and presentation of slides, and organization of lectures are discussed. (CHK)

  4. 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...

  5. 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...

  6. 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…

  7. Auteur Description: From the Director's Creative Vision to Audio Description

    Science.gov (United States)

    Szarkowska, Agnieszka

    2013-01-01

    In this report, the author follows the suggestion that a film director's creative vision should be incorporated into Audio description (AD), a major technique for making films, theater performances, operas, and other events accessible to people who are blind or have low vision. The author presents a new type of AD for auteur and artistic films:…

  8. Real-time Loudspeaker Distance Estimation with Stereo Audio

    DEFF Research Database (Denmark)

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

    2015-01-01

    . 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 variance...

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

    Science.gov (United States)

    Schoeffler, Michael; Herre, Jürgen

    2016-09-01

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

  10. Market potential for interactive audio-visual media

    NARCIS (Netherlands)

    Leurdijk, A.; Limonard, S.

    2005-01-01

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

  11. Digital audio recordings improve the outcomes of patient consultations

    DEFF Research Database (Denmark)

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

    2016-01-01

    OBJECTIVES: To investigate the effects on patients' outcome of the consultations when provided with: a Digital Audio Recording (DAR) of the consultation and a Question Prompt List (QPL). METHODS: This is a three-armed randomised controlled cluster trial. One group of patients received standard care...

  12. Video genre categorization and representation using audio-visual information

    Science.gov (United States)

    Ionescu, Bogdan; Seyerlehner, Klaus; Rasche, Christoph; Vertan, Constantin; Lambert, Patrick

    2012-04-01

    We propose an audio-visual approach to video genre classification using content descriptors that exploit audio, color, temporal, and contour information. Audio information is extracted at block-level, which has the advantage of capturing local temporal information. At the temporal structure level, we consider action content in relation to human perception. Color perception is quantified using statistics of color distribution, elementary hues, color properties, and relationships between colors. Further, we compute statistics of contour geometry and relationships. The main contribution of our work lies in harnessing the descriptive power of the combination of these descriptors in genre classification. Validation was carried out on over 91 h of video footage encompassing 7 common video genres, yielding average precision and recall ratios of 87% to 100% and 77% to 100%, respectively, and an overall average correct classification of up to 97%. Also, experimental comparison as part of the MediaEval 2011 benchmarking campaign demonstrated the efficiency of the proposed audio-visual descriptors over other existing approaches. Finally, we discuss a 3-D video browsing platform that displays movies using feature-based coordinates and thus regroups them according to genre.

  13. 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.

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

    Science.gov (United States)

    Pakarinen, Jyri

    2010-12-01

    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.

  15. 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.

  16. 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.

  17. Effect of continuous speech and non-speech signals on stuttering frequency in adults who stutter.

    Science.gov (United States)

    Dayalu, Vikram N; Guntupalli, Vijaya K; Kalinowski, Joseph; Stuart, Andrew; Saltuklaroglu, Tim; Rastatter, Michael P

    2011-10-01

    The inhibitory effects of continuously presented audio signals (/a/, /s/, 1,000 Hz pure-tone) on stuttering were examined. Eleven adults who stutter participated. Participants read four 300-syllable passages (i.e. in the presence and absence of the audio signals). All of the audio signals induced a significant reduction in stuttering frequency relative to the control condition (P = 0.005). A significantly greater reduction in stuttering occurred in the /a/ condition (P 0.05). These findings are consistent with the notion that the percept of a second signal as speech or non-speech can respectively augment or attenuate the potency for reducing stuttering frequency.

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

    Science.gov (United States)

    Dickinson Public Schools, ND. Instructional Media Center.

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

  19. Audio steganography by amplitude or phase modification

    Science.gov (United States)

    Gopalan, Kaliappan; Wenndt, Stanley J.; Adams, Scott F.; Haddad, Darren M.

    2003-06-01

    This paper presents the results of embedding short covert message utterances on a host, or cover, utterance by modifying the phase or amplitude of perceptually masked or significant regions of the host. In the first method, the absolute phase at selected, perceptually masked frequency indices was changed to fixed, covert data-dependent values. Embedded bits were retrieved at the receiver from the phase at the selected frequency indices. Tests on embedding a GSM-coded covert utterance on clean and noisy host utterances showed no noticeable difference in the stego compared to the hosts in speech quality or spectrogram. A bit error rate of 2 out of 2800 was observed for a clean host utterance while no error occurred for a noisy host. In the second method, the absolute phase of 10 or fewer perceptually significant points in the host was set in accordance with covert data. This resulted in a stego with successful data retrieval and a slightly noticeable degradation in speech quality. Modifying the amplitude of perceptually significant points caused perceptible differences in the stego even with small changes of amplitude made at five points per frame. Finally, the stego obtained by altering the amplitude at perceptually masked points showed barely noticeable differences and excellent data recovery.

  20. 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"--

  1. Video equipment of tele dosimetry and audio; Video equipo de teledosimetria y audio

    Energy Technology Data Exchange (ETDEWEB)

    Ojeda R, M.A.; Padilla C, I. [CFE, Central Laguna Verde, Subgerencia General de Operacion, Proteccion Radiologica, Veracruz (Mexico)]. e-mail: aojega@cfe.gob.mx

    2007-07-01

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

  2. 一种抗DA/AD转换攻击的音频信息隐藏算法%An Audio Information Hiding Algorithm Against Attacks from D/A and A/D Conversions

    Institute of Scientific and Technical Information of China (English)

    郑玉婷; 梁猛; 曹雅萍; 王春河; 刘继红

    2016-01-01

    在经历播放和转录的音频信息隐藏应用系统中,DA/AD变换引起的攻击是限制其性能的主要因素之一。通过分析DA/AD转换过程中音频信号线性伸缩、文件大小变化引起的失同步,提出了一种基于DCT域逐项定位的音频信息隐藏算法,实验结果表明,该算法能够有效抵抗DA/AD转换过程中的攻击,具有良好的鲁棒性。%A DA/AD transform attack resistance of audio information hiding is to pick a transcribed to the digital audio broadcast after complex DA/AD conversion process, general audio information hiding algorithm is difficult to resist the process of attack, and poor practicability. Paper analyzed before and after the audio DA/AD conversion impact and put forward relevant solutions, is presented based on the positioning of the audio frequency DCT domain audio information hiding algorithm was proposed. The experimental results show that the algorithm can effectively resist attacks from DA/AD conversion, with strong robustness.

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

    Science.gov (United States)

    Rui, Rui; Bao, Chang-chun

    2013-03-01

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

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

    Science.gov (United States)

    Barton, Robert J., III; Chodura, Hartmut

    2000-07-01

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

  5. An 18-bit high performance audio σ-Δ D/A converter

    Science.gov (United States)

    Hao, Zhang; Xiaowei, Huang; Yan, Han; Cheung, Ray C.; Xiaoxia, Han; Hao, Wang; Guo, Liang

    2010-07-01

    A multi-bit quantized high performance sigma-delta (σ-Δ) audio DAC is presented. Compared to its single-bit counterpart, the multi-bit quantization offers many advantages, such as simpler σ-Δ modulator circuit, lower clock frequency and smaller spurious tones. With the data weighted average (DWA) mismatch shaping algorithm, element mismatch errors induced by multi-bit quantization can be pushed out of the signal band, hence the noise floor inside the signal band is greatly lowered. To cope with the crosstalk between digital and analog circuits, every analog component is surrounded by a guard ring, which is an innovative attempt. The 18-bit DAC with the above techniques, which is implemented in a 0.18 μm mixed-signal CMOS process, occupies a core area of 1.86 mm2. The measured dynamic range (DR) and peak SNDR are 96 dB and 88 dB, respectively.

  6. An 18-bit high performance audio {sigma}-{Delta} D/A converter

    Energy Technology Data Exchange (ETDEWEB)

    Zhang Hao; Han Yan; Han Xiaoxia; Wang Hao; Liang Guo [Institute of Microelectronics and Photoelectronics, Zhejiang University, Hangzhou 310027 (China); Huang Xiaowei [CISD, Institute of Microelectronic CAD, Hangzhou 310018 (China); Cheung, Ray C., E-mail: huangxw@hdu.edu.c [Department of Electronic Engineering, City University of Hong Kong (Hong Kong)

    2010-07-15

    A multi-bit quantized high performance sigma-delta ({sigma}-{Delta}) audio DAC is presented. Compared to its single-bit counterpart, the multi-bit quantization offers many advantages, such as simpler {sigma}-{Delta} modulator circuit, lower clock frequency and smaller spurious tones. With the data weighted average (DWA) mismatch shaping algorithm, element mismatch errors induced by multi-bit quantization can be pushed out of the signal band, hence the noise floor inside the signal band is greatly lowered. To cope with the crosstalk between digital and analog circuits, every analog component is surrounded by a guard ring, which is an innovative attempt. The 18-bit DAC with the above techniques, which is implemented in a 0.18 {mu}m mixed-signal CMOS process, occupies a core area of 1.86 mm{sup 2}. The measured dynamic range (DR) and peak SNDR are 96 dB and 88 dB, respectively.

  7. Investigation of Nanosecond Pulsed Discharge and Its Audio Characteristics in Atmospheric-pressure Air

    Institute of Scientific and Technical Information of China (English)

    REN Chengyan; RAN Huijuan; WANG Jue; WANG Tao; YAN Ping

    2013-01-01

    There was no well-resolved mechanism of audible noise caused by corona discharge on UHV transmission lines.Hence we measured the sound pressure of pulsed discharges between needle-plane electrodes under different discharge conditions in air,for revealing the intrinsic relationship between discharge and its audible noise(AN).The relationship between discharge parameters and audio characteristics was drawn from the analysis of the electric and sound signals obtained in experiments.Experiment results showed that nanosecond pulsed discharges produce the sound pressure with a microsecond pulse lagging behind the discharge pulse in their waveforms.The peak value of the sound pulse decreases and its high frequency component gradually attenuates,when the measuring distance from discharges increases.The sound pulses correlate with the discharge current and voltage significantly,especially the current.The audible noise produced by repetitive pulsed discharge increases with the strength,duration,and pulse repetition rate of discharge.

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

    Science.gov (United States)

    Jia, Yuan; Liming, Chen; Zenghui, Yu; Yong, Hei

    2014-07-01

    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.

  9. 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.

  10. Digital signal processing techniques for pitch shifting and time scaling of audio signals

    Science.gov (United States)

    Buś, Szymon; Jedrzejewski, Konrad

    2016-09-01

    In this paper, we present the techniques used for modifying the spectral content (pitch shifting) and for changing the time duration (time scaling) of an audio signal. A short introduction gives a necessary background for understanding the discussed issues and contains explanations of the terms used in the paper. In subsequent sections we present three different techniques appropriate both for pitch shifting and for time scaling. These techniques use three different time-frequency representations of a signal, namely short-time Fourier transform (STFT), continuous wavelet transform (CWT) and constant-Q transform (CQT). The results of simulation studies devoted to comparison of the properties of these methods are presented and discussed in the paper.

  11. A comparative study of transforms for use in digital audio data compression.

    Science.gov (United States)

    Rulon, B E; Shaw, M F; Donohue, K D

    2000-07-01

    This paper provides a subjective quality analysis of transforms used in audio compression algorithms for a class of music signals. A 34-subject listener test compares three transforms in conjunction with an MPEG I layer 1 compression scheme. One test compares the performances of the discrete wavelet packet transform (DWPT) and the modified discrete cosine transform (MDCT) used in MPEG. Another test compares the performances of a DWPT eight-level nonuniform critical-band split and a DWPT five-level uniform subband split. Results indicate that the critical-band split provides significantly better quality than the uniform subband split for sounds with tonal and strong low-frequency content, while the DWPT outperforms the MDCT with significant improvement for nontonal sounds.

  12. 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.

  13. 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.

  14. “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.

  15. Evaluation of embedded audio feedback on writing assignments.

    Science.gov (United States)

    Graves, Janet K; Goodman, Joely T; Hercinger, Maribeth; Minnich, Margo; Murcek, Christina M; Parks, Jane M; Shirley, Nancy

    2015-01-01

    The purpose of this pilot study was to compare embedded audio feedback (EAF), which faculty provided using the iPad(®) application iAnnotate(®) PDF to insert audio comments and written feedback (WF), inserted electronically on student papers in a series of writing assignments. Goals included determining whether EAF provides more useful guidance to students than WF and whether EAF promotes connectedness among students and faculty. An additional goal was to ascertain the efficiency and acceptance of EAF as a grading tool by nursing faculty. The pilot study was a quasi-experimental, cross-over, posttest-only design. The project was completed in an Informatics in Health Care course. Faculty alternated the two feedback methods on four papers written by each student. Results of surveys and focus groups revealed that students and faculty had mixed feelings about this technology. Student preferences were equally divided between EAF and WF, with 35% for each, and 28% were undecided.

  16. Audio-visual interactions in product sound design

    Science.gov (United States)

    Özcan, Elif; van Egmond, René

    2010-02-01

    Consistent product experience requires congruity between product properties such as visual appearance and sound. Therefore, for designing appropriate product sounds by manipulating their spectral-temporal structure, product sounds should preferably not be considered in isolation but as an integral part of the main product concept. Because visual aspects of a product are considered to dominate the communication of the desired product concept, sound is usually expected to fit the visual character of a product. We argue that this can be accomplished successfully only on basis of a thorough understanding of the impact of audio-visual interactions on product sounds. Two experimental studies are reviewed to show audio-visual interactions on both perceptual and cognitive levels influencing the way people encode, recall, and attribute meaning to product sounds. Implications for sound design are discussed defying the natural tendency of product designers to analyze the "sound problem" in isolation from the other product properties.

  17. A Robust Zero-Watermarking Algorithm for Audio

    Directory of Open Access Journals (Sweden)

    Jie Zhu

    2008-03-01

    Full Text Available In traditional watermarking algorithms, the insertion of watermark into the host signal inevitably introduces some perceptible quality degradation. Another problem is the inherent conflict between imperceptibility and robustness. Zero-watermarking technique can solve these problems successfully. Instead of embedding watermark, the zero-watermarking technique extracts some essential characteristics from the host signal and uses them for watermark detection. However, most of the available zero-watermarking schemes are designed for still image and their robustness is not satisfactory. In this paper, an efficient and robust zero-watermarking technique for audio signal is presented. The multiresolution characteristic of discrete wavelet transform (DWT, the energy compression characteristic of discrete cosine transform (DCT, and the Gaussian noise suppression property of higher-order cumulant are combined to extract essential features from the host audio signal and they are then used for watermark recovery. Simulation results demonstrate the effectiveness of our scheme in terms of inaudibility, detection reliability, and robustness.

  18. Dynamic range control of audio signals by digital signal processing

    Science.gov (United States)

    Gilchrist, N. H. C.

    It is often necessary to reduce the dynamic range of musical programs, particularly those comprising orchestral and choral music, for them to be received satisfactorily by listeners to conventional FM and AM broadcasts. With the arrival of DAB (Digital Audio Broadcasting) a much wider dynamic range will become available for radio broadcasting, although some listeners may prefer to have a signal with a reduced dynamic range. This report describes a digital processor developed by the BBC to control the dynamic range of musical programs in a manner similar to that of a trained Studio Manager. It may be used prior to transmission in conventional broadcasting, replacing limiters or other compression equipment. In DAB, it offers the possibility of providing a dynamic range control signal to be sent to the receiver via an ancillary data channel, simultaneously with the uncompressed audio, giving the listener the option of the full dynamic range or a reduced dynamic range.

  19. 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.

  20. 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 research and annotated. They have been applied in a migration scenario, where radio broadcasts are to be migrated for long term preservation....

  1. Audio Signal Generator System Based On State Machines

    Institute of Scientific and Technical Information of China (English)

    王维喜

    2009-01-01

    A state machine can make program designing quicker, simpler and more efficient. This paper describes in detail the model for a state machine and the idea for its designing and gives the design process of the state machine through an example of audio signal generator system based on Labview. The result shows that the introduction of the state machine can make complex design processes more clear and the revision of programs easier.

  2. 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-squar......-squares minimization. Rate-distortion curves and listening tests show that, compared to a constant-amplitude sinusoidal coder, the proposed decomposition offers perceptually significant improvements in critical transient signals....

  3. Quality and Distortion Evaluation of Audio Signal by Spectrum

    OpenAIRE

    Er. Niranjan Singh; Dr. Bhupendra Verma

    2012-01-01

    Information hiding in digital audio can be used for such diverse applications as proof ofownership, authentication, integrity, secret communication, broadcast monitoring and eventannotation. To achieve secure and undetectable communication, stegano-objects, anddocuments containing a secret message, should be indistinguishable from cover-objects, andshow that documents not containing any secret message. In this respect, Steganalysis is the setof techniques that aim to distinguish between cover...

  4. 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.

  5. High capacity reversible watermarking for audio by histogram shifting and predicted error expansion.

    Science.gov (United States)

    Wang, Fei; Xie, Zhaoxin; Chen, Zuo

    2014-01-01

    Being reversible, the watermarking information embedded in audio signals can be extracted while the original audio data can achieve lossless recovery. Currently, the few reversible audio watermarking algorithms are confronted with following problems: relatively low SNR (signal-to-noise) of embedded audio; a large amount of auxiliary embedded location information; and the absence of accurate capacity control capability. In this paper, we present a novel reversible audio watermarking scheme based on improved prediction error expansion and histogram shifting. First, we use differential evolution algorithm to optimize prediction coefficients and then apply prediction error expansion to output stego data. Second, in order to reduce location map bits length, we introduced histogram shifting scheme. Meanwhile, the prediction error modification threshold according to a given embedding capacity can be computed by our proposed scheme. Experiments show that this algorithm improves the SNR of embedded audio signals and embedding capacity, drastically reduces location map bits length, and enhances capacity control capability.

  6. A Detailed look of Audio Steganography Techniques using LSB and Genetic Algorithm Approach

    Directory of Open Access Journals (Sweden)

    Gunjan Nehru

    2012-01-01

    Full Text Available This paper is the study of various techniques of audio steganography using different algorithmis like genetic algorithm approach and LSB approach. We have tried some approaches that helps in audio steganography. As we know it is the art and science of writing hidden messages in such a way that no one, apart from the sender and intended recipient, suspects the existence of the message, a form of security through obscurity. In steganography, the message used to hide secret message is called host message or cover message. Once the contents of the host message or cover message are modified, the resultant message is known as stego message. In other words, stego message is combination of host message and secret message. Audio steganography requires a text or audio secret message to be embedded within a cover audio message. Due to availability of redundancy, the cover audio message before steganography, stego message after steganography remains same. for information hiding.

  7. Efficient Query-by-Content Audio Retrieval by Locality Sensitive Hashing and Partial Sequence Comparison

    Science.gov (United States)

    Yu, Yi; Joe, Kazuki; Downie, J. Stephen

    This paper investigates suitable indexing techniques to enable efficient content-based audio retrieval in large acoustic databases. To make an index-based retrieval mechanism applicable to audio content, we investigate the design of Locality Sensitive Hashing (LSH) and the partial sequence comparison. We propose a fast and efficient audio retrieval framework of query-by-content and develop an audio retrieval system. Based on this framework, four different audio retrieval schemes, LSH-Dynamic Programming (DP), LSH-Sparse DP (SDP), Exact Euclidian LSH (E2LSH)-DP, E2LSH-SDP, are introduced and evaluated in order to better understand the performance of audio retrieval algorithms. The experimental results indicate that compared with the traditional DP and the other three compititive schemes, E2LSH-SDP exhibits the best tradeoff in terms of the response time, retrieval accuracy and computation cost.

  8. High Capacity Reversible Watermarking for Audio by Histogram Shifting and Predicted Error Expansion

    Directory of Open Access Journals (Sweden)

    Fei Wang

    2014-01-01

    Full Text Available Being reversible, the watermarking information embedded in audio signals can be extracted while the original audio data can achieve lossless recovery. Currently, the few reversible audio watermarking algorithms are confronted with following problems: relatively low SNR (signal-to-noise of embedded audio; a large amount of auxiliary embedded location information; and the absence of accurate capacity control capability. In this paper, we present a novel reversible audio watermarking scheme based on improved prediction error expansion and histogram shifting. First, we use differential evolution algorithm to optimize prediction coefficients and then apply prediction error expansion to output stego data. Second, in order to reduce location map bits length, we introduced histogram shifting scheme. Meanwhile, the prediction error modification threshold according to a given embedding capacity can be computed by our proposed scheme. Experiments show that this algorithm improves the SNR of embedded audio signals and embedding capacity, drastically reduces location map bits length, and enhances capacity control capability.

  9. 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.

  10. Design and Research on Sigma-Delta Digital-to-Analog Converters for Audio Power Amplifiers

    OpenAIRE

    Puidokas, Vytenis

    2011-01-01

    The dissertation investigates the issues of analyzing a digital Sigma-Delta digital-to-analog converter (DAC) for audio power amplifiers. The main objects of research include a digital Sigma-Delta audio power DAC, improvement of its structure and an experimental research. The primary purpose of the dissertation is to suggest methods for improvement the structure of digital Sigma-Delta audio power DAC interpolator and the converter analysis. Disertacijoje nagrinėjami Sigma-Delta skaitmenini...

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

    DEFF Research Database (Denmark)

    Ljusev, Petar; Andersen, Michael Andreas E.

    2005-01-01

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

  12. 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......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...

  13. Acoustic contrast sensitivity to transfer function errors in the design of a personal audio system.

    Science.gov (United States)

    Park, Jin-Young; Choi, Jung-Woo; Kim, Yang-Hann

    2013-07-01

    An analytic means to evaluate the error sensitivity of a personal audio system is proposed. The personal audio system, which focuses acoustic energy into a zone of interest using multiple loudspeakers, is subject to various errors when implemented. The performance of a personal audio system, defined as an energy ratio between the zone of interest and the rest, is inevitably influenced by errors. Thus the ability to predict performance change at the design stage is crucial when building a robust personal audio system. The dependence of the energy ratio change on various types of errors is formulated.

  14. 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.

  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. Sinusoidal Analysis-Synthesis of Audio Using Perceptual Criteria

    Directory of Open Access Journals (Sweden)

    Ted Painter

    2003-01-01

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

  17. 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

  18. 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.

  19. Audio Hijack Pro万能录音机

    Institute of Scientific and Technical Information of China (English)

    2004-01-01

    Audio Hijack Pro是由Rogue amoeba开发的音频软件,它的功能非常强大只要是你的Mac能放的声音。这个程序都可以录下来.从流媒体广播到DVD音频.还可以为任何程序作数字声效处理,可以使iTunes和Quicktime电台效果明显改善。

  20. Differences between the Audio-lingual Methodand the Communicative Approach

    Institute of Scientific and Technical Information of China (English)

    涂艳; 刘俊

    2016-01-01

    There are some differences between the two kinds of foreign language teaching methods .The Audio-lingual Method can help students gain control over grammatical structures as well as develop their oral ability, and the teaching focus is often on forms rather than functions, so students have learned a lot of structures or patterns without knowing how to use them appropriately in real situations. While the aim of the Communicative Approach is to develop student's communicative competence, which includes both the knowledge about the language and the knowledge about how to use the language appropriately in communication situations.

  1. Audio-haptic interaction in simulated walking experiences

    DEFF Research Database (Denmark)

    Serafin, Stefania

    2011-01-01

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

  2. 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

  3. A listening test system for automotive audio - listeners

    DEFF Research Database (Denmark)

    Choisel, Sylvain; Hegarty, Patrick; Christensen, Flemming;

    2007-01-01

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

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

    Energy Technology Data Exchange (ETDEWEB)

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

    2008-07-15

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

  5. Anthropomorphic Coding of Speech and Audio: A Model Inversion Approach

    Directory of Open Access Journals (Sweden)

    W. Bastiaan Kleijn

    2005-06-01

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

  6. 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; ·     �...

  7. Information-Driven Active Audio-Visual Source Localization.

    Directory of Open Access Journals (Sweden)

    Niclas Schult

    Full Text Available We present a system for sensorimotor audio-visual source localization on a mobile robot. We utilize a particle filter for the combination of audio-visual information and for the temporal integration of consecutive measurements. Although the system only measures the current direction of the source, the position of the source can be estimated because the robot is able to move and can therefore obtain measurements from different directions. These actions by the robot successively reduce uncertainty about the source's position. An information gain mechanism is used for selecting the most informative actions in order to minimize the number of actions required to achieve accurate and precise position estimates in azimuth and distance. We show that this mechanism is an efficient solution to the action selection problem for source localization, and that it is able to produce precise position estimates despite simplified unisensory preprocessing. Because of the robot's mobility, this approach is suitable for use in complex and cluttered environments. We present qualitative and quantitative results of the system's performance and discuss possible areas of application.

  8. Audio Editing Skills%音频剪辑技巧

    Institute of Scientific and Technical Information of China (English)

    范炜; 杨澍彬; 谭忠凯

    2015-01-01

    随着近年来影视剧的蓬勃发展,各个相关领域也由以前的冷门慢慢变得越来越受到重视,有必要进行专门的研究,以便更好地为影视剧制作进行服务。根据多年的影视剧制作经验,通过分析一些精彩影视剧中的音频制作技巧,来阐述音频剪辑在影视剧制作中的重要性。%As the vigorous development of the film and television drama develop vigorously in recent years,va-rious related fields also by previous unpopular slowly become more and more attention,it is necessary to carry out specialized research to server better for the TV drama making service.According to many years of industry experience,the author of this paper is to elaborate the importance of audio clip in the production of television drama through analyzing the audio production skills of some wonderful film and television drama.

  9. Head Tracking of Auditory, Visual, and Audio-Visual Targets.

    Science.gov (United States)

    Leung, Johahn; Wei, Vincent; Burgess, Martin; Carlile, Simon

    2015-01-01

    The ability to actively follow a moving auditory target with our heads remains unexplored even though it is a common behavioral response. Previous studies of auditory motion perception have focused on the condition where the subjects are passive. The current study examined head tracking behavior to a moving auditory target along a horizontal 100° arc in the frontal hemisphere, with velocities ranging from 20 to 110°/s. By integrating high fidelity virtual auditory space with a high-speed visual presentation we compared tracking responses of auditory targets against visual-only and audio-visual "bisensory" stimuli. Three metrics were measured-onset, RMS, and gain error. The results showed that tracking accuracy (RMS error) varied linearly with target velocity, with a significantly higher rate in audition. Also, when the target moved faster than 80°/s, onset and RMS error were significantly worst in audition the other modalities while responses in the visual and bisensory conditions were statistically identical for all metrics measured. Lastly, audio-visual facilitation was not observed when tracking bisensory targets.

  10. Head Tracking of Auditory, Visual and Audio-Visual Targets

    Directory of Open Access Journals (Sweden)

    Johahn eLeung

    2016-01-01

    Full Text Available The ability to actively follow a moving auditory target with our heads remains unexplored even though it is a common behavioral response. Previous studies of auditory motion perception have focused on the condition where the subjects are passive. The current study examined head tracking behavior to a moving auditory target along a horizontal 100° arc in the frontal hemisphere, with velocities ranging from 20°/s to 110°/s. By integrating high fidelity virtual auditory space with a high-speed visual presentation we compared tracking responses of auditory targets against visual-only and audio-visual bisensory stimuli. Three metrics were measured – onset, RMS and gain error. The results showed that tracking accuracy (RMS error varied linearly with target velocity, with a significantly higher rate in audition. Also, when the target moved faster than 80°/s, onset and RMS error were significantly worst in audition the other modalities while responses in the visual and bisensory conditions were statistically identical for all metrics measured. Lastly, audio-visual facilitation was not observed when tracking bisensory targets.

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

    Directory of Open Access Journals (Sweden)

    Anja Kuchenbuch

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

  12. RESEARCH ANNOUNCEMENTS A Uniform Approximation Model for Instantaneous Frequency Analysis

    Institute of Scientific and Technical Information of China (English)

    李应岐; 武瑛; 韩国栋

    2012-01-01

    0 Introduction Signal decomposition is a crucial issue and a key step in signal analysis,which provides the multiscale components for further processing.In many applications related to image and speech/audio processing,it is always necessary to decompose an input signal to extract information such as amplitudes,phases and frequencies.The classical decomposition methods include

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

    DEFF Research Database (Denmark)

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

    2012-01-01

    and presented in either mono, stereo, 3D, or interactive 3D, and performance was evaluated by asking factual questions about details in the audio. Results show that spatial cues can increase attention to background sounds while reducing attention to narrated text, indicating that spatial audio can...

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

    Science.gov (United States)

    Branje, Carmen J.; Fels, Deborah I.

    2012-01-01

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

  15. 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…

  16. 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

  17. 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 ...

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

    DEFF Research Database (Denmark)

    Gelineck, Steven; Korsgaard, Dannie Michael

    2015-01-01

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

  19. 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.

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

    Science.gov (United States)

    Kim, Yong-Jin; Chang, Nam-Kee

    2001-01-01

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

  1. 47 CFR 73.4275 - Tone clusters; audio attention-getting devices.

    Science.gov (United States)

    2010-10-01

    ... 47 Telecommunication 4 2010-10-01 2010-10-01 false Tone clusters; audio attention-getting devices. 73.4275 Section 73.4275 Telecommunication FEDERAL COMMUNICATIONS COMMISSION (CONTINUED) BROADCAST... clusters; audio attention-getting devices. See Public Notice, FCC 76-610, dated July 2, 1976. 60 FCC 2d...

  2. Audio Feedback: Richer Language but No Measurable Impact on Student Performance

    Science.gov (United States)

    Chalmers, Charlotte; MacCallum, Janis; Mowat, Elaine; Fulton, Norma

    2014-01-01

    Audio feedback has been shown to be popular and well received by students. However, there is little published work to indicate how effective audio feedback is in improving student performance. Sixty students from a first year science degree agreed to take part in the study; thirty were randomly assigned to receive written feedback on coursework,…

  3. Audio-video decision support for patients: the documentary genre as a basis for decision aids

    NARCIS (Netherlands)

    Volandes, A.E.; Barry, M.J.; Wood, F.; Elwyn, G.

    2013-01-01

    Objective Decision support tools are increasingly using audio-visual materials. However, disagreement exists about the use of audio-visual materials as they may be subjective and biased. Methods This is a literature review of the major texts for documentary film studies to extrapolate issues of obje

  4. A Management Review and Analysis of Purdue University Libraries and Audio-Visual Center.

    Science.gov (United States)

    Baaske, Jan; And Others

    A management review and analysis was conducted by the staff of the libraries and audio-visual center of Purdue University. Not only were the study team and the eight task forces drawn from all levels of the libraries and audio-visual center staff, but a systematic effort was sustained through inquiries, draft reports and open meetings to involve…

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

    DEFF Research Database (Denmark)

    Turchet, Luca; Serafin, Stefania; Dimitrov, Smilen

    2010-01-01

    We describe an audio-haptic experiment conducted using a system which simulates in real-time the auditory and haptic sensation of walking on different surfaces. The system is based on physical models, that drive both the haptic and audio synthesizers, and a pair of shoes enhanced with sensors...

  6. 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...

  7. 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.

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

    Science.gov (United States)

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

    2001-05-01

    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 data from a recent clinic-based experiment indicating that the touch audio-CASI system is stable, robust, and suitable for administering relatively long and complex questionnaires on sensitive topics, including drug use and sexual behaviors associated with HIV and other sexually transmitted diseases.

  9. 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.

  10. An UHF frequency-modulated continuous wave wind profiler - receiver and audio module development

    OpenAIRE

    Garrido López, David

    2010-01-01

    Projecte final de carrera fet en col.laboració amb University of Massachusetts - Amherst, the Microwave Sensing Laboratory (MIRSL) The measurement of winds and processes taking place in the atmosphere is a fun- damental requirement in both research and operational meteorology. This project is focused on the processes taking place in the lower troposphere called the atmospheric boundary layer (ABL). The ABL is important meteorologically in terms of assessing of convective in...

  11. 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.

  12. 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

    This paper introduces the concept of affective offset, which is the difference between a user's perceived affective state and the affective annotation of the content they wish to see. We show how this affective offset can be used within a framework for providing recommendations for TV programs....... 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...

  13. Real Time Recognition Of Speakers From Internet Audio Stream

    Directory of Open Access Journals (Sweden)

    Weychan Radoslaw

    2015-09-01

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

  14. 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.

  15. Detection of vibrations in the audio range using photorefractive polymers

    Science.gov (United States)

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

    2006-08-01

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

  16. Audio collection in the SASA Institute of Musicology

    Directory of Open Access Journals (Sweden)

    Lajić-Mihajlović Danka

    2010-01-01

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

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

    DEFF Research Database (Denmark)

    Gelineck, Steven; Overholt, Daniel

    2015-01-01

    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......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...... visual feedback with interfaces such as the Novint Falcon and rely primarily on haptic cues, allowing more focus on the spatial sound elements. Results of the evaluation support this claim, as users preferred the interaction paradigm using the Falcon with no visual feedback. Furthermore, users disliked...

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

    Energy Technology Data Exchange (ETDEWEB)

    Kierkegaard, P.

    1998-10-01

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

  19. 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.

  20. The audio-visual revolution: do we really need it?

    Science.gov (United States)

    Townsend, I

    1979-03-01

    In the United Kingdom, The audio-visual revolution has steadily gained converts in the nursing profession. Nurse tutor courses now contain information on the techniques of educational technology and schools of nursing increasingly own (or wish to own) many of the sophisticated electronic aids to teaching that abound. This is taking place at a time of hitherto inexperienced crisis and change. Funds have been or are being made available to buy audio-visual equipment. But its purchase and use relies on satisfying personal whim, prejudice or educational fashion, not on considerations of educational efficiency. In the rush of enthusiasm, the overwhelmed teacher (everywhere; the phenomenon is not confined to nursing) forgets to ask the searching, critical questions: 'Why should we use this aid?','How effective is it?','And, at what?'. Influential writers in this profession have repeatedly called for a more responsible attitude towards published research work of other fields. In an attempt to discover what is known about the answers to this group of questions, an eclectic look at media research is taken and the widespread dissatisfaction existing amongst international educational technologists is noted. The paper isolates out of the literature several causative factors responsible for the present state of affairs. Findings from the field of educational television are cited as representative of an aid which has had a considerable amount of time and research directed at it. The concluding part of the paper shows the decisions to be taken in using or not using educational media as being more complicated than might at first appear.

  1. APPLICATION OF PARTIAL LEAST SQUARES REGRESSION FOR AUDIO-VISUAL SPEECH PROCESSING AND MODELING

    Directory of Open Access Journals (Sweden)

    A. L. Oleinik

    2015-09-01

    Full Text Available Subject of Research. The paper deals with the problem of lip region image reconstruction from speech signal by means of Partial Least Squares regression. Such problems arise in connection with development of audio-visual speech processing methods. Audio-visual speech consists of acoustic and visual components (called modalities. Applications of audio-visual speech processing methods include joint modeling of voice and lips’ movement dynamics, synchronization of audio and video streams, emotion recognition, liveness detection. Method. Partial Least Squares regression was applied to solve the posed problem. This method extracts components of initial data with high covariance. These components are used to build regression model. Advantage of this approach lies in the possibility of achieving two goals: identification of latent interrelations between initial data components (e.g. speech signal and lip region image and approximation of initial data component as a function of another one. Main Results. Experimental research on reconstruction of lip region images from speech signal was carried out on VidTIMIT audio-visual speech database. Results of the experiment showed that Partial Least Squares regression is capable of solving reconstruction problem. Practical Significance. Obtained findings give the possibility to assert that Partial Least Squares regression is successfully applicable for solution of vast variety of audio-visual speech processing problems: from synchronization of audio and video streams to liveness detection.

  2. 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.  

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

    Science.gov (United States)

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

    2012-01-01

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

  4. 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.

  5. 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.

  6. The temporal window of audio-tactile integration in speech perception

    Science.gov (United States)

    Gick, Bryan; Ikegami, Yoko; Derrick, Donald

    2010-01-01

    Asynchronous cross-modal information is integrated asymmetrically in audio-visual perception. To test whether this asymmetry generalizes across modalities, auditory (aspirated “pa” and unaspirated “ba” stops) and tactile (slight, inaudible, cutaneous air puffs) signals were presented synchronously and asynchronously. Results were similar to previous AV studies: the temporal window of integration for the enhancement effect (but not the interference effect) was asymmetrical, allowing up to 200 ms of asynchrony when the puff followed the audio signal, but only up to 50 ms when the puff preceded the audio signal. These findings suggest that perceivers accommodate differences in physical transmission speed of different multimodal signals. PMID:21110549

  7. Steganography: Applying and Evaluating Two Algorithms for Embedding Audio Data in an Image

    Directory of Open Access Journals (Sweden)

    Khaled Nasser ElSayed

    2015-03-01

    Full Text Available Information transmission is increasing with grow of using WEB. So, information security has become very important. Security of data and information is the major task for scientists and political and military people. One of the most secure methods is embedding data (steganography in different media like text, audio, digital images. this paper present two experiments in steganography of digital audio data file. It applies empirically, two algorithms in steganography in images through random insertion of digital audio data using bytes and pixels in image files. Finally, it evaluates both experiments, in order to enhance security of transmitted data.

  8. 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.

  9. Self-oscillating modulators for direct energy conversion audio power amplifiers

    Energy Technology Data Exchange (ETDEWEB)

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

    2005-07-01

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

  10. 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)

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

    DEFF Research Database (Denmark)

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

    2015-01-01

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

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

    Science.gov (United States)

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

    2012-01-01

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

  13. Design and implementation of a two-way real-time communication system for audio over CATV networks

    Science.gov (United States)

    Cho, Choong Sang; Oh, Yoo Rhee; Lee, Young Han; Kim, Hong Kook

    2007-09-01

    In this paper, we design and implement a two-way real-time communication system for audio over cable television (CATV) networks to provide an audio-based interaction between the CATV broadcasting station and CATV subscribers. The two-way real-time communication system consists of a real-time audio encoding/decoding module, a payload formatter based on a transmission control protocol/Internet protocol (TCP/IP), and a cable network. At the broadcasting station, audio signals from a microphone are encoded by an audio codec that is implemented using a digital signal processor (DSP), where the MPEG-2 Layer II audio codec is used for the audio codec and TMS320C6416 is used for a DSP. Next, a payload formatter constructs a TCP/IP packet from an audio bitstream for transmission to a cable modem. Another payload formatter at the subscriber unpacks the TCP/IP packet decoded from the cable modem into audio bitstream. This bitstream is decoded by the MPEG-2 Layer II audio decoder. Finally the decoded audio signals are played out to the speaker. We confirmed that the system worked in real-time, with a measured delay of around 150 ms including the algorithmic and processing time delays.

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

    Directory of Open Access Journals (Sweden)

    Soraya Ferreira Alves

    2016-12-01

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

  15. 脉冲激光与电刺激治疗系统音频信号处理模块的研制%Design of Pulse Laser and Electrotherapy System Audio Signal Processing Module

    Institute of Scientific and Technical Information of China (English)

    黄俊杰; 黄时俊; 黄丹阳; 陈仲本

    2012-01-01

    目的:设计一款基于TMS320VC5402 DSP的音频信号处理模块,用于采集处理多模式脉冲激光与电刺激治疗系统的音乐信号,探讨不同频率成分的音频信号对治疗高血压的影响,实现多模式脉冲激光与电刺激治疗高血压治疗处方的多样化.方法:使用音频编解码芯片TLV320AIC23B实现对多模式脉冲激光与电刺激治疗系统的音乐信号的采集,利用高性能数字信号处理芯片TMS320VC5402对采集的信号进行相应的信号分析与处理.通过数字信号处理技术得到新的治疗处方应用于多模式脉冲激光与电刺激治疗系统,用于探讨不同频率成分的音频信号对治疗高血压的影响.结果:设计的硬件平台稳定可靠,可实时采集音频信号,可用于寻找对高血压治疗的有效频率成分.结论:该设计可实时采集音频信号,并运用各种信号处理的手段,产生不同的高血压治疗处方,为寻找有效治疗高血压的音频频率成分提供了可靠稳定的硬件平台.%Objective:To design an audio signal processing module based on DSP TMS320VC5402 used to sample and process audio signal from multi-mode pulse laser and electrotherapy system, discuss different frequency components of audio signal influence hypertension treatment and achieve hypertension treatment prescription for multi-mode pulse laser and electrotherapy system of diversification. Methods: We use audio codec TLV320AIC23B for sampling audio signal from multi-mode pulse laser and electrotherapy system, and use high performance digital signal processing chip TMS320VC5402 for analyzing and processing audio signal. Through digital signal processing technology to generate new treatment prescription applied in multi-mode pulse laser and electrotherapy system, is used to explore different frequency components of audio signal influence hypertension treatment. Results: The design of hardware platform is stable and reliable, which can sample real-time audio

  16. 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...

  17. 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.

  18. 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. 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...... 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....

  19. 17 CFR 232.304 - Graphic, image, audio and video material.

    Science.gov (United States)

    2010-04-01

    ... delivered to investors and others is deemed part of the electronic filing and subject to the civil liability..., image, audio or video material, they are not subject to the civil liability and anti-fraud provisions...

  20. 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.

  1. 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.

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

    Directory of Open Access Journals (Sweden)

    Iwano Koji

    2007-01-01

    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.

  3. Audio-Visual Integration Modifies Emotional Judgment in Music

    Directory of Open Access Journals (Sweden)

    Shen-Yuan Su

    2011-10-01

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

  4. Human performance measures for interactive haptic-audio-visual interfaces.

    Science.gov (United States)

    Jia, Dawei; Bhatti, Asim; Nahavandi, Saeid; Horan, Ben

    2013-01-01

    Virtual reality and simulation are becoming increasingly important in modern society and it is essential to improve our understanding of system usability and efficacy from the users' perspective. This paper introduces a novel evaluation method designed to assess human user capability when undertaking technical and procedural training using virtual training systems. The evaluation method falls under the user-centered design and evaluation paradigm and draws on theories of cognitive, skill-based and affective learning outcomes. The method focuses on user interaction with haptic-audio-visual interfaces and the complexities related to variability in users' performance, and the adoption and acceptance of the technologies. A large scale user study focusing on object assembly training tasks involving selecting, rotating, releasing, inserting, and manipulating three-dimensional objects was performed. The study demonstrated the advantages of the method in obtaining valuable multimodal information for accurate and comprehensive evaluation of virtual training system efficacy. The study investigated how well users learn, perform, adapt to, and perceive the virtual training. The results of the study revealed valuable aspects of the design and evaluation of virtual training systems contributing to an improved understanding of more usable virtual training systems.

  5. Person identification for mobile robot using audio-visual modality

    Science.gov (United States)

    Kim, Young-Ouk; Chin, Sehoon; Lee, Jihoon; Paik, Joonki

    2005-10-01

    Recently, we experienced significant advancement in intelligent service robots. The remarkable features of an intelligent robot include tracking and identification of person using biometric features. The human-robot interaction is very important because it is one of the final goals of an intelligent service robot. Many researches are concentrating in two fields. One is self navigation of a mobile robot and the other is human-robot interaction in natural environment. In this paper we will present an effective person identification method for HRI (Human Robot Interaction) using two different types of expert systems. However, most of mobile robots run under uncontrolled and complicated environment. It means that face and speech information can't be guaranteed under varying conditions, such as lighting, noisy sound, orientation of a robot. According to a value of illumination and signal to noise ratio around mobile a robot, our proposed fuzzy rule make a reasonable person identification result. Two embedded HMM (Hidden Marhov Model) are used for each visual and audio modality to identify person. The performance of our proposed system and experimental results are compared with single modality identification and simply mixed method of two modality.

  6. A Detailed look of Audio Steganography Techniques using LSB and Genetic Algorithm Approach

    OpenAIRE

    Gunjan Nehru; Puja Dhar

    2012-01-01

    This paper is the study of various techniques of audio steganography using different algorithmis like genetic algorithm approach and LSB approach. We have tried some approaches that helps in audio steganography. As we know it is the art and science of writing hidden messages in such a way that no one, apart from the sender and intended recipient, suspects the existence of the message, a form of security through obscurity. In steganography, the message used to hide secret message is called hos...

  7. Steganography: Applying and Evaluating Two Algorithms for Embedding Audio Data in an Image

    OpenAIRE

    Khaled Nasser ElSayed

    2015-01-01

    Information transmission is increasing with grow of using WEB. So, information security has become very important. Security of data and information is the major task for scientists and political and military people. One of the most secure methods is embedding data (steganography) in different media like text, audio, digital images. this paper present two experiments in steganography of digital audio data file. It applies empirically, two algorithms in steganography in images through random in...

  8. Integrating switch mode audio amplifiers and electro dynamic loudspeakers for a higher power efficiency

    DEFF Research Database (Denmark)

    Poulsen, Søren; Andersen, Michael Andreas E.

    2004-01-01

    The work presented in this paper is related to integration of switch mode audio amplifiers and electro dynamic loudspeakers, using the speaker's voice coil as output filter, and the magnetic structure as heatsink for the amplifier.......The work presented in this paper is related to integration of switch mode audio amplifiers and electro dynamic loudspeakers, using the speaker's voice coil as output filter, and the magnetic structure as heatsink for the amplifier....

  9. Integrating switch mode audio power amplifiers and electro dynamic loudspeakers for a higher power efficiency

    DEFF Research Database (Denmark)

    Poulsen, Søren; Andersen, Michael Andreas E.

    2004-01-01

    The work presented in this paper is related to integration of switch mode audio amplifiers and electro dynamic loudspeakers, using the speaker's voice coil as output filter, and the magnetic structure as heatsink for the amplifier.......The work presented in this paper is related to integration of switch mode audio amplifiers and electro dynamic loudspeakers, using the speaker's voice coil as output filter, and the magnetic structure as heatsink for the amplifier....

  10. An Audio Data Encryption with Single and Double Dimension Discrete-Time Chaotic Systems

    OpenAIRE

    AKGÜL, Akif; KAÇAR, Sezgin; Pehlivan, İhsan

    2015-01-01

    — In this article, a study on increasing security of audio data encryption with single and double dimension discrete-time chaotic systems was carried out and application and security analyses were executed. Audio data samples of both mono and stereo types were encrypted. In the application here, single and double dimension discrete-time chaotic systems were used. In order to enhance security during encryption, a different method was applied by also using a non-linear function. In the chaos ba...

  11. 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...

  12. 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.

  13. Reasons to rethink the use of audio and video lectures in online courses

    OpenAIRE

    Stetz, Thomas A.; Bauman, Antonina A.

    2013-01-01

    Recent technological developments allow any instructor to create audio and video lectures for the use in online classes. However, it is questionable if it is worth the time and effort that faculty put into preparing those lectures. This paper presents thirteen factors that should be considered before preparing and using audio and video lectures in online classes. In addition, recommendations for when and how to use lectures in online classes are presented. SIN FINANCIACIÓN Sin índice de...

  14. Development of the Audio Enhancement Method by Using the Reflected Signals in the Reverberant Environment

    OpenAIRE

    Shyang-Jye Chang; Hung-Wei Hsieh

    2015-01-01

    This paper presents a new audio signal enhancement method based on reflection signal detection in the reverberant environment. This technique enhances the audio signal from the target sound source in time domain. The advantages of this technique are enhancing the target sound source by a simple algorithm and reducing the background noise effectively. The effects of distances between the speaker and microphone and the coefficient of correlation are discussed in this paper. The coefficient of c...

  15. ASLP-MULAN: Audio speech and language processing for multimedia analytics

    OpenAIRE

    2016-01-01

    Our intention is generating the right mixture of audio, speech and language technologies with big data ones. Some audio, speech and language automatic technologies are available or gaining enough degree of maturity as to be able to help to this objective: automatic speech transcription, query by spoken example, spoken information retrieval, natural language processing, unstructured multimedia contents transcription and description, multimedia files summarization, spoken emotion detection and ...

  16. Effects of audio-visual information and mode of speech on listener perceptions of alaryngeal speakers.

    Science.gov (United States)

    Evitts, Paul M; Van Dine, Ami; Holler, Aline

    2009-01-01

    There is minimal research on listener perceptions of an individual with a laryngectomy (IWL) based on audio-visual information. The aim of this research was to provide preliminary insight into whether listeners have different perceptions of an individual with a laryngectomy based on mode of presentation (audio-only vs. audio-visual) and mode of speech (tracheoesophageal, oesophageal, electrolaryngeal, normal). Thirty-four naïve listeners were randomly presented with a standard reading passage produced by one typical speaker from each mode of speech in both audio-only and audio-visual presentation mode. Listeners used a visual analogue scale (10 cm line) to indicate their perceptions of each speaker's personality. A significant effect for mode of speech was present. There was no significant difference in listener perceptions between mode of presentation using individual ratings. However, principal component analysis showed ratings were more favourable in the audio-visual mode. Results of this study suggest that visual information may only have a minor impact on listener perceptions of a speakers' personality and that mode of speech and degree of speech proficiency may only play a small role in listener perceptions. However, results should be interpreted with caution as results are based on only one speaker per mode of speech.

  17. Speech and non-speech audio-visual illusions: a developmental study.

    Directory of Open Access Journals (Sweden)

    Corinne Tremblay

    Full Text Available It is well known that simultaneous presentation of incongruent audio and visual stimuli can lead to illusory percepts. Recent data suggest that distinct processes underlie non-specific intersensory speech as opposed to non-speech perception. However, the development of both speech and non-speech intersensory perception across childhood and adolescence remains poorly defined. Thirty-eight observers aged 5 to 19 were tested on the McGurk effect (an audio-visual illusion involving speech, the Illusory Flash effect and the Fusion effect (two audio-visual illusions not involving speech to investigate the development of audio-visual interactions and contrast speech vs. non-speech developmental patterns. Whereas the strength of audio-visual speech illusions varied as a direct function of maturational level, performance on non-speech illusory tasks appeared to be homogeneous across all ages. These data support the existence of independent maturational processes underlying speech and non-speech audio-visual illusory effects.

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

    Science.gov (United States)

    Hage, Steffen R; Nieder, Andreas

    2015-05-06

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

  19. 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.

  20. 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.

  1. Studies on a Spatialized Audio Interface for Sonar

    Science.gov (United States)

    2011-10-03

    a piano. Chords from C6 (fundamental frequency 1046.5Hz) to C7 (fundamental frequency 2093Hz) were recorded. Similarly to speech signals, the...button push response required, while for the auditory task, sounds were presented in diotic mode over a headset with a vocal response required. To

  2. 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

  3. 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.

    Science.gov (United States)

    Schwartz, Jean-Luc; Savariaux, Christophe

    2014-07-01

    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.

  4. 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.

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

    Science.gov (United States)

    Drood, Pooya; Asl, Hanieh Davatgari

    2016-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Giovany Vega

    2016-06-01

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

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

    Directory of Open Access Journals (Sweden)

    Warrick Roseboom

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

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

    Science.gov (United States)

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

    2014-01-01

    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 the treatment procedure. Anxiety was measured by using Venham's picture test, pulse rate, blood pressure and oxygen saturation. Results: ‘Audio distraction’ was found efficacious in alleviating anxiety of pediatric dental patients. Conclusion: ‘Audio distraction’ did decrease the anxiety in pediatric patients to a significant extent. How to cite this article: Singh D, Samadi F, Jaiswal JN, Tripathi AM. Stress Reduction through Audio Distraction in Anxious Pediatric Dental Patients: An Adjunctive Clinical Study. Int J Clin Pediatr Dent 2014;7(3):149-152. PMID:25709291

  9. A New Audio Watermarking Method Based on Discrete Cosine Transform with a Gray Image

    Directory of Open Access Journals (Sweden)

    Mohammad Ibrahim Khan

    2012-09-01

    Full Text Available Many effective watermarking algorithms have been proposed and implemented for digital images anddigital video. However, a few algorithms have been proposed for audio watermarking. This is due to thefact that, human auditory system (HAS is far more complex and sensitive than human visual system (HVS.In this research work, a new method of embedding image data into the audio signal and additive audiowatermarking algorithm based on Discrete Cosine Transformation (DCT domain is proposed. First, theoriginal audio is transformed into DCT domain. The DCT coefficients are divided into a fixed number ofsubsections and the energy of each subsection is calculated. Next, watermark is generated from image byimage processing algorithm. Watermarks are then embedded into selected peaks of highest energysubsection. Experimental results demonstrate that the watermark is inaudible and this algorithm is robustto common operations of digital audio signal processing, such as noise addition, re-sampling, requantization and so on. To evaluate the performance of the proposed audio watermarking method,subjective and objective quality tests including Bit Error Rate (BER and Signal to Noise ratio (SNR areconducted.

  10. Audio networking at the Center for Art and Media Technology Karlsruhe

    Science.gov (United States)

    Dutilleux, Pierre

    1993-01-01

    The Center for Art and Media Technology is dedicated to art and its relationship to new media. The Center supports music as well as graphic art. It also will house museums. The Center will be fully operational by the middle of 1996. The audio network will interconnect five recording studios and a large theater with three control rooms. With the additional facilities, the number of 40 interconnected rooms is reached. As to the quality and the versatility, the network can be compared, to some extent, to that of a broadcast-building. Traditional networking techniques involve many kilometers of high quality audio-cables and bulky automated patch-bays. Still, we wish even more freedom in the way the rooms are interconnected. Digital audio and computer network technology are promising. Although digital audio technology is spreading, the size of the totally digital systems is still limited. Fiber optic and large capacity optical disks offer attractive alternatives to traditional techniques (cabling, multitrack recorders, sound archives, routing). The digital audio standards are evolving from point to point communication to network communication. A 1 Gbit/s network could be the backbone of a solution.

  11. The average direct current offset values for small digital audio recorders in an acoustically consistent environment.

    Science.gov (United States)

    Koenig, Bruce E; Lacey, Douglas S

    2014-07-01

    In this research project, nine small digital audio recorders were tested using five sets of 30-min recordings at all available recording modes, with consistent audio material, identical source and microphone locations, and identical acoustic environments. The averaged direct current (DC) offset values and standard deviations were measured for 30-sec and 1-, 2-, 3-, 6-, 10-, 15-, and 30-min segments. The research found an inverse association between segment lengths and the standard deviation values and that lengths beyond 30 min may not meaningfully reduce the standard deviation values. This research supports previous studies indicating that measured averaged DC offsets should only be used for exclusionary purposes in authenticity analyses and exhibit consistent values when the general acoustic environment and microphone/recorder configurations were held constant. Measured average DC offset values from exemplar recorders may not be directly comparable to those of submitted digital audio recordings without exactly duplicating the acoustic environment and microphone/recorder configurations.

  12. A Survey on Steganography Techniques in Real Time Audio Signals and Evaluation

    Directory of Open Access Journals (Sweden)

    Abdulaleem Z. Al-Othmani

    2012-01-01

    Full Text Available Steganography has proven to be one of the practical ways of securing data. It is a new kind of secret communication used mainly to hide secret data inside other innocent digital mediums. Most of existing steganographic techniques use digital multimedia files as cover mediums to hide secret data. Audio files and signals make appropriate mediums for steganography due to the high data transmission rate and the high level of redundancy. Hiding data in real time communication audio signals is not a simple mission. Steganography requirements as well as real time communication requirements are supposed to be met in order to construct a useful and useful data hiding application. In this paper we will survey the general principles of hiding secret information using audio technology, and provide an overview of current functions and techniques. These techniques will be evaluated across both, steganography and real time communication requirements.

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

    Directory of Open Access Journals (Sweden)

    Jose Juan Garcia-Hernandez

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

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

    Science.gov (United States)

    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 parameters and operation conditions are proposed. Moreover, in order to carry out fingerprint detection using just a fraction of the pirate audio clip, block-based embedding and its corresponding detector is proposed. Extensive simulations show the robustness of the proposed system against average collusion attack. Moreover, by using an efficient Fast Fourier Transform core and standard computer machines it is shown that the proposed system is suitable for real-world scenarios.

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

    Science.gov (United States)

    Kolekar, M. H.; Sengupta, S.

    2005-07-01

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

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

    DEFF Research Database (Denmark)

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

    2013-01-01

    Recommending TV content to groups of viewers is best carried out when relevant information such as the demographics of the group is available. However, it can be difficult and time consuming to extract information for every user in the group. This paper shows how an audio analysis of the age...... and gender of a group of users watching the TV can be used for recommending a sequence of N short TV content items for the group. First, a state of the art audio-based classifier determines the age and gender of each user in an M-user group and creates a group profile. A genetic recommender algorithm......, and the other half were selected using the audio-derived demographics. The recommended advertisements received a significant higher median rating of 7.75, as opposed to 4.25 for the randomly selected advertisements 1....

  17. A Hybrid Model for Secure Data Transfer in Audio Signals using HCNN and DD DWT

    Directory of Open Access Journals (Sweden)

    B. Geetha vani

    2013-08-01

    Full Text Available In today’s world, there are a number of cryptographic and steganography techniques used in order to have secured data transfer between a sender and a receiver. In this paper a new hybrid approach that integrates the merits of cryptography and audio steganography is presented. First, the original message is encrypted using chaotic neural network and the resultant cipher text is embedded into a cover audio using Double Density Discrete Wavelet Transform (DD DWT. The resultant stego audio is transmitted to the receiver and the reverse process is done in order to get back the original plain text. The proposed method presents a Steganography scheme along with the cryptography scheme which enhances the security of the algorithm.

  18. Audio Recorded Guided Imagery Method to Reduce Stress Hospitalisazation in School Age Children in Palu Hospital

    Directory of Open Access Journals (Sweden)

    Fitria Masulili

    2013-12-01

    Full Text Available Hospitalization is a condition of a person because of illness and hospital admission. Research objectives determine the influence of audio recorded guided imagery method to stress of hospitalization in school-age children in hospital in Palu. Quasi-experimental research design with pre and post test design with control group. The sample of children aged 7-12 years were 26 respondents intervention group and 26 control group respondents. Intervention is the method of audio recorded guided imagery, three times a day for two days (one session equal to15 minutes. The results showed the significant difference mean stress score of hospitalization after the intervention (Pv = 0.004. No contribution of confounding variables. Based on these results, audio recorded guided imagery intervention can be applied to care the sick pediatric in hospital.

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

    DEFF Research Database (Denmark)

    Christensen, Mads Græsbøll; Sturm, Bob L.

    2011-01-01

    In this paper, we consider the problem of finding sparse representations of audio signals for coding purposes. In doing so, it is of utmost importance that when only a subset of the present components of an audio signal are extracted, it is the perceptually most important ones. To this end, we...... propose a new iterative algorithm based on two principles: 1) a reweighted l1-norm based measure of sparsity; and 2) a reweighted l2-norm based measure of perceptual distortion. Using these measures, the considered problem is posed as a constrained convex optimization problem that can be solved optimally...... 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....

  20. STUDY ON AUDIO INFORMATION HIDING METHOD BASED ON MODIFIED PHASE PARTITION

    Institute of Scientific and Technical Information of China (English)

    Tong Ming; Hao Chongyang; Liu Xiaojun; Chen Yanpu

    2005-01-01

    Hiding efficiency of traditional audio information hiding methods is always low since the sentience similarity cannot be guaranteed. A new audio information hiding method is proposed in this letter which can impose the insensitivity with the audio phase for auditory and realize the information hiding through specific algorithm in order to modify local phase within the auditory perception. The algorithm is to introduce the operation of "set 1" and "set 0" for every phase vectors, then the phases must lie on the boundary of a phase area after modified. If it lies on "1" boundary, it comes by set 1 operation. If it lies on "0" boundary, it comes by set 0 operation. The results show that, compared with the legacy method, the proposed method has better auditory similarity, larger information embedding capacity and lower code error rate. As a kind of blind detect method, it fits for application scenario without channel interference.

  1. Audio-magnetotelluric investigation of allochthonous iron formations in the Archaean Reguibat shield (Mauritania): structural and mining implications

    Science.gov (United States)

    Bronner, G.; Fourno, J. P.

    1992-11-01

    The M'Haoudat range, considered as an allochthonous unit amid the strongly metamorphosed Archaean basement (Tiris Group), belongs to the Lower Proterozoic Ijil Group, weakly metamorphosed, constituted mainly by iron quartzites including red jaspers and high grade iron ore. Audio-magnetotelluric (AMT) soundings (frequency range 1-7500 HZ) were performed together with the systematic survey of the range (SNIM mining company). The non-linear least squares method was used to perform a smoothness-constrained data model. The obvious AMT resistivity contrasts between the M'Haoudat Unit (150-3500 ohm. m) and the Archaean basement (20 000 ohm. m) allow to state precisely that the two thrust surfaces, on both sides of the range, join together at a depth which increases from North-West to South-East, as the ore bodies. Inside the steeply dipping M'Haoudat Unit, the main beds of iron quartzites (1500-3500 ohm. m), schists (1000-1500 ohm. m) and hematite ores (150-300 ohm. m) were distinguished when their thickness exceeded 30 to 50 m. The existence of an hydrostatic level (1-50 ohm. m) and the steeply dipping architecture, very likely responsible for the lack of resistivity contrast on the upper part of some profiles, complicate the interpretation at high frequencies the thin layers being poorly defined.

  2. An Audio-Visual Resource Notebook for Adult Consumer Education. An Annotated Bibliography of Selected Audio-Visual Aids for Adult Consumer Education, with Special Emphasis on Materials for Elderly, Low-Income and Handicapped Consumers.

    Science.gov (United States)

    Virginia State Dept. of Agriculture and Consumer Services, Richmond, VA.

    This document is an annotated bibliography of audio-visual aids in the field of consumer education, intended especially for use among low-income, elderly, and handicapped consumers. It was developed to aid consumer education program planners in finding audio-visual resources to enhance their presentations. Materials listed include 293 resources…

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

    Science.gov (United States)

    Smayda, Kirsten E; Van Engen, Kristin J; Maddox, W Todd; Chandrasekaran, Bharath

    2016-01-01

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

  4. Frequency Selective Filtering of the Modulation Spectrum and its Impact on Consonant Identification

    DEFF Research Database (Denmark)

    Christiansen, Thomas Ulrich; Greenberg, Steven

    2009-01-01

    ] refers to the syllable-final liquid segment). Each syllable was processed so that only a portion of the original audio spectrum was present. Narrow (three-quarter octave) bands of speech, with center frequencies of 750 Hz, 1500 Hz and 3000 Hz, were presented individually and in combination with each...

  5. Constant Switching Frequency Self-Oscillating Controlled Class-D Amplifiers

    DEFF Research Database (Denmark)

    Nguyen-Duy, Khiem; Knott, Arnold; Andersen, Michael A. E.

    2014-01-01

    The self-oscillating control approach has been used extensively in class-D amplifiers. It has several advantages such as high bandwidth and high audio performance. However, one of the primary disadvantages in a self-oscillating controlled system is that the switching frequency of the amplifier va...

  6. Investigation of switching frequency variations and EMI properties in self-oscillating class D amplifiers

    DEFF Research Database (Denmark)

    Nielsen, Dennis; Knott, Arnold; Pfaffinger, Gerhard;

    2009-01-01

    Class D audio amplifiers have gained significant influence in sound reproduction due to their high efficiency. One of the most commonly used control methods in these amplifiers is self-oscillation. A parameter of key interest in self-oscillating amplifiers is the switching frequency, which is known...

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

    DEFF Research Database (Denmark)

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

    2013-01-01

    This paper deals with system-level optimization of a digital-to-analog converter (DAC) for hearing-aid audio Class D output stage. We discuss the ΣΔ modulator system-level design parameters – the order, the oversampling ratio (OSR) and the number of bits in the quantizer. We show that combining...... by comparing two ΣΔ modulator designs. The proposed optimization has impact on the whole hearing-aid audio back-end system including less hardware in the interpolation filter and half the switching rate in the digital-pulse-width-modulation (DPWM) block and Class D output stage...

  8. Instructional Insights: Audio Feedback as Means of Engaging the Occupational Therapy Student.

    Science.gov (United States)

    Nielsen, Sarah K

    2016-01-01

    Constructivist learning approaches require faculty to engage students in the reflective learning process, yet students can begin to view this process as mundane and at times not engage in the process or utilize feedback provided. This article describes the results of applying audio feedback to overcome these obstacles in a practicum integration course. Student report and assignment performance indicated increased learning and engagement. The instructor found giving audio feedback more efficient than written feedback as it overcame inflection issues associated with the written word. Recorded files also alleviated additional student appointments for clarification of the feedback.

  9. Content-based audio authentication using a hierarchical patchwork watermark embedding

    Science.gov (United States)

    Gulbis, Michael; Müller, Erika

    2010-05-01

    Content-based audio authentication watermarking techniques extract perceptual relevant audio features, which are robustly embedded into the audio file to protect. Manipulations of the audio file are detected on the basis of changes between the original embedded feature information and the anew extracted features during verification. The main challenges of content-based watermarking are on the one hand the identification of a suitable audio feature to distinguish between content preserving and malicious manipulations. On the other hand the development of a watermark, which is robust against content preserving modifications and able to carry the whole authentication information. The payload requirements are significantly higher compared to transaction watermarking or copyright protection. Finally, the watermark embedding should not influence the feature extraction to avoid false alarms. Current systems still lack a sufficient alignment of watermarking algorithm and feature extraction. In previous work we developed a content-based audio authentication watermarking approach. The feature is based on changes in DCT domain over time. A patchwork algorithm based watermark was used to embed multiple one bit watermarks. The embedding process uses the feature domain without inflicting distortions to the feature. The watermark payload is limited by the feature extraction, more precisely the critical bands. The payload is inverse proportional to segment duration of the audio file segmentation. Transparency behavior was analyzed in dependence of segment size and thus the watermark payload. At a segment duration of about 20 ms the transparency shows an optimum (measured in units of Objective Difference Grade). Transparency and/or robustness are fast decreased for working points beyond this area. Therefore, these working points are unsuitable to gain further payload, needed for the embedding of the whole authentication information. In this paper we present a hierarchical extension

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

    OpenAIRE

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

    2013-01-01

    Part 21: Electronics: Applications; International audience; This paper deals with system-level optimization of a digital-to-analog converter (DAC) for hearing-aid audio Class D output stage. We discuss the ΣΔ modulator system-level design parameters – the order, the oversampling ratio (OSR) and the number of bits in the quantizer. We show that combining a reduction of the OSR with an increase of the order results in considerable power savings while the audio quality is kept. For further savin...

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

    DEFF Research Database (Denmark)

    Kaplanis, Neofytos; Bech, Søren; Sakari, Tervo;

    2016-01-01

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

  12. Transference & Retrieval of Pulse-code modulation Audio over Short Messaging Service

    CERN Document Server

    Khan, Muhammad Fahad

    2012-01-01

    The paper presents the method of transferring PCM (Pulse-Code Modulation) based audio messages through SMS (Short Message Service) over GSM (Global System for Mobile Communications) network. As SMS is text based service, and could not send voice. Our method enables voice transferring through SMS, by converting PCM audio into characters. Than Huffman coding compression technique is applied in order to reduce numbers of characters which will latterly set as payload text of SMS. Testing the said method we develop an application using J2me platform

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

    DEFF Research Database (Denmark)

    Laursen, Ditte

    is structured through association, offering an experience more comparable to an audio documentary than a traditional guided tour. Instead of directing visitors' focus of attention to selected points and objects provided by the museum as a producer, the sound collage relates indirectly to the various objects...... that visitors are fond of using their own mobile phones - but they have several problems with their phones in downloading the MP3 file. Read more: Guided Expectations: A Case Study of a Sound Collage Audio Guide | conference.archimuse.com...

  14. Reliable Transmission of Audio Streams in Lossy Channels Using Application Level Data Hiding

    Directory of Open Access Journals (Sweden)

    Parag Agarwal

    2008-12-01

    Full Text Available The paper improves the reliability of audio streams in a lossy channel. The mechanism groups audio data samples into source and carrier sets. The carrier set carry the information about the source set which is encoded using data hiding methodology - quantization index modulation. At the receiver side, a missing source data sample can be reconstructed using the carrier set and the remaining source set. Based on reliability constraints a hybrid design combining interleaving and data hiding is presented. Experiments show an improved reliability as compared to forward error correction and interleaving.

  15. The Application of Audio-lingual Teaching Method to College English Teaching in China

    Institute of Scientific and Technical Information of China (English)

    邢继强

    2010-01-01

    @@ 0 Introduction With the appearance of different linguistic schools,different language teaching methods come into being since the 19th century. In the long development of language teaching, each method is closely related to linguistic research, teaching practice and the social needs,so does audio-lingual teaching method.However, the English teaching in college is experiencing a whole reform and transfers the goal to cultivate the ability of the language use,especially the listening and speaking ability. Therefore, audio -lingual language teaching method will have new application and instruction to college English teachers.

  16. Audio-magnetotelluric survey to characterize the Sunnyside porphyry copper system in the Patagonia Mountains, Arizona

    Science.gov (United States)

    Sampson, Jay A.; Rodriguez, Brian D.

    2010-01-01

    The Sunnyside porphyry copper system is part of the concealed San Rafael Valley porphyry system located in the Patagonia Mountains of Arizona. The U.S. Geological Survey is conducting a series of multidisciplinary studies as part of the Assessment Techniques for Concealed Mineral Resources project. To help characterize the size, resistivity, and skin depth of the polarizable mineral deposit concealed beneath thick overburden, a regional east-west audio-magnetotelluric sounding profile was acquired. The purpose of this report is to release the audio-magnetotelluric sounding data collected along that east-west profile. No interpretation of the data is included.

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

    DEFF Research Database (Denmark)

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

    2012-01-01

    This paper deals with a design of a digital interpolation filter for a 3rd order multi-bit ΣΔ modulator with over-sampling ratio OSR = 64. The interpolation filter and the ΣΔ modulator are part of the back-end of an audio signal processing system in a hearing-aid application. The aim in this paper...... in the interpolation filter are investigated. Proposed design simplifications presented here result in the least hardware demanding combination of oversampling ratio, number of stages and number of filter taps among a number of filters reported for audio applications....

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

    DEFF Research Database (Denmark)

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

    2009-01-01

    . Electromagnetic interference between switch-mode audio power amplifiers and receivers show the same physical obstacle as the described ANC endeavors are targeting. The principle of active electromagnetic interference cancelation (AEC) is derived in this paper on a theoretical basis with verifications...... in simulation and experiment. The resulting switch-mode audio power amplifier of this experiment keeps its high efficiency and is able to deliver the signal with less than 0.1 % distortion, while improving the source of electromagnetic interference by 15 dB....

  19. Implementación de efectos de audio en la tarjeta Raspberry Pi B+

    OpenAIRE

    Sampedro Llopis, Hermes

    2015-01-01

    Este proyecto consiste en el diseño e implementación de un procesador digital de efectos de audio en tiempo real orientado a instrumentos eléctricos tales como guitarras, bajos, teclados, etc. El procesador está basado en la tarjeta Raspberry Pi B+, ordenador de placa reducida de bajo coste, desarrollado en Reino unido y cuyo lanzamiento tuvo lugar en el año 2012. En primer lugar, ha sido necesario lograr que la tarjeta asuma la funcionalidad de un procesador de audio en tiempo real. Para ell...

  20. Wireless data transmission through in-band on-channel digital audio broadcasting

    Science.gov (United States)

    Vigil, A. J.

    1995-12-01

    USA Digital Radio (USADR) is finalizing the development of new state-of-the-art formats for in-band on-channel (IBOC) delivery of digital audio broadcasting (DAB). USADR's IBOC DAB systems are designed for top-notch digital audio delivery as well as enhanced ancillary data transmission capabilities. USADR's AM and FM IBOC DAB systems employ MUSICAMR source encoding as well as innovative modulation techniques which address the various radio channel impairments characteristic of AM and FM band propagation. The USADR IBOC DAB systems are designed to be backwards compatible with conventional AM and FM broadcasting for a seamless and cost-effective transition to DAB.