WorldWideScience

Sample records for audio frequency

  1. Audio Frequency Analysis in Mobile Phones

    Science.gov (United States)

    Aguilar, Horacio Munguía

    2016-01-01

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

  2. Audio Classification from Time-Frequency Texture

    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.

  3. Calibration of an audio frequency noise generator

    DEFF Research Database (Denmark)

    Diamond, Joseph M.

    1966-01-01

    A noise generator of known output is very convenient in noise measurement. At low audio frequencies, however, all devices, including noise sources, may be affected by excess noise (1/f noise). It is therefore very desirable to be able to check the spectral density of a noise source before it is...... a noise bandwidth Bn = π/2 × (3dB bandwidth). To apply this method to low audio frequencies, the noise bandwidth of the low Q parallel resonant circuit has been found, including the effects of both series and parallel damping. The method has been used to calibrate a General Radio 1390-B noise...

  4. Frequency Hopping Method for Audio Watermarking

    Directory of Open Access Journals (Sweden)

    A. Anastasijević

    2012-11-01

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

  5. Robust Audio Watermarking Based on Log-Polar Frequency Index

    Science.gov (United States)

    Yang, Rui; Kang, Xiangui; Huang, Jiwu

    In this paper, we analyze the audio signal distortions introduced by pitch-scaling, random cropping and DA/AD conversion, and find a robust feature, average Fourier magnitude over the log-polar frequency index(AFM), which can resist these attacks. Theoretical analysis and extensive experiments demonstrate that AFM is an appropriate embedding region for robust audio watermarking. This is the first work on applying log-polar mapping to audio watermark. The usage of log-polar mapping in our work is basically different from the existing works in image watermarking. The log-polar mapping is only applied to the frequency index, not to the transform coefficients, which avoids the reconstruction distortion of inverse log-polar transform and reduces the computation cost. Comparison with the existing methods, the proposed AFM-based watermarking scheme has the outstanding performance on resisting pitch-scaling and random cropping, as well as very approving robustness to DA/AD conversion and TSM (Time-Scale Modification). The watermarked audio achieves high auditory quality. Experimental results show that the scheme is very robust to common audio signal processing and distortions introduced in Stirmark for Audio.

  6. Electric Discharge in Pin-Plate Audio Frequency Plasma

    Institute of Scientific and Technical Information of China (English)

    A. A. Azooz; S. K. Talal

    2011-01-01

    Experimental results on some properties of electric discharge initiated by audio frequency voltages in the range of 50-10000 Hz are presented. These results indicate that there are at least two modes of plasma ionic oscillations. A resonance-type behavior is seen when the driving field frequency becomes equal to the plasma ionic frequency. The results for plasma density and plasma temperature for both modes are presented.%Experimental results on some properties of electric discharge initiated by audio frequency voltages in the range of 50-10000Hz are presented.These results indicate that there are at least two modes of plasma ionic oscillations.A resonance-type behavior is seen when the driving field frequency becomes equal to the plasma ionic frequency.The resultsfor plasma density and plasma temperature for both modes are presented.Electric discharge studies have led to great developments in a wide range of science and technology applications.One may broadly categorize the electric discharges in gases into two main types,i.e.direct dc and ac discharges.A wide range of dc discharge properties related to geometry,type of gases,pressure etc.has been studied theoretically and experimentally.There has been a comparable amount of work related to ac discharges,which involve both inductive and capacitive coupling of an electric field to gases.However,for this type of discharge,one can argue that the emphases have been more focused on the radio and microwave ranges of ac frequency spectra.Results on both dc and rf discharges is now contained in many textbooks.[1-3] Although many studies related to audio frequency ranges have been carried out,their number is by no means comparable to those in the rf range.This may be due to two main reasons.The first stems from the theoretical arguments that as far as capacitive coupling is concerued,one would not expect to observe any major differences between dc and audio frequency discharge[4] due to the relaxation time properties of

  7. Audio Signal Processing Using Time-Frequency Approaches: Coding, Classification, Fingerprinting, and Watermarking

    Directory of Open Access Journals (Sweden)

    K. Umapathy

    2010-01-01

    Full Text Available Audio signals are information rich nonstationary signals that play an important role in our day-to-day communication, perception of environment, and entertainment. Due to its non-stationary nature, time- or frequency-only approaches are inadequate in analyzing these signals. A joint time-frequency (TF approach would be a better choice to efficiently process these signals. In this digital era, compression, intelligent indexing for content-based retrieval, classification, and protection of digital audio content are few of the areas that encapsulate a majority of the audio signal processing applications. In this paper, we present a comprehensive array of TF methodologies that successfully address applications in all of the above mentioned areas. A TF-based audio coding scheme with novel psychoacoustics model, music classification, audio classification of environmental sounds, audio fingerprinting, and audio watermarking will be presented to demonstrate the advantages of using time-frequency approaches in analyzing and extracting information from audio signals.

  8. 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.

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

    Science.gov (United States)

    Evangelista, Gianpaolo; Cavaliere, Sergio

    2001-12-01

    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.

  10. 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.

  11. 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...

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

    Directory of Open Access Journals (Sweden)

    Sadchenko A. V.

    2016-02-01

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

  13. 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.

  14. 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.

  15. 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.

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

    Science.gov (United States)

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

    2016-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Anita Arvind Deshmukh

    2014-12-01

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

  18. Performance Analysis of Gated Ring Oscillator Designed for Audio Frequency Range Asynchronous ADC

    Directory of Open Access Journals (Sweden)

    Anita Arvind Deshmukh

    2014-12-01

    Full Text Available This paper presents performance analysis of Gated R ing Oscillator (GRO. Proposed GRO is designed to employ in implementation of Time to Digital Convert er (TDC block of Asynchronous ADC. For an audio frequency range ADC, minimum GRO stages are designe d using asynchronous technique. So leads to reduced area and power. Compared to conventional Ri ng Oscillator (RO, we avoided to employ the gated clock; to evade clock design related problems like jitter, additional area and power. Instead we prefe rred gating of ring oscillator itself. Consequently duri ng sleep mode, GRO disables automatically which sav es the dynamic power. Furthermore it also provides fir st order noise shaping of the quantization and mismatch noise. Proposed GRO is implemented with 0. 18 μ m CMOS Digital Technology in Cadence Virtuso environment. GRO performance analysis shows oscillation frequency as 286 KHz with 327ps jitter and average power consumption of 1.08 μ W.

  19. 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.

  20. 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.;

    on the masking waveform in both domains. A significant release from masking of a 32-Hz signal in the modulation frequency domain was obtained only when the venelope fluctuations were slower than 1–2 Hz. Both experiments suggest a relatively weak contribution of venelope cues in the AM domain when compared......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...... in the envelope-frequency domain. One set of experiments examined the relationship between gated intensity discrimination and continuous-carrier increment detection. In contrast to the asymmetry observed in the audio-frequency domain (listeners are more sensitive to increments), AM-depth discrimination thresholds...

  1. 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. PMID:27442454

  2. 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

  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 2008: Audio Fixation

    Science.gov (United States)

    Kaye, Alan L.

    2008-01-01

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

  5. Piezoelectric two-layer stacks of cellular polypropylene ferroelectrets: transducer response at audio and ultrasound frequencies.

    Science.gov (United States)

    Wegener, Michael; Bergweiler, Steffen; Wirges, Werner; Pucher, Andreas; Tuncer, Enis; Gerhard-Multhaupt, Reimund

    2005-09-01

    Piezoelectric cellular polypropylene films, so-called ferroelectrets, are assembled in a stack with two active transducer layers. The stack is characterized with respect to its linear and quadratic response in a frequency range from 1 kHz to 80 kHz. A relatively smooth frequency response in the sound-pressure level is found for the individual layers as well as for both layers driven in phase. The piezoelectric response of the two-layer stack is twice the response of an individual layer over a rather broad frequency range. Furthermore, the influence of the preparation conditions on the resonance frequency and the effect of the quadratic distortion on the radiated sound are investigated both for the individual transducer films in the stack and for the stack system as a whole. PMID:16285459

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

    Science.gov (United States)

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

    2016-06-01

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

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

    Institute of Scientific and Technical Information of China (English)

    2001-01-01

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

  8. 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.

  9. 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....

  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. 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.

  12. 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.

  13. A Study of Audio Tape: Part II

    Science.gov (United States)

    Reen, Noel K.

    1975-01-01

    To evaluate reel audio tape, tests were performed to identify: signal-to-noise ratio, total harmonic distortion, dynamic response, frequency response, biased and virgin tape noise, dropout susceptibility and oxide coating uniformity. (SCC)

  14. Enhancing Manual Scan Registration Using Audio Cues

    Science.gov (United States)

    Ntsoko, T.; Sithole, G.

    2014-04-01

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

  15. Watermarking Algorithm Based on Audio Features and Smaller Values of Low Frequency Coefficients%基于音频特征和低频系数较小值的水印算法

    Institute of Scientific and Technical Information of China (English)

    杨得国; 李智; 姜金娣

    2012-01-01

    This paper presents a digital watermarking algorithm based on audio features and the relatively smaller values of the low frequency of DWT coefficients. The proposed algorithm analyzes the zero-cross ratio and the short-time energy of frame to select the appropriate threshold for initial discarding the high-frequency signal components of the audio frame, selecting the audio frame to be processed. The selected audio frames are stitched together and perform discrete wavelet transform, choosing the low-frequency coefficients and segmenting, using the sum binary value of the neighboring three watermark bit combination to determine the location of the watermark embedding, and make its value smaller. The proposed algorithm for blind watermark detection reduces the extraction time complexity of low-frequency components through the analysis of audio features, and improves the robustness of the watermark.%提出一种基于音频特征和DWT低频系数相对较小值的数字水印算法.分析音频帧的过零率及短时能量,选取适当的阈值初步舍弃表明信号中高频信号成分的音频帧,筛选出待处理的音频帧.将选定的音频帧拼接在一起进行小波变换,选取低频系数并分段,通过相邻3个水印比特组合的二进制之和确定水印在该段中的嵌入位置,将系数修改为相邻系数中较小的值.实验结果表明,该算法通过对音频特征的分析,能降低提取低频分量的时间复杂度,实现水印信息的盲检测,提高水印的鲁棒性.

  16. 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 materialit...

  17. 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.

  18. Audio Watermarking Algorithm Based on Centroid and Statistical Features

    Science.gov (United States)

    Zhang, Xiaoming; Yin, Xiong

    Experimental testing shows that the relative relation in the number of samples among the neighboring bins and the audio frequency centroid are two robust features to the Time Scale Modification (TSM) attacks. Accordingly, an audio watermark algorithm based on frequency centroid and histogram is proposed by modifying the frequency coefficients. The audio histogram with equal-sized bins is extracted from a selected frequency coefficient range referred to the audio centroid. The watermarked audio signal is perceptibly similar to the original one. The experimental results show that the algorithm is very robust to resample TSM and a variety of common attacks. Subjective quality evaluation of the algorithm shows that embedded watermark introduces low, inaudible distortion of host audio signal.

  19. 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

  20. 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...

  1. Audio fingerprint extraction for content identification

    Science.gov (United States)

    Shiu, Yu; Yeh, Chia-Hung; Kuo, C. C. J.

    2003-11-01

    In this work, we present an audio content identification system that identifies some unknown audio material by comparing its fingerprint with those extracted off-line and saved in the music database. We will describe in detail the procedure to extract audio fingerprints and demonstrate that they are robust to noise and content-preserving manipulations. The main feature in the proposed system is the zero-crossing rate extracted with the octave-band filter bank. The zero-crossing rate can be used to describe the dominant frequency in each subband with a very low computational cost. The size of audio fingerprint is small and can be efficiently stored along with the compressed files in the database. It is also robust to many modifications such as tempo change and time-alignment distortion. Besides, the octave-band filter bank is used to enhance the robustness to distortion, especially those localized on some frequency regions.

  2. Audio Source Separation Using a Deep Autoencoder

    OpenAIRE

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

    2014-01-01

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

  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. Categorizing Video Game Audio

    DEFF Research Database (Denmark)

    Westerberg, Andreas Rytter; Schoenau-Fog, Henrik

    2015-01-01

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

  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. The Digital Audio Editor as a Teaching and Laboratory Tool

    Science.gov (United States)

    Latta, Gregory

    2001-10-01

    Digital audio editors such as Software Audio Workshop and Cool Edit Pro are powerful tools used in the radio and audio recording fields for editing digital audio. However, they are also powerful tools in the physics classroom and laboratory. During this presentation the author will show how a digital audio editor, combined with a library of audio .wav files produced by the author as part of sabbatical work, can be used to: 1. demonstrate quantitatively and qualitatively the relationship between the decibel, sound intensity, and loudness perception, 2. demonstrate quantitatively and qualitatively the relationship between frequency and pitch perception, 3. perform additive and subtractive sound synthesis, 4. demonstrate comb filtering, 5. demonstrate constructive and destructive interference, and 6. turn the computer into an accurate signal generator (sine wave, square wave, etc.) with a frequency resolution of 1Hz. Availability of the required software and .wav file library will also be discussed.

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

    Directory of Open Access Journals (Sweden)

    P. Dhanalakshmi

    2010-12-01

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

  8. 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

  9. 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.

  10. 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.

  11. DWT-Based High Capacity Audio Watermarking

    Science.gov (United States)

    Fallahpour, Mehdi; Megías, David

    This letter suggests a novel high capacity robust audio watermarking algorithm by using the high frequency band of the wavelet decomposition, for which the human auditory system (HAS) is not very sensitive to alteration. The main idea is to divide the high frequency band into frames and then, for embedding, the wavelet samples are changed based on the average of the relevant frame. The experimental results show that the method has very high capacity (about 5.5kbps), without significant perceptual distortion (ODG in [-1, 0] and SNR about 33dB) and provides robustness against common audio signal processing such as added noise, filtering, echo and MPEG compression (MP3).

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

    Institute of Scientific and Technical Information of China (English)

    姜仕仁; 黄俊; 韩省华; 曾宪霖

    2011-01-01

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

  13. 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.

  14. Portable Audio Design

    DEFF Research Database (Denmark)

    Groth, Sanne Krogh

    2014-01-01

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

  15. 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...... by the implementation presented. Future work could include further refinement of the implementation of the concepts, electromagnetic interference investigations or PCB design....

  16. Spatial domain entertainment audio decompression/compression

    Science.gov (United States)

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

    2014-02-01

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

  17. Multi Carrier Modulation Audio Power Amplifier with Programmable Logic

    DEFF Research Database (Denmark)

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

    2009-01-01

    for performance and out of band spectral amplitudes. The basic principle in MCM is to use programmable logic to combine two or more Pulse Width Modulated (PWM) audio signals at different switching frequencies. In this way the out of band spectrum will be lowered compared with conventional class D amplifiers...... frequencies entering the audio band. Still, the MS MCM topology with two carrier signals shows a 6 dB reduction of the switching frequency amplitudes as well as THD across the audio band below 1% at 55 W output power open loop....

  18. 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.

  19. 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.

  20. Structure Learning in Audio

    OpenAIRE

    Nielsen, Andreas Brinch; Hansen, Lars Kai

    2009-01-01

    By having information about the setting a user is in, a computer is able to make decisions proactively to facilitate tasks for the user. Two approaches are taken in this thesis to achieve more information about an audio environment. One approach is that of classifying audio, and a new approach using pitch dynamics is suggested. The other approach is finding structures between the mixings of multiple sources based on an assumption of statistical independence of the sources. Three different aud...

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

    Institute of Scientific and Technical Information of China (English)

    苏杰; 吴广宁; 曹晓斌; 马御棠; 李瑞芳; 周炜明

    2011-01-01

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

  2. 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.

  3. 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.

  4. 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.

  5. 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 ...

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

    OpenAIRE

    Adler, Daniel

    1996-01-01

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

  7. DAFX Digital Audio Effects

    CERN Document Server

    2011-01-01

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

  8. 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

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

    International Nuclear Information System (INIS)

    We demonstrate the use of audio electronics-based signals to perform on-chip electrochemical measurements. Cell phones and portable music players are examples of consumer electronics that are easily operated and are ubiquitous worldwide. Audio output (play) and input (record) signals are voltage based and contain frequency and amplitude information. A cell phone, laptop soundcard and two compact audio players are compared with respect to frequency response; the laptop soundcard provides the most uniform frequency response, while the cell phone performance is found to be insufficient. The audio signals in the common portable music players and laptop soundcard operate in the range of 20 Hz to 20 kHz and are found to be applicable, as voltage input and output signals, to impedance-based electrochemical measurements in microfluidic systems. Validated impedance-based measurements of concentration (0.1–50 mM), flow rate (2–120 µL min−1) and particle detection (32 µm diameter) are demonstrated. The prevailing, lossless, wave audio file format is found to be suitable for data transmission to and from external sources, such as a centralized lab, and the cost of all hardware (in addition to audio devices) is ∼10 USD. The utility demonstrated here, in combination with the ubiquitous nature of portable audio electronics, presents new opportunities for impedance-based measurements in portable microfluidic systems. (technical note)

  10. 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.

  11. Efectos digitales de audio con Web Audio API

    OpenAIRE

    GARCÍA CHAPARRO, SAMUEL

    2015-01-01

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

  12. 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

  13. 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.

  14. 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…

  15. 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...

  16. 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.

  17. Efficiency Optimization in Class-D Audio Amplifiers

    DEFF Research Database (Denmark)

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

    2015-01-01

    This paper presents a new power efficiency optimization routine for designing Class-D audio amplifiers. The proposed optimization procedure finds design parameters for the power stage and the output filter, and the optimum switching frequency such that the weighted power losses are minimized under...... lead to around 30 % of efficiency improvement at 1.3 W output power without significant effects on both audio performance and the efficiency at high power levels....

  18. Digital Audio Watermarking: An Overview

    OpenAIRE

    Bhuvnesh Kumar Singh; Alok Kumar Singh

    2013-01-01

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

  19. Self-Healing Audio System

    OpenAIRE

    Sharma, Shubham; Sridhar, Aditya; Krishnia, Jai Prakash

    2015-01-01

    Installed sound applications typically involve a large number of audio processors, amplifiers and speaker systems spread across the venue. They could be spatially distributed at the venue across different rack rooms and floors. These systems are commissioned and configured by sound engineers using software application(s). This is essentially a one-time activity, following which, the audio systems run independently. Detection of faults and reconfiguration of any audio device(s) that fail(s) is...

  20. 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

  1. Structure Learning in Audio

    DEFF Research Database (Denmark)

    Nielsen, Andreas Brinch

    and speech, using novel features based on pitch dynamics. Within instrument classification two different harmonic models have been compared. Finally voiced/unvoiced segmentation of popular music is done based on MFCC’s and AR coefficients. The structures in the mixings of multiple sources have been...... investigated. A fast and computationally simple approach that compares recordings and classifies if they are from the same audio environment have been developed, and shows very high accuracy and the ability to synchronize recordings in the case of recording devices which are not connected. A more general model...

  2. Efficient Audio Power Amplification - Challenges

    DEFF Research Database (Denmark)

    Andersen, Michael Andreas E.

    2005-01-01

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

  3. 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)

  4. 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.

  5. 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.

  6. 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

  7. 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.

  8. 2.5维可控源音频大地电磁法Occam反演理论及应用%2.5 D controlled source audio-frequency magnetotellurics OCCAM inversion

    Institute of Scientific and Technical Information of China (English)

    何梅兴; 胡祥云; 叶益信; 罗文行

    2011-01-01

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

  9. Lossless Audio Watermarking Based on the Alpha Statistic Modulation

    Directory of Open Access Journals (Sweden)

    Sunita V. Dhavale

    2012-08-01

    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 sized audio frames. Experimental results reveal that the proposed watermarking scheme maintains high audio quality and is simultaneously highly robust to pirate attacks, including MP3 compression, cropping, filtering, re-sampling, and re-quantization.

  10. Digital Audio Watermarking: An Overview

    Directory of Open Access Journals (Sweden)

    Bhuvnesh Kumar Singh

    2013-10-01

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

  11. 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...

  12. 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....

  13. 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.

  14. 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,

  15. 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

  16. 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.

  17. 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.

  18. ENERGY STAR Certified Audio Video

    Data.gov (United States)

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

  19. Tourism research and audio methods

    DEFF Research Database (Denmark)

    Jensen, Martin Trandberg

    2016-01-01

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

  20. Python for audio signal processing

    OpenAIRE

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

    2011-01-01

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

  1. A Reproducible Research Framework for Audio Inpainting

    OpenAIRE

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

    2011-01-01

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

  2. Development of an audio input toolkit for multiple sources

    OpenAIRE

    Kosch, Thomas

    2013-01-01

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

  3. Class-D audio amplifiers with negative feedback

    OpenAIRE

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

    2006-01-01

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

  4. 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.

  5. 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

  6. Location audio simplified capturing your audio and your audience

    CERN Document Server

    Miles, Dean

    2014-01-01

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

  7. A Joint Audio-Visual Approach to Audio Localization

    DEFF Research Database (Denmark)

    Jensen, Jesper Rindom; Christensen, Mads Græsbøll

    2015-01-01

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

  8. 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.

  9. 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

  10. 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. 

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

    OpenAIRE

    Lepa, Steffen

    2013-01-01

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

  12. Audio 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

  13. QRDA: Quantum Representation of Digital Audio

    Science.gov (United States)

    Wang, Jian

    2016-03-01

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

  14. Digital Audio Application to Short Wave Broadcasting

    Science.gov (United States)

    Chen, Edward Y.

    1997-01-01

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

  15. 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

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

    Science.gov (United States)

    Wood, Paul; Sinton, David

    2010-08-01

    We demonstrate the use of audio electronics-based signals to perform on-chip electrochemical measurements. Cell phones and portable music players are examples of consumer electronics that are easily operated and are ubiquitous worldwide. Audio output (play) and input (record) signals are voltage based and contain frequency and amplitude information. A cell phone, laptop soundcard and two compact audio players are compared with respect to frequency response; the laptop soundcard provides the most uniform frequency response, while the cell phone performance is found to be insufficient. The audio signals in the common portable music players and laptop soundcard operate in the range of 20 Hz to 20 kHz and are found to be applicable, as voltage input and output signals, to impedance-based electrochemical measurements in microfluidic systems. Validated impedance-based measurements of concentration (0.1-50 mM), flow rate (2-120 µL min-1) and particle detection (32 µm diameter) are demonstrated. The prevailing, lossless, wave audio file format is found to be suitable for data transmission to and from external sources, such as a centralized lab, and the cost of all hardware (in addition to audio devices) is ~10 USD. The utility demonstrated here, in combination with the ubiquitous nature of portable audio electronics, presents new opportunities for impedance-based measurements in portable microfluidic systems.

  17. 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.

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

    OpenAIRE

    Nitin Arora

    2010-01-01

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

  19. 36 CFR 2.12 - Audio disturbances.

    Science.gov (United States)

    2010-07-01

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

  20. 50 CFR 27.72 - Audio equipment.

    Science.gov (United States)

    2010-10-01

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

  1. Presence and the utility of audio spatialization

    DEFF Research Database (Denmark)

    Bormann, Karsten

    2005-01-01

    The primary concern of this paper is whether the utility of audio spatialization, as opposed to the fidelity of audio spatialization, impacts presence. An experiment is reported that investigates the presence-performance relationship by decoupling spatial audio fidelity (realism) from task...

  2. Bit rates in audio source coding

    NARCIS (Netherlands)

    Veldhuis, Raymond N.J.

    1992-01-01

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

  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. 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)

  7. Haptic and Audio Interaction Design

    DEFF Research Database (Denmark)

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

  8. 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…

  9. 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.

  10. Audio Watermarking Based On The PSK Modulation

    Directory of Open Access Journals (Sweden)

    Wahid Barkouti

    2011-09-01

    Full Text Available Audio watermarking is a technique, which can be used to embed information into the digital representation of audio signals. The main challenge is to hide data representing some information withoutcompromising the quality of the watermarked track and at the same time ensure that the embedded watermark is robust against removal attacks. Especially providing perfect audio quality combined withhigh robustness against a wide variety of attacks is not adequately addressed and evaluated in current watermarking systems. In this paper, we present a new phase modulation audio watermarking technique,which among other features provides evidence for high audio quality. PSK modulation has been proposed as an effective approach to watermarking.

  11. 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.

  12. 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.

  13. 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.

  14. WAVE : an audio virtual environment

    OpenAIRE

    Valbom, Leonel; Marcos, Adérito

    2004-01-01

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

  15. Audio Interfaces for Improved Accessibility

    OpenAIRE

    Duarte, Carlos; Carrico, Lu&#;s

    2008-01-01

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

  16. Audio Watermarking with Error Correction

    OpenAIRE

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

    2011-01-01

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

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

    OpenAIRE

    Dangarwala, Kruti; Shah, Jigar

    2010-01-01

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

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

    OpenAIRE

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

    2012-01-01

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

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

    OpenAIRE

    Lepa, Steffen

    2013-01-01

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

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

    OpenAIRE

    Lepa, Steffen

    2013-01-01

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

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

    OpenAIRE

    Lepa, Steffen

    2013-01-01

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

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

    OpenAIRE

    Lepa, Steffen

    2013-01-01

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

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

    Science.gov (United States)

    Giannakopoulos, Theodoros

    2015-01-01

    Audio information plays a rather important role in the increasing digital content that is available today, resulting in a need for methodologies that automatically analyze such content: audio event recognition for home automations and surveillance systems, speech recognition, music information retrieval, multimodal analysis (e.g. audio-visual analysis of online videos for content-based recommendation), etc. This paper presents pyAudioAnalysis, an open-source Python library that provides a wide range of audio analysis procedures including: feature extraction, classification of audio signals, supervised and unsupervised segmentation and content visualization. pyAudioAnalysis is licensed under the Apache License and is available at GitHub (https://github.com/tyiannak/pyAudioAnalysis/). Here we present the theoretical background behind the wide range of the implemented methodologies, along with evaluation metrics for some of the methods. pyAudioAnalysis has been already used in several audio analysis research applications: smart-home functionalities through audio event detection, speech emotion recognition, depression classification based on audio-visual features, music segmentation, multimodal content-based movie recommendation and health applications (e.g. monitoring eating habits). The feedback provided from all these particular audio applications has led to practical enhancement of the library. PMID:26656189

  4. 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.

  5. 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.

  6. 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.

  7. Implementing Audio-CASI on Windows’ Platforms

    OpenAIRE

    Cooley, Philip C.; Turner, Charles F.

    1998-01-01

    Audio computer-assisted self interviewing (Audio-CASI) technologies have recently been shown to provide important and sometimes dramatic improvements in the quality of survey measurements. This is particularly true for measurements requiring respondents to divulge highly sensitive information such as their sexual, drug use, or other sensitive behaviors. However, DOS-based Audio-CASI systems that were designed and adopted in the early 1990s have important limitations. Most salient is the poor ...

  8. Weakly Supervised Scalable Audio Content Analysis

    OpenAIRE

    Kumar, Anurag; Raj, Bhiksha

    2016-01-01

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

  9. Audio Watermarking Based On The PSK Modulation

    OpenAIRE

    Wahid Barkouti; Sihem Nasri; Adnane Cherif

    2011-01-01

    Audio watermarking is a technique, which can be used to embed information into the digital representation of audio signals. The main challenge is to hide data representing some information withoutcompromising the quality of the watermarked track and at the same time ensure that the embedded watermark is robust against removal attacks. Especially providing perfect audio quality combined withhigh robustness against a wide variety of attacks is not adequately addressed and evaluated in current w...

  10. MODIS: an audio motif discovery software

    OpenAIRE

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

    2013-01-01

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

  11. 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

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

    OpenAIRE

    Theodoros Giannakopoulos

    2015-01-01

    Audio information plays a rather important role in the increasing digital content that is available today, resulting in a need for methodologies that automatically analyze such content: audio event recognition for home automations and surveillance systems, speech recognition, music information retrieval, multimodal analysis (e.g. audio-visual analysis of online videos for content-based recommendation), etc. This paper presents pyAudioAnalysis, an open-source Python library that provides a wid...

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

    OpenAIRE

    Gartenlaub, Arie Gal

    1994-01-01

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

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

    NARCIS (Netherlands)

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

    2015-01-01

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

  15. 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

  16. Audio-visual gender recognition

    Science.gov (United States)

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

    2007-11-01

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

  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. 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.

  19. Audio Quality for a Simple Forward Error Correcting Code

    OpenAIRE

    Calas, Yvan; Jean-Marie, Alain

    2004-01-01

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

  20. Progressive Syntax-Rich Coding of Multichannel Audio Sources

    OpenAIRE

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

    2003-01-01

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

  1. Progressive Syntax-Rich Coding of Multichannel Audio Sources

    OpenAIRE

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

    2003-01-01

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

  2. Custom Architecture for Immersive-Audio Applications

    NARCIS (Netherlands)

    Theodoropoulos, D.N.

    2011-01-01

    In this dissertation, we propose a new approach for rapid development of multi-core immersive-audio systems. We study two popular immersive-audio techniques, namely the Beamforming and the Wave Field Synthesis (WFS). Beamforming utilizes microphone arrays to extract acoustic sources recorded in a no

  3. 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...

  4. 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)

  5. 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…

  6. 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.

  7. An audio encryption using transposition method

    Directory of Open Access Journals (Sweden)

    Ahmad Jawahir

    2015-07-01

    Full Text Available Encryption is a technique to secure sounds data from attackers. In this study, transposition technique that corresponds to a WAV file extension is used. The performance of the transposition technique is measured using the mean square error (MSE. In the test, the value of MSE of the original and encrypted audio files were compared; the original and decrypted audio files used the correct password is ‘SEMBILAN’ and the incorrect password is ‘DELAPAN’. The experimental results showed that the original and encrypted audio files, and the original and decrypted audio files used the correct password that has a value of MSE = 0, and with the incorrect one with a value of MSE 0.00000428 or ≠ 0. In other words, the transposition technique is able to ensure the security of audio data files.

  8. 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

  9. 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

  10. High-Fidelity Piezoelectric Audio Device

    Science.gov (United States)

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

    2003-01-01

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

  11. Implementation of Psychoacoustic model in Audio Compression using Munich and Gammachirp Wavelets

    Directory of Open Access Journals (Sweden)

    D.Naveen,

    2010-05-01

    Full Text Available Audio compression is the technology of converting human speech into an efficiently encoded representation that can later be decoded to produce a close approximation of the original signal. This paperpresents an algorithm to compress high quality audio signal and maintaining transparent quality at low bit rates. Most psychoacoustic models for coding applications use a uniform spectral decomposition toapproximate the frequency selectivity of the human auditory system; however the equal filter properties of the uniform subbands do not match the non uniform characteristics of the cochlear filters. For implementing this algorithm a design of psycho-acoustic model was developed following the model used in the standard MPEG-1 audio layer 3. The architecture is based on appropriate wavelet packet decomposition instead of Short Term Fourier Transformation. Wavelet transform can be efficiently used to represent human speechwith less number of bits that can be decoded to produce a close approximation of the original speech signal. The psycho-acoustic model is designed to take advantage of the masking effect in human hearing. They minimize the number of bits required to represent each frame of audio material at a fixed distortion level . This paper compares the performance of Munich and Gammachirp audio coders based on compression ratio. For evaluating the best model five different audio signals are considered and applied for these two coders.This analysis shows that the Gammachirp offers best compression ratio and sound quality in comparison to the Munich coders.

  12. 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.

  13. Complete guide to high-end audio

    CERN Document Server

    Harley, Robert

    2010-01-01

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

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

    OpenAIRE

    Lepa, Steffen

    2013-01-01

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

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

    OpenAIRE

    Lepa, Steffen

    2013-01-01

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

  16. 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.

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

    OpenAIRE

    鷲尾, 槙也; 渡邊, 曜大

    2014-01-01

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

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

    OpenAIRE

    Mathew, Justin; Huot, Stéphane; Blum, Alan

    2014-01-01

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

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

    OpenAIRE

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

    2016-01-01

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

  20. Quantitative characterisation of audio data by ordinal symbolic dynamics

    Science.gov (United States)

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

    2013-06-01

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

  1. 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

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

    DEFF Research Database (Denmark)

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

    2014-01-01

    An experiment to determine the perceptual attributes of the experience of listening to a target audio program in the presence of an audio interferer was performed. The first stage was a free elicitation task in which a total of 572 phrases were produced. In the second stage, a consensus vocabulary......, annoyance, balance and blend, and confusion. Ratings using these attributes were collected in the fourth stage, and a principal component analysis performed. This suggested two dimensions underlying the perception of an audio-on-audio interference situation: The first dimension was labeled “distraction” and...

  3. CERN automatic audio-conference service

    CERN Document Server

    Sierra Moral, R

    2010-01-01

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

  4. 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...

  5. 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.

  6. Virtual Microphones for Multichannel Audio Resynthesis

    Science.gov (United States)

    Mouchtaris, Athanasios; Narayanan, Shrikanth S.; Kyriakakis, Chris

    2003-12-01

    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.

  7. 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.

  8. 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.

  9. 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.

  10. Implementation of Audio signal by using wavelet transform

    OpenAIRE

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

    2010-01-01

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

  11. 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...

  12. 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.

  13. 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

  14. 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.

  15. Digital Multicasting of Multiple Audio Streams

    Science.gov (United States)

    Macha, Mitchell; Bullock, John

    2007-01-01

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

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

    Science.gov (United States)

    Romero-Fresco, Pablo; Fryer, Louise

    2013-01-01

    Introduction: Time constraints limit the quantity and type of information conveyed in audio description (AD) for films, in particular the cinematic aspects. Inspired by introductory notes for theatre AD, this study developed audio introductions (AIs) for "Slumdog Millionaire" and "Man on Wire." Each AI comprised 10 minutes of…

  17. Audio Steganography Using GA with Multilevel Security

    Directory of Open Access Journals (Sweden)

    K. Bhowal

    2013-05-01

    Full Text Available In this paper we present a novel method for digital audio steganography where messages are embedded into image and image is embedded into the host audio. “Audio Steganography using GA with multilevel security” is a proposed system which is based on Steganography and Encryption; the system ensures secured data transfer between the source and destination. Here a novel approach is presented to resolve the remained problems of substitution technique of audio Steganography. In the first level of security, encrypted message bits are inserted into image using LSB algorithm. In the second level, a secured GA based LSB (Least Significant Bit Algorithm is used to encode the image data into audio data. Here image bits are embedded into random and higher LSB layers, resulting in increased robustness against noise addition. The robustness specially would be increased against those intentional attacks which try to reveal the hidden message and also some unintentional attacks like noise addition as well. On the other hand, multi-objective GA is used to reduce distortion

  18. Audio stream classification for multimedia database search

    Science.gov (United States)

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

    2013-03-01

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

  19. A Genetic Algorithm Optimization Technique for Multiwavelet-Based Digital Audio Watermarking

    Directory of Open Access Journals (Sweden)

    Kumsawat Prayoth

    2010-01-01

    Full Text Available We propose a new approach for optimization in digital audio watermarking using genetic algorithm. The watermarks are embedded into the low frequency coefficients in discrete multiwavelet transform domain. The embedding technique is based on quantization process which does not require the original audio signal in the watermark extraction. We have developed an optimization technique using the genetic algorithm to search for four optimal quantization steps in order to improve both quality of watermarked audio and robustness of the watermark. In addition, we analyze the performance of the proposed algorithm in terms of signal-to-noise ratio, normalized correlation, and bit error rate. The experimental results show that the proposed scheme can achieve a good robustness against most of the attacks which were included in this study.

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

    Directory of Open Access Journals (Sweden)

    Chan Jun Chun

    2009-12-01

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

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

    International Nuclear Information System (INIS)

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

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

    Science.gov (United States)

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

    2009-02-01

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

  3. Robust Audio Watermarking Against the D/A and A/D conversions

    CERN Document Server

    Xiang, Shijun

    2007-01-01

    Audio watermarking has played an important role in multimedia security. In many applications using audio watermarking, D/A and A/D conversions (denoted by DA/AD in this paper) are often involved. In previous works, however, the robustness issue of audio watermarking against the DA/AD conversions has not drawn sufficient attention yet. In our extensive investigation, it has been found that the degradation of a watermarked audio signal caused by the DA/AD conversions manifests itself mainly in terms of wave magnitude distortion and linear temporal scaling, making the watermark extraction failed. Accordingly, a DWT-based audio watermarking algorithm robust against the DA/AD conversions is proposed in this paper. To resist the magnitude distortion, the relative energy relationships among different groups of the DWT coefficients in the low-frequency sub-band are utilized in watermark embedding by adaptively controlling the embedding strength. Furthermore, the resynchronization is designed to cope with the linear t...

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

    Energy Technology Data Exchange (ETDEWEB)

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

    2009-02-15

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

  5. 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.

  6. Evaluation of Perceived Spatial Audio Quality

    Directory of Open Access Journals (Sweden)

    Jan Berg

    2006-04-01

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

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

    Science.gov (United States)

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

    2016-08-16

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

  8. 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.

  9. Personalized Audio Systems - a Bayesian Approach

    DEFF Research Database (Denmark)

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

    2013-01-01

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

  10. Synchronization and comparison of Lifelog audio recordings

    DEFF Research Database (Denmark)

    Nielsen, Andreas Brinch; Hansen, Lars Kai

    2008-01-01

    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......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...... of combining the two. In particular we show that the combination allows sample resolution synchronization and scalability....

  11. Near-field Localization of Audio

    DEFF Research Database (Denmark)

    Jensen, Jesper Rindom; Christensen, Mads Græsbøll

    2014-01-01

    Localization of audio sources using microphone arrays has been an important research problem for more than two decades. Many traditional methods for solving the problem are based on a two-stage procedure: first, information about the audio source, such as time differences-of-arrival (TDOAs......, where the desired signal is modeled using TDOAs and GROAs, which are determined by the source location. This facilitates the derivation of one-stage, maximum likelihood methods under a white Gaussian noise assumption that is applicable in both near- and far-field scenarios. Simulations show...

  12. 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.

  13. 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.

  14. 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.

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

    DEFF Research Database (Denmark)

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

    2015-01-01

    listener can be viewed as having a personal sound zone system. In order to evaluate and optimize such situations in a perceptually relevant manner, the authors created a predictive model using the features that contribute to the distraction from unwanted sounds. Feature extraction was motivated...... by a qualitative analysis of subject responses. Distraction ratings were collected for one hundred randomly created audio-on-audio interference situations with music target and interferer programs. The selected features were related to the overall loudness, loudness ratio, perceptual evaluation of audio source...

  16. A Study of Audio Tape

    Science.gov (United States)

    Reen, Noel K.

    1975-01-01

    This is part I of a report on a study comparing reel and cassette tapes for signal-to-noise ration, total harmonic distortion, dynamic response, frequency response, bias and virgin noise and oxide coating uniformity. Test equipment and procedures are described and results are discussed. Charts detail research findings. (CHK)

  17. Audio Mining with emphasis on Music Genre Classification

    DEFF Research Database (Denmark)

    Meng, Anders

    2004-01-01

    etc. is receiving quite a lot of attention. The first breakthough in audio mining was created by MuscleFish in 1996, a simple audio retrieval system. With the increasing amount of audio material being accessible through the web, e.g. Apple's iTunes (700,000+ songs), Sony, Amazon, new methods...... 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...... to choose from. Basically every audio mining system is more or less consisting of the same stages as for the music genre setting. My research so far has mainly focussed on finding relevant features for music genre classification living at different timescales using early and late information fusion. It has...

  18. 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...

  19. 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.

  20. Learning bimodal structure in audio-visual data

    OpenAIRE

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

    2009-01-01

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

  1. Assessment of spatial audio quality based on sound attributes

    OpenAIRE

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

    2012-01-01

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

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

    OpenAIRE

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

    2008-01-01

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

  3. 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.

  4. 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.

  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. 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...

  8. 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...

  9. 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... SERVICES RADIO BROADCAST SERVICES Digital Audio Broadcasting § 73.403 Digital audio broadcasting service requirements. (a) Broadcast radio stations using IBOC must transmit at least one over-the-air digital...

  10. 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%.

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

    Science.gov (United States)

    Keitel, Christian; Müller, Matthias M

    2016-05-01

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

  12. Providing Students with Formative Audio Feedback

    Science.gov (United States)

    Brearley, Francis Q.; Cullen, W. Rod

    2012-01-01

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

  13. 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…

  14. Audio-visual classification video browser

    OpenAIRE

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

    2014-01-01

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

  15. 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...

  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. 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.

  18. Structuring Broadcast Audio for Information Access

    Science.gov (United States)

    Gauvain, Jean-Luc; Lamel, Lori

    2003-12-01

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

  19. 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,…

  20. Audio-visual integration in schizophrenia

    NARCIS (Netherlands)

    Gelder, B.LM.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

  1. 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

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

    OpenAIRE

    Lepa, Steffen

    2013-01-01

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

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

    OpenAIRE

    Dubnov, Shlomo; Assayag, Gerard; Cont, Arshia

    2007-01-01

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

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

    OpenAIRE

    Montecchio, Nicola; Cont, Arshia

    2011-01-01

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

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

    OpenAIRE

    Lepa, Steffen

    2013-01-01

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

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

    OpenAIRE

    Lepa, Steffen

    2013-01-01

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

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

    OpenAIRE

    Lepa, Steffen

    2013-01-01

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

  8. Three-dimensional audio using loudspeakers

    Science.gov (United States)

    Gardner, William G.

    1997-12-01

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

  9. 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.

  10. 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.

  11. 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.

  12. Efficiently Synchronized Spread-Spectrum Audio Watermarking with Improved Psychoacoustic Model

    Directory of Open Access Journals (Sweden)

    Xing He

    2008-01-01

    Full Text Available This paper presents an audio watermarking scheme which is based on an efficiently synchronized spread-spectrum technique and a new psychoacoustic model computed using the discrete wavelet packet transform. The psychoacoustic model takes advantage of the multiresolution analysis of a wavelet transform, which closely approximates the standard critical band partition. The goal of this model is to include an accurate time-frequency analysis and to calculate both the frequency and temporal masking thresholds directly in the wavelet domain. Experimental results show that this watermarking scheme can successfully embed watermarks into digital audio without introducing audible distortion. Several common watermark attacks were applied and the results indicate that the method is very robust to those attacks.

  13. 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.

  14. 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...

  15. Audible Aliasing Distortion in Digital Audio Synthesis

    OpenAIRE

    J. Schimmel

    2012-01-01

    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'...

  16. 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.

  17. Synthecology: sound use of audio in teleimmersion

    Science.gov (United States)

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

    2006-02-01

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

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

    OpenAIRE

    Kruti Dangarwala; Jigar Shah

    2010-01-01

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

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

    OpenAIRE

    Baillie, M.; Jose, J.M.

    2004-01-01

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

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

    OpenAIRE

    Magron, Paul; Badeau, Roland; David, Bertrand

    2016-01-01

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

  1. A power-efficient audio amplifier combining switching and linear techniques

    OpenAIRE

    Zee, van der, KG Kristoffer; Tuijl, van, B.A.J.

    1999-01-01

    Integrated Class D audio amplifiers are very power efficient, but require an external filter which prevents further integration. Also due to this filter, large feedback factors are hard to realise, so that the load influences the distortion- and transfer characteristics. The amplifier presented in this paper consists of a switching part that contains a much simpler filter, and a linear part that ensures a low distortion and flat frequency response. A 30W version was realised. The switching pa...

  2. 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.

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

    DEFF Research Database (Denmark)

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

    2013-01-01

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

  4. Audio Classical Composer Identification by Deep Neural Network

    OpenAIRE

    Hu, Zhen; Fu, Kun; Zhang, Changshui

    2013-01-01

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

  5. Use of Effective Audio in E-learning Courseware

    OpenAIRE

    Ray, Kisor

    2015-01-01

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

  6. 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。

  7. A ROBUST WAVELET BASED WATERMARKING SCHEME FOR DIGITAL AUDIO

    Directory of Open Access Journals (Sweden)

    Ayad Ibrahim Abdulsada

    2012-06-01

    Full Text Available In this paper, a robust wavelet based watermarking scheme has been proposed for digital audio. A single bit is embedded in the approximation part of each frame. The watermark bits are embedded in two subsets of indexes randomly generated by using two keys for security purpose. The embedding process is done in adaptively fashion according to the mean of each approximation part. The detection of watermark does not depend on the original audio. To measure the robustness of the algorithm, different signal processing operations have been applied on the watermarked audio. Several experimental results have been conducted to illustrate the robustness and efficiency of the proposed watermarked audio scheme.

  8. 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.

  9. 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.

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

    Science.gov (United States)

    Carter, Curtis W.

    2012-01-01

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

  11. 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...

  12. 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.

  13. 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...

  14. Predistortion of a Bidirectional Cuk Audio Amplifier

    DEFF Research Database (Denmark)

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

    2014-01-01

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

  15. 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...

  16. 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

  17. 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

  18. 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.

  19. 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.

  20. 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.

  1. Robust High-Capacity Audio Watermarking Based on FFT Amplitude Modification

    Science.gov (United States)

    Fallahpour, Mehdi; Megías, David

    This paper proposes a novel robust audio watermarking algorithm to embed data and extract it in a bit-exact manner based on changing the magnitudes of the FFT spectrum. The key point is selecting a frequency band for embedding based on the comparison between the original and the MP3 compressed/decompressed signal and on a suitable scaling factor. The experimental results show that the method has a very high capacity (about 5kbps), without significant perceptual distortion (ODG about -0.25) and provides robustness against common audio signal processing such as added noise, filtering and MPEG compression (MP3). Furthermore, the proposed method has a larger capacity (number of embedded bits to number of host bits rate) than recent image data hiding methods.

  2. Simple Solutions for Space Station Audio Problems

    Science.gov (United States)

    Wood, Eric

    2016-01-01

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

  3. Semantic Labeling of Nonspeech Audio Clips

    Directory of Open Access Journals (Sweden)

    Xiaojuan Ma

    2010-01-01

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

  4. Beyond Podcasting: Creative Approaches to Designing Educational Audio

    Science.gov (United States)

    Middleton, Andrew

    2009-01-01

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

  5. 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…

  6. Content Discovery from Composite Audio: An unsupervised approach

    NARCIS (Netherlands)

    Lu, L.

    2009-01-01

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

  7. Using Audio Books to Improve Reading and Academic Performance

    Science.gov (United States)

    Montgomery, Joel R.

    2009-01-01

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

  8. A Case Study on Audio Feedback with Geography Undergraduates

    Science.gov (United States)

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

    2011-01-01

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

  9. Use of Audio Modification in Science Vocabulary Assessment

    Science.gov (United States)

    Adiguzel, Tufan

    2011-01-01

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

  10. 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.

  11. 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...

  12. The Practical Audio-Visual Handbook for Teachers.

    Science.gov (United States)

    Scuorzo, Herbert E.

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

  13. FPGA implementation of DWT for Audio Watermarking Application

    Directory of Open Access Journals (Sweden)

    Naveen.S.Hampannavar

    2013-06-01

    Full Text Available Digital water marking is a technique of embedding extra information into the multimedia content, which can be extracted to prove the copy rights. Compared to human visual system, audio system is more sensitive. As a result very few audio watermarking algorithms have been robust and imperceptible. In this paper we are implementing audio watermarking using discrete wavelet transform (DWT. Anaudio signal in the form of .wav file is decomposed into multi level DWT coefficients. A watermark signal is embedded in the final level coefficients. The audio is reconstructed from the embedded co-efficient using inverse DWT. The simulation results will be verified by comparing watermarked audio with the original audio for its perceptibility. The watermarked audio will be tested for its robustness towards retaining watermark. Computation of DWT involves large number of arithmetic operations. Hence, a hardware chip for the same would help in achieving real time performance, low power consumption and lesser area utilization. This hardware implementation through FPGA eases the integration of watermarking feature with the existing audio electronic systems.

  14. IELTS speaking instruction through audio/voice conferencing

    OpenAIRE

    Hamed Ghaemi; Hossein Khodabakhshzade; Hamid R. Kargozari

    2012-01-01

    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-testanalysiswase...

  15. 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

  16. 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

  17. Four-quadrant flyback converter for direct audio power amplification

    DEFF Research Database (Denmark)

    Ljusev, Petar; Andersen, Michael Andreas E.

    2005-01-01

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

  18. Multilevel inverter based class D audio amplifier for capacitive transducers

    DEFF Research Database (Denmark)

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

    2014-01-01

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

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

    Science.gov (United States)

    Bavaro, Sandra

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

  20. 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…

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

    NARCIS (Netherlands)

    Pocta, P.; Beerends, J.G.

    2015-01-01

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

  2. 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.

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

    OpenAIRE

    Lepa, Steffen

    2013-01-01

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

  4. Techniques in audio and acoustic measurement

    Science.gov (United States)

    Kite, Thomas D.

    2003-10-01

    Measurement of acoustic devices and spaces is commonly performed with time-delay spectrometry (TDS) or maximum length sequence (MLS) analysis. Both techniques allow an impulse response to be measured with a signal-to-noise ratio (SNR) that can be traded off against the measurement time. However, TDS suffers from long measurement times because of its linear sweep, while MLS suffers from the corruption of the impulse response by distortion. Recently a logarithmic sweep-based method has been devised which offers high SNR, short measurement times, and the ability to separate the linear impulse response from the impulse responses of distortion products. The applicability of these methods to audio and acoustic measurement will be compared.

  5. Scalar Quantization for Audio Data Coding

    CERN Document Server

    Kudryashov, Boris D; Oh, Eunmi L

    2008-01-01

    This paper is concerned with scalar quantization of transform coefficients in an audio codec. The generalized Gaussian distribution (GGD) is used as an approximation of one-dimensional probability density function for transform coefficients obtained by modulated lapped transform (MLT) or modified cosine transform (MDCT) filterbank. The rationale of the model is provided in comparison with theoretically achievable rate-distortion function. The rate-distortion function computed for the random sequence obtained from a real sequence of samples from a large database is compared with that computed for random sequence obtained by a GGD random generator. A simple algorithm of constructing the Extended Zero Zone (EZZ) quantizer is proposed. Simulation results show that the EZZ quantizer yields a negligible loss in terms of coding efficiency compared to optimal scalar quantizers. Furthermore, we describe an adaptive version of the EZZ quantizer which works efficiently with low bitrate requirements for transmitting side...

  6. Audio visual information materials for risk communication

    International Nuclear Information System (INIS)

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

  7. 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.

  8. 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.

  9. 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.

  10. Noise-Canceling Helmet Audio System

    Science.gov (United States)

    Seibert, Marc A.; Culotta, Anthony J.

    2007-01-01

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

  11. Audio-visual speech cue combination.

    Directory of Open Access Journals (Sweden)

    Derek H Arnold

    Full Text Available BACKGROUND: Different sources of sensory information can interact, often shaping what we think we have seen or heard. This can enhance the precision of perceptual decisions relative to those made on the basis of a single source of information. From a computational perspective, there are multiple reasons why this might happen, and each predicts a different degree of enhanced precision. Relatively slight improvements can arise when perceptual decisions are made on the basis of multiple independent sensory estimates, as opposed to just one. These improvements can arise as a consequence of probability summation. Greater improvements can occur if two initially independent estimates are summated to form a single integrated code, especially if the summation is weighted in accordance with the variance associated with each independent estimate. This form of combination is often described as a Bayesian maximum likelihood estimate. Still greater improvements are possible if the two sources of information are encoded via a common physiological process. PRINCIPAL FINDINGS: Here we show that the provision of simultaneous audio and visual speech cues can result in substantial sensitivity improvements, relative to single sensory modality based decisions. The magnitude of the improvements is greater than can be predicted on the basis of either a Bayesian maximum likelihood estimate or a probability summation. CONCLUSION: Our data suggest that primary estimates of speech content are determined by a physiological process that takes input from both visual and auditory processing, resulting in greater sensitivity than would be possible if initially independent audio and visual estimates were formed and then subsequently combined.

  12. 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.

  13. 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.

  14. Quality and Distortion Evaluation of Audio Signal by Spectrum

    Directory of Open Access Journals (Sweden)

    Er. Niranjan Singh

    2012-02-01

    Full Text Available 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-objects and stegano-objects [1]. A coveraudio object can be converted into a stegano-audio object via steganographic methods. In thispaper we present statistical method to detect the presence of hidden messages in audio signals.The basic idea is that, the distribution of various statistical distance measures, calculated oncover audio signals and on stegano-audio signals vis-à-vis their de-noised versions, arestatistically different. A distortion metric based on Signal spectrum was designed specifically todetect modifications and additions to audio media. We used the Signal spectrum to measure thedistortion. The distortion measurement was obtained at various wavelet decomposition levelsfrom which we derived high-order statistics as features for a classifier to determine the presenceof hidden information in an audio signal. This paper looking at evidence in a criminal caseprobably has no reason to alter any evidence files. However, it is part of an ongoing terroristsurveillance might well want to disrupt the hidden information, even if it cannot be recovered.

  15. 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.

  16. 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.

  17. The effect of reverberation on personal audio devices.

    Science.gov (United States)

    Simón-Gálvez, Marcos F; Elliott, Stephen J; Cheer, Jordan

    2014-05-01

    Personal audio refers to the creation of a listening zone within which a person, or a group of people, hears a given sound program, without being annoyed by other sound programs being reproduced in the same space. Generally, these different sound zones are created by arrays of loudspeakers. Although these devices have the capacity to achieve different sound zones in an anechoic environment, they are ultimately used in normal rooms, which are reverberant environments. At high frequencies, reflections from the room surfaces create a diffuse pressure component which is uniform throughout the room volume and thus decreases the directional characteristics of the device. This paper shows how the reverberant performance of an array can be modeled, knowing the anechoic performance of the radiator and the acoustic characteristics of the room. A formulation is presented whose results are compared to practical measurements in reverberant environments. Due to reflections from the room surfaces, pressure variations are introduced in the transfer responses of the array. This aspect is assessed by means of simulations where random noise is added to create uncertainties, and by performing measurements in a real environment. These results show how the robustness of an array is increased when it is designed for use in a reverberant environment. PMID:24815249

  18. Penguat Audio Kelas D dengan Umpan Balik Tipe Butterworth

    Directory of Open Access Journals (Sweden)

    Gunawan Dewantoro

    2016-03-01

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

  19. 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.

  20. 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.

  1. 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).

  2. High-Order Sparse Linear Predictors for Audio Processing

    DEFF Research Database (Denmark)

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

    2010-01-01

    of interesting features that make the idea of using it in audio processing not far fetched, e.g., the strong ability of modeling the spectral peaks that play a dominant role in perception. In this paper, we provide some preliminary conjectures and experiments on the use of high-order sparse linear predictors......Linear prediction has generally failed to make a breakthrough in audio processing, as it has done in speech processing. This is mostly due to its poor modeling performance, since an audio signal is usually an ensemble of different sources. Nevertheless, linear prediction comes with a whole set...

  3. Aurally Aided Visual Search Performance Comparing Virtual Audio Systems

    DEFF Research Database (Denmark)

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

    2014-01-01

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

  4. 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....

  5. Applications of Wavelets in 3-D Audio Simulation

    Institute of Scientific and Technical Information of China (English)

    2000-01-01

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

  6. 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.

  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. 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.

  9. 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.

  10. 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...

  11. 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...

  12. 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)

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

    Directory of Open Access Journals (Sweden)

    Uttam Kr. Mondal

    2011-11-01

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

  14. Fabrication of Message Digest to Authenticate Audio Signals with Alternation of Coefficients of Harmonics in Multi-Stages (MDAC)

    CERN Document Server

    Mondal, Uttam Kr

    2012-01-01

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

  15. Fabrication of Message Digest to Authenticate Audio Signals with Alternation of Coefficients of Harmonics in Multi-Stages (MDAC

    Directory of Open Access Journals (Sweden)

    Uttam Kr. Mondal

    2011-12-01

    Full Text Available Providing security to audio songs for maintaining its intellectual property right (IPR is one ofchanllenging fields in commercial world especially in creative industry. In this paper, an effectiveapproach has been incorporated to fabricate authentication of audio song through application of messagedigest method with alternation of coefficients of harmonics in multi-stages of higher frequency domainwithout affecting its audible quality. Decomposing constituent frequency components of song signal usingFourier transform with generating secret code via applying message digest followed by alternatingcoefficients of specific harmonics in multi-stages generates a secret code and this unique code is utilized todetect the originality of the song. A comparative study has been made with similar existing techniques andexperimental results are also supported with mathematical formula based on Microsoft WAVE (".wav"stereo sound file.

  16. 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.

  17. A Novel Digital Audio Watermarking Scheme in the Wavelet Domain

    Institute of Scientific and Technical Information of China (English)

    WANG Xiang-yang; YANG Hong-ying; ZHAO Hong

    2005-01-01

    We present a novel quantization-based digital audio watermarking scheme in wavelet domain. By quantizing a host audio's wavelet coefficients (Integer Lifting Wavelet Transform ) and utilizing the characteristics of human auditory system ( HAS), the gray image is embedded using our watermarking method. Experimental results show that the proposed watermarking scheme is inaudible and robust against various signal processing such as noising adding, lossy compression, low pass filtering, re-sampling, and re-quantifying.

  18. Design guidelines for audio presentation of graphs and tables

    OpenAIRE

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

    2003-01-01

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

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

    OpenAIRE

    Saeed Farahani

    1998-01-01

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

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

    OpenAIRE

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

    2010-01-01

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

  1. 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.

  2. Audio Analogue的两件旗舰产品

    Institute of Scientific and Technical Information of China (English)

    马龙辉

    2003-01-01

    @@ 几年前,意大利Audio Anakogue(雅乐)公司生产了一款名为Puccini(普契尼)的合并式放大器.由于这款放大器品质良加之推广声势浩大,在音响业的名声几乎家喻户晓,以至人们只知道Puccini而不知有Audio Analogue.

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

    OpenAIRE

    Mahon, M. S.

    2009-01-01

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

  4. 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.

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

    OpenAIRE

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

    2010-01-01

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

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

    OpenAIRE

    Cheng Ji-Shiung; Yu Pao-Ta; Tsai Hung-Hsu

    2003-01-01

    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. Fina...

  7. 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.

  8. 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.

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

    International Nuclear Information System (INIS)

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

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

    Science.gov (United States)

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

    2011-04-01

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

  11. 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.

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

    Energy Technology Data Exchange (ETDEWEB)

    Feng Yong; Peng Zhenfei; Yang Shanshan; Hong Zhiliang [State Key Laboratory of ASIC and System, Fudan University Shanghai 201203 (China); Liu Yang, E-mail: zlhong@fudan.edu.cn [Shanghai Design Center, Analog Devices, Shanghai 200021 (China)

    2011-04-15

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

  13. Clinical and Audio Vestibular Profile of Meniere's Disease in a Tertiary Care Centre in India.

    Science.gov (United States)

    Selvakumar, Paul; Balraj, Achamma; Kurien, Regi; Krishnan, Thenmozhi

    2012-12-01

    The aims of this study are to determine the frequency of patients presenting with Meniere's Disease(MD) in an Indian setting, using the American Academy of Otolaryngology-Head and Neck Surgery (AAO) diagnostic criteria, and to describe the clinical and audio vestibular profiles of these patients. The study was based on prospective case series design in the settings of a tertiary referral hospital. The study included all consecutive patients aged between 5 and 75 years presenting with the history of hearing loss, vertigo, tinnitus and or aural fullness as participants, satisfying inclusion and exclusion criteria for MD (AAO 1995) recruited over a 12 month period. Main outcome measures comprised the evaluation of epidemiological profile, clinical features, and results of audio vestibular investigations like Pure Tone Audiometry with and without glycerol, Impedance Audiometry, Electrocochleography (ECohG), Distortion Product Otoacoustic Emission and Electronystagmography (ENG). The results of the study are as follows: The frequency of MD was 15.6%, being commoner in males than females (2.6:1) and occurring more in the age group 40-49  years among males and 30-39 years among females. High frequency tinnitus was commoner than low frequency tinnitus. Extra tympanic ECohG had a positive predictive value of 76% for endolymphatic hydrops. ENG was useful for demonstrating canal paresis pattern of nystagmus in 61%. Indian patients with MD commonly present to tertiary care at the functional level scale of 3. The results of this study revealed that the frequency of MD is not as low in the Indian ENT setting as earlier believed. There is a high chance of missing cases in the routine ENT outpatient clinic setting unless a structured proforma incorporating the AAO 1995 diagnostic criteria is used. PMID:24294577

  14. 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. 

  15. Talker variability in audio-visual speech perception.

    Science.gov (United States)

    Heald, Shannon L M; Nusbaum, Howard C

    2014-01-01

    A change in talker is a change in the context for the phonetic interpretation of acoustic patterns of speech. Different talkers have different mappings between acoustic patterns and phonetic categories and listeners need to adapt to these differences. Despite this complexity, listeners are adept at comprehending speech in multiple-talker contexts, albeit at a slight but measurable performance cost (e.g., slower recognition). So far, this talker variability cost has been demonstrated only in audio-only speech. Other research in single-talker contexts have shown, however, that when listeners are able to see a talker's face, speech recognition is improved under adverse listening (e.g., noise or distortion) conditions that can increase uncertainty in the mapping between acoustic patterns and phonetic categories. Does seeing a talker's face reduce the cost of word recognition in multiple-talker contexts? We used a speeded word-monitoring task in which listeners make quick judgments about target word recognition in single- and multiple-talker contexts. Results show faster recognition performance in single-talker conditions compared to multiple-talker conditions for both audio-only and audio-visual speech. However, recognition time in a multiple-talker context was slower in the audio-visual condition compared to audio-only condition. These results suggest that seeing a talker's face during speech perception may slow recognition by increasing the importance of talker identification, signaling to the listener a change in talker has occurred. PMID:25076919

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

    Institute of Scientific and Technical Information of China (English)

    马春庭; 谭业双; 李荣祥; 杜杰

    2012-01-01

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

  17. 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.

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

    OpenAIRE

    Cano Vila, Pedro

    2007-01-01

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

  19. High Capacity Method for Real-Time Audio Data Hiding Using the FFT Transform

    Science.gov (United States)

    Fallahpour, Mehdi; Megías, David

    This paper presents a very efficient method for audio data hiding which is suitable for real-time applications. The FFT magnitudes which are in a band of frequencies between 5 and 15 kHz are modified slightly and the frequencies which have a magnitude less than a threshold are used for embedding. Its low complexity is one of the most important properties of this method making it appropriate for real-time applications. In addition, the suggested scheme is blind, since it does not need the original signal for extracting the hidden bits. The Experimental results show that it has a very good capacity (5 kbps), without significant perceptual distortion and provides robustness against MPEG compression (MP3).

  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. 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.

  2. Objective quality measurement for audio time-scale modification

    Science.gov (United States)

    Liu, Fang; Lee, Jae-Joon; Kuo, C. C. J.

    2003-11-01

    The recent ITU-T Recommendation P.862, known as the Perceptual Evaluation of Speech Quality (PESQ) is an objective end-to-end speech quality assessment method for telephone networks and speech codecs through the measurement of received audio quality. To ensure that certain network distortions will not affect the estimated subjective measurement determined by PESQ, the algorithm takes into account packet loss, short-term and long-term time warping resulted from delay variation. However, PESQ does not work well for time-scale audio modification or temporal clipping. We investigated the factors that impact the perceived quality when time-scale modification is involved. An objective measurement of time-scale modification is proposed in this research, where the cross-correlation values obtained from time-scale modification synchronization are used to evaluate the quality of a time-scaled audio sequence. This proposed objective measure has been verified by a subjective test.

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

    DEFF Research Database (Denmark)

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

    different departments: Orthopedics, Urology, Internal Medicine and Pediatrics. A total of 5,460 patients will be included from the outpatient clinics. All patients randomized to an intervention group are offered audio recording of their consultation. An Interactive Voice Response platform enables an audio...... the 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...... recording of the dialogue between the patient and the clinician via the telephone in the consultation room. This technique ensures minimal time consumption for clinicians and high sound quality. By dialing their social security number in combination with a PIN, patients can hear their consultation again...

  4. 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...

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

    CERN Document Server

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

    2013-01-01

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

  6. INFORMATION HIDING USING AUDIO STEGANOGRAPHY – A SURVEY

    Directory of Open Access Journals (Sweden)

    Jayaram P

    2011-08-01

    Full Text Available Today’s large demand of internet applications requires data to be transmitted in a secure manner. Data transmission in public communication system is not secure because of interception and improper manipulation by eavesdropper. So the attractive solution for this problem is Steganography, which is the art and science of writing hidden messages in such a way that no one, apart from the sender and intend recipient, suspects the existence of the message, a form of security through obscurity. Audio steganography is the scheme of hiding the existence of secret information by concealing it into another medium such as audio file. In this paper we mainly discuss different types of audio steganographic methods, advantages and disadvantages.

  7. 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.

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

    Directory of Open Access Journals (Sweden)

    Shingchern D. You

    2013-01-01

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

  9. Music identification system using MPEG-7 audio signature descriptors.

    Science.gov (United States)

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

    2013-01-01

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

  10. 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.

  11. Real-time Loudspeaker Distance Estimation with Stereo Audio

    DEFF Research Database (Denmark)

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

    2015-01-01

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

  12. 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 .

  13. 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

  14. An audio file tagging mobile game, mTagATune

    OpenAIRE

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

    2011-01-01

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

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

    OpenAIRE

    LATORRE PLAYÁN, JAVIER

    2015-01-01

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

  16. 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.

  17. 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.

  18. 47 CFR 25.202 - Frequencies, frequency tolerance and emission limitations.

    Science.gov (United States)

    2010-10-01

    ... GHz 24.65-24.75 GHz (6) The following frequencies are available for use by the Satellite Digital Audio... in the broadcasting-satellite service. 4 This band is shared on an equal basis with the Government... broadcasting-satellite service, and the sub-band 17.7-17.8 GHz is shared coequally with terrestrial...

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

    Science.gov (United States)

    ... audiodescription.html Question: Is audio description available for videos on MedlinePlus? To use the sharing features on ... page, please enable JavaScript. Answer: Audio description of videos helps make the content of videos accessible to ...

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

    OpenAIRE

    DiGirolamo, Vincent

    1985-01-01

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

  1. Modelling and extraction of fundamental frequency in speech signals

    OpenAIRE

    Pawi, Alipah

    2014-01-01

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

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

    OpenAIRE

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

    2001-01-01

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

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

    Science.gov (United States)

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

    2015-06-11

    The objective of the study was to assess the safety of the HyperSound® Audio System (HSS), a novel audio system using ultrasound technology, in normal hearing subjects under normal use conditions; we considered pre-exposure and post-exposure test design. We investigated primary and secondary outcome measures: i) temporary threshold shift (TTS), defined as >10 dB shift in pure tone air conduction thresholds and/or a decrement in distortion product otoacoustic emissions (DPOAEs) >10 dB at two or more frequencies; ii) presence of new-onset otologic symptoms after exposure. Twenty adult subjects with normal hearing underwent a pre-exposure assessment (pure tone air conduction audiometry, tympanometry, DPOAEs and otologic symptoms questionnaire) followed by exposure to a 2-h movie with sound delivered through the HSS emitter followed by a post-exposure assessment. No TTS or new-onset otological symptoms were identified. HSS demonstrates excellent safety in normal hearing subjects under normal use conditions. PMID:26779330

  4. The viability of speech-in-noise audiometric screening using domestic audio equipment.

    Science.gov (United States)

    Culling, John F; Zhao, Fei; Stephens, Dafydd

    2005-12-01

    Speech-in-noise audiometry has potential application as a low-cost, self-screening test for sensorineural hearing loss. To realize this potential, the influence of variations in audio equipment and listening environment need assessment. The present study assessed: 1) the frequency response and distortion produced by a wide range of commercially available audio equipment; 2) the effects of such variations upon test results with normally hearing subjects using a simple, open-set, word-identification test; 3) the effect of distortion on the speech reception threshold using digitally applied distortion; and 4) the reliability of the test in listening environments with different levels of reverberation. In addition, preliminary tests were conducted with elderly listeners. The results indicate that variations in equipment have negligible effects on speech-in-noise audiometry. The only factor that substantially elevated normally hearing listeners' thresholds was high levels of room reverberation when using loudspeaker presentation. Variations in equipment and environment thus present no significant obstacle to the development of a self-administered audiometric screening test based on speech in noise. PMID:16450920

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

    Directory of Open Access Journals (Sweden)

    Pakarinen Jyri

    2010-01-01

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

  6. 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.

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

    Directory of Open Access Journals (Sweden)

    Ritvik P. Mehta

    2015-11-01

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

  8. 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.

  9. 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.

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

    Science.gov (United States)

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

    2013-01-01

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

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

    Science.gov (United States)

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

    2008-01-01

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

  12. 47 CFR 73.9005 - Compliance requirements for covered demodulator products: Audio.

    Science.gov (United States)

    2010-10-01

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

  13. 77 FR 16890 - Second Meeting: RTCA Special Committee 226, Audio Systems and Equipment

    Science.gov (United States)

    2012-03-22

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

  14. 76 FR 79755 - First Meeting: RTCA Special Committee 226 Audio Systems and Equipment

    Science.gov (United States)

    2011-12-22

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

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

    DEFF Research Database (Denmark)

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

    2012-01-01

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

  16. 37 CFR 202.22 - Acquisition and deposit of unpublished audio and audiovisual transmission programs.

    Science.gov (United States)

    2010-07-01

    ... unpublished audio and audiovisual transmission programs. 202.22 Section 202.22 Patents, Trademarks, and... REGISTRATION OF CLAIMS TO COPYRIGHT § 202.22 Acquisition and deposit of unpublished audio and audiovisual... and copies of unpublished audio and audiovisual transmission programs by the Library of Congress...

  17. 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.

  18. 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...

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

    Science.gov (United States)

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

    2009-01-01

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

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

    Science.gov (United States)

    Gasser, P.

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

  1. 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...

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

    Directory of Open Access Journals (Sweden)

    Martin Žvelc

    2011-01-01

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

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

    Science.gov (United States)

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

    2014-01-01

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

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

    DEFF Research Database (Denmark)

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

    2008-01-01

    -mode audio power amplifiers while keeping the performance measures to excellent levels is therefore of high general interest. A modulator utilizing multiple carrier signals to generate a two level pulse train will be shown in this paper. The performance of the modulator will be compared in simulation...

  6. 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.

  7. 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...

  8. 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.

  9. 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.

  10. 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…

  11. 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...

  12. 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...

  13. 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…

  14. 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…

  15. 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...

  16. 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

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

    NARCIS (Netherlands)

    Zee, van der R.A.R.; Tuijl, van A.J.M.

    1998-01-01

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

  18. 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.

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

    CERN Document Server

    Lerch, Alexander

    2012-01-01

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

  20. Subjective audio quality evaluation of embedded-optimization-based distortion precompensation algorithms.

    Science.gov (United States)

    Defraene, Bruno; van Waterschoot, Toon; Diehl, Moritz; Moonen, Marc

    2016-07-01

    Subjective audio quality evaluation experiments have been conducted to assess the performance of embedded-optimization-based precompensation algorithms for mitigating perceptible linear and nonlinear distortion in audio signals. It is concluded with statistical significance that the perceived audio quality is improved by applying an embedded-optimization-based precompensation algorithm, both in case (i) nonlinear distortion and (ii) a combination of linear and nonlinear distortion is present. Moreover, a significant positive correlation is reported between the collected subjective and objective PEAQ audio quality scores, supporting the validity of using PEAQ to predict the impact of linear and nonlinear distortion on the perceived audio quality. PMID:27475197

  1. Consequences of outer hair cell damage for otoacoustic emissions and audio-vocal feedback in the mustached bat.

    Science.gov (United States)

    Kössl, M; Vater, M

    2000-12-01

    The cochlea of the mustached bat is adapted to process ultrasonic echolocation signals. To assess the involvement of active sound amplification by outer hair cells in high-frequency hearing and in audio-vocal interaction, selective hair cell damage was induced by the antibiotic Amikacin. Amikacin preferentially damaged the first row of outer hair cells in the basal cochlear turn. The cochlear regions coding for the second-harmonic constant-frequency component of the echolocation call (CF2) at 61 kHz and for frequencies between 75 and 100 kHz were the most affected. This was reflected in an increase of mechanical thresholds obtained by measuring distortion-product otoacoustic emissions. During initial periods of minor hair cell damage, when thresholds had deteriorated by less than 40 dB, a sharp, mechanical, cochlear resonance that is responsible for enhanced tuning to 61 kHz was still measurable as a stimulus-frequency otoacoustic emission and its frequency decreased by 350 Hz. The persistence of the resonance suggests that passive structures like the tectorial or basilar membrane are important for generation of the resonance. Behaviorally, the bats reacted to the change in cochlear micromechanics with a decrease of their CF2 frequency by 360 Hz. After larger hair cell damage, when the cochlear resonance had disappeared, the bats vocalized only sparsely and the CF2 frequency increased by up to 2 kHz, which may correspond to a state without audiovocal feedback. This indicates that audio-vocal feedback in the nondamaged animal works to lower the call frequency. PMID:11547810

  2. 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.

  3. 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.

  4. 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.

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

    International Nuclear Information System (INIS)

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

  6. Dual Function of Audio Watermarking Algorithm in Hybrid Domain%基于混合域的双功能音频水印算法

    Institute of Scientific and Technical Information of China (English)

    杨志疆; 叶阿勇

    2015-01-01

    A dual function of watermark scheme in hybrid domain is proposed ,which can realize both the digital audio copyright protection and content authentication . Firstly , the original digital audio is decomposed into many segments according to the size of the watermark . Secondly , the DWT is performed on each audio seg‐ment .Finally ,robust watermark is embedded in the low‐frequency component while the fragile watermark is embedded in the high frequency component .Independent blind detection was achieved for both robust and frag‐ile watermarks .Simulation experiments show that the robust watermark can be used for audio copyright pro‐tection ,because the signal has strong robustness in common audio signal processing ;while the fragile water‐mark is sensitive to common signal processing operations , and may have accurate location for tampered area .%本文提出一种基于混合域的双功能水印算法,能同时实现对数字音频的版权保护与内容认证。该算法根据水印的大小将原数字音频分段,对每一段进行多级DW T变换,并选取低频分量嵌入鲁棒水印,高频分量嵌入脆弱水印,检测时两种水印独立盲提取。仿真实验表明,此算法中的鲁棒水印信号在常见的音频信号处理中具有强鲁棒性,可用于音频版权保护;而脆弱水印信号不仅对常见的信号处理操作敏感,而且能对篡改区域进行较准确的定位。

  7. Audio Linguistic Disorders in Autistic Children

    International Nuclear Information System (INIS)

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

  8. “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.

  9. Digital Audio Watermarking Using Psychoacoustic Model and CDMA Modulation

    Directory of Open Access Journals (Sweden)

    Wahid Barkouti

    2011-09-01

    Full Text Available DIGITAL WATERMARKING IS USED TO INSERT INFORMATION (A SIGNATURE IN A COMPUTER DOCUMENT. THE ADDITION OF THE SIGNATURE MUST BE IMPERCEPTIBLE AND UNDETECTABLE BY ANY SYSTEM IGNORING ITS MODE OF INSERTION. IN PARTICULAR, IT MUST BE COMPLETELY INVISIBLE TO THE HUMAN EYE. THIS METHOD IS DIFFERENT FROM CRYPTOGRAPHY, WHICH HIDES A MESSAGE BY MAKING IT UNREADABLE. DIGITAL WATERMARKING NOW INCLUDES OTHER DATA WITHIN MUSIC SIGNALS. AUDIO WATERMARKING CONSISTS IN EMBEDDING INAUDIBLE INFORMATION IN AN AUDIO SIGNAL. THE WATERMARKING SYSTEM MUST GUARANTEE ONE TRANSMISSION OF THE INAUDIBLE, RELIABLE AND ROBUST INFORMATION FACE A SET OF DISRUPTIONS. IN THIS GOAL, WE PROPOSE A NEW STRATEGY OF INSERTION ADAPTED TO A WATERMARKING SYSTEM. THIS STRATEGY PERMITS TO CONSTRUCT AN INAUDIBLE WATERMARKING AND OF MAXIMAL HARDINESS TO THE ADDITION OF A NOISE.

  10. DIGITAL AUDIO WATERMARKING USING PSYCHOACOUSTIC MODEL AND CDMA MODULATION

    Directory of Open Access Journals (Sweden)

    Wahid Barkouti

    2011-06-01

    Full Text Available DIGITAL WATERMARKING IS USED TO INSERT INFORMATION (A SIGNATURE IN A COMPUTER DOCUMENT. THE ADDITION OF THE SIGNATURE MUST BE IMPERCEPTIBLE AND UNDETECTABLE BY ANY SYSTEM IGNORING ITS MODE OF INSERTION. IN PARTICULAR, IT MUST BE COMPLETELY INVISIBLE TO THE HUMAN EYE. THIS METHOD IS DIFFERENT FROM CRYPTOGRAPHY, WHICH HIDES A MESSAGE BY MAKING IT UNREADABLE. DIGITAL WATERMARKING NOW INCLUDES OTHER DATA WITHIN MUSIC SIGNALS. AUDIO WATERMARKING CONSISTS IN EMBEDDING INAUDIBLE INFORMATION IN AN AUDIO SIGNAL. THE WATERMARKING SYSTEM MUST GUARANTEE ONE TRANSMISSION OF THE INAUDIBLE, RELIABLE AND ROBUST INFORMATION FACE A SET OF DISRUPTIONS. IN THIS GOAL, WE PROPOSE A NEW STRATEGY OF INSERTION ADAPTED TO A WATERMARKING SYSTEM. THIS STRATEGY PERMITS TO CONSTRUCT AN INAUDIBLE WATERMARKING AND OF MAXIMAL HARDINESS TO THE ADDITION OF A NOISE.

  11. Quantization Audio Watermarking with Optimal Scaling on Wavelet Coefficients

    CERN Document Server

    Chen, S -T; Tu, S -Y

    2011-01-01

    In recent years, discrete wavelet transform (DWT) provides an useful platform for digital information hiding and copyright protection. Many DWT-based algorithms for this aim are proposed. The performance of these algorithms is in term of signal-to-noise ratio (SNR) and bit-error-rate (BER) which are used to measure the quality and the robustness of an embedded audio. However, there is a tradeoff relationship between the embedded-audio quality and robustness. The tradeoff relationship is a signal processing problem in the wavelet domain. To solve this problem, this study presents an optimization-based scaling scheme using optimal multi-coefficients quantization in the wavelet domain. Firstly, the multi-coefficients quantization technique is rewritten as an equation with arbitrary scaling on DWT coefficients and set SNR to be a performance index. Then, a functional connecting the equation and the performance index is derived. Secondly, Lagrange Principle is used to obtain the optimal solution. Thirdly, the scal...

  12. 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.

  13. Underdetermined Blind Audio Source Separation Using Modal Decomposition

    Directory of Open Access Journals (Sweden)

    Aïssa-El-Bey Abdeldjalil

    2007-01-01

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

  14. Underdetermined Blind Audio Source Separation Using Modal Decomposition

    Directory of Open Access Journals (Sweden)

    Abdeldjalil Aïssa-El-Bey

    2007-03-01

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

  15. Audio-haptic interaction in simulated walking experiences

    DEFF Research Database (Denmark)

    Serafin, Stefania

    2011-01-01

    In this paper an overview of the work conducted on audio-haptic physically based simulation and evaluation of walking is provided. This work has been performed in the context of the Natural Interactive Walking (NIW) project, whose goal is to investigate possibilities for the integrated and interc......In this paper an overview of the work conducted on audio-haptic physically based simulation and evaluation of walking is provided. This work has been performed in the context of the Natural Interactive Walking (NIW) project, whose goal is to investigate possibilities for the integrated...... and 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...... and haptic feedback in walking tasks....

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

    DEFF Research Database (Denmark)

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

    2014-01-01

    , 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......It was recently shown that delta-sigma quantization (DSQ) can be used for optimal multiple description (MD) coding of Gaussian sources. The DSQ scheme combined oversampling, prediction, and noise-shaping in order to trade off side distortion for central distortion in MD coding. It was shown...... that asymptotically in the dimensions of the resampling, prediction, and noise-shaping filters as well as asymptotically in the quantizer dimensions, all rate-distortion points on the symmetric quadratic Gaussian MD rate-distortion function could be achieved. In this work, we show that this somewhat theoretical...

  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. 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.

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

    Directory of Open Access Journals (Sweden)

    Saeed Farahani

    1998-03-01

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

  20. 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.

  1. 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.

  2. 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.

  3. 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.

  4. Comparing Audio Features and Playlist Statistics for Music Classification

    OpenAIRE

    Vatolkin, Igor; Bonnin, Geoffray; Jannach, Dietmar

    2014-01-01

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

  5. 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....

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

    OpenAIRE

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

    2012-01-01

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

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

    Science.gov (United States)

    Chapin, William L.; Foster, Scott

    1992-06-01

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

  8. Random Numbers Generated from Audio and Video Sources

    OpenAIRE

    I-Te Chen

    2013-01-01

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

  9. Audio Processing Solution for Video Conference Based Aerobics

    OpenAIRE

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

    2010-01-01

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

  10. Audio-visual speech perception: a developmental ERP investigation

    OpenAIRE

    Knowland, V.; Mercure, E.; Karmiloff-Smith, A; Dick, F; Thomas, M.

    2014-01-01

    Being able to see a talking face confers a considerable advantage for speech perception in adulthood. However, behavioural data currently suggest that children fail to make full use of these available visual speech cues until age 8 or 9. This is particularly surprising given the potential utility of multiple informational cues during language learning. We therefore explored this at the neural level. The event-related potential (ERP) technique has been used to assess the mechanisms of audio-vi...

  11. Audio Message Transmitter Secured Through Elliptical Curve Cryptosystem

    OpenAIRE

    Luma, Artan; Selimi, Besnik; Ameti, Lirim

    2014-01-01

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

  12. 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.

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

    OpenAIRE

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

    2010-01-01

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

  14. Investigating Lay Design through Prototyping a Toolkit for Audio Equipment

    OpenAIRE

    Hermans, Guido

    2015-01-01

    Post-industrial design challenges the current way of design and production of consumer products. Leveraging from 3D printing and the potential of layperson involvement in the design process, we investigated how the relation between the professional designer and layperson might change in a democratized design process, where the layperson is an active participant, mediated by toolkit software. In this research through design study we examined this relation by prototyping a toolkit for audio equ...

  15. Efficiency of low power audio amplifiers and loudspeakers

    OpenAIRE

    Burrow, SG; Grant, Duncan A

    2001-01-01

    In this paper we look at the load presented to audio amplifiers by real transducers. We consider the power losses in Class-AB and Class-D amplifier topologies, and determine that in order to predict efficiency it is necessary to consider the amplifier/transducer combination. The ability of the class-D amplifier to recycle quadrature load current offers new ways to improve efficiency.

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

    Science.gov (United States)

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

    2002-05-01

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

  17. 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.

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

    Science.gov (United States)

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

    2003-04-01

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

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

    CERN Document Server

    Vogel, Burkhard

    2013-01-01

    The 34 chapters of the 2nd edition of How to Gain Gain give a detailed insight into a collection (54) of the most common gain producing, constant current generating possibilities, and electronic noise creation of triodes for audio pre-amplifier purposes. These chapters also offer complete sets of formulae to calculate gain, frequency and phase responses, and signal-to-noise ratios of certain building blocks built-up with this type of vacuum valve (tube). In all cases detailed derivations of the gain formulae are also presented. All what is needed are the data sheet valve characteristic figures of the triode's mutual conductance, the gain factor and the internal plate (anode) resistance. To calculate frequency and phase responses of gain stages the different data sheet based input and output capacitances have to be taken into account too. To calculate transfer functions and signal-to-noise ratios for any kind of triode driven gain stage, including all its bias setting, frequency, phase, and electronic noise in...

  20. A robust zero-watermarking algorithm for audio%一种鲁棒音频零水印算法

    Institute of Scientific and Technical Information of China (English)

    崔得龙; 左敬龙; 彭志平

    2011-01-01

    为了实现数字音频的版权保护,根据音频信号的短时平稳特性和离散小波变换的多分辨率分析特性,设计了一种基于范重心和提升小波变换的数字音频零水印算法.算法首先将音频信号进行分帧,其次对分帧音频进行三级小波提升,提取低频近似分量的范重心,并根据分量范重心与均值向量之间的关系生成特征向量,最后将特征向量与水印运算得到代表原始音频的版权信息.实验结果表明,该算法对音频信号遭受的常见攻击具有较强的鲁棒性,同时密钥的使用保证了算法的安全性.%According to the transient steady property of audio signal and multi-resolution analysis property of discrete wavelet transform, an audio zero watermarking scheme based on NCG (Normed Centre of Gravity) and lifting-based wavelet is proposed in this paper to achieve the copyright protection of digital audio. Firstly, the audio signal is framed by equalized length. Secondly, wavelet transformation based on three-level lifting for the framed audio is conducted, and then NCGS are detected from the low frequency components, then the feature vector is generated by taking relationship between NCGS and mean value vector. Finally, the copyright information is obtained by calculating the watermark and feature vector. Experimental results show that the scheme is robust against common signal processing attacks, meanwhile security of the algorithm is guaranteed by using secret keys.

  1. 基于FPGA的音频分析系统实现%Implementation of Audio Analysis System Based on FPGA

    Institute of Scientific and Technical Information of China (English)

    黄扬帆; 王璐; 甘平; 黄发

    2011-01-01

    采用Altera公司的FPGA芯片EP2C20作为控制和运算核心,实现对频率范围在20 Hz~20 kHz的音频信号分析和测试.采用FFT算法对音频信号进行频谱分析和计算处理,实现失真度分析,并完成信号总功率、各分量频率、周期等参数分析测量.采用TFT-LCD作为显示终端,能够直观显示功率谱曲线和各测试参数值.通过软件分析对比测试发现,该系统实时性强,测量准确度高,误差优于0.3%.可以有效完成音频信号分析和处理.%Altera's FPGA chip EP2C20 is used as the control and operation core to analyse and test the audio signal in the frequency range of 20 Hz ~20 kHz. FFT algorithm used for spectral analysis and computation of audio signal,distortion analysis,analysing and measuring the signal's total power,frequency of each component,period and other parameters. TFT-LCD is used as its display terminal so as to get a directviewing show of power spectrum and test parameters. By software analysis and comparison test,it is proved that this system is strong real-time,with high measurement accuracy and less than 0. 3% error. The audio signal analysis and processing can be completed effectively.

  2. 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.

  3. 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.

  4. 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.

  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. 音频数据加密的二值音频可听密码方案%An Audio Cryptography Scheme for Binary Audio

    Institute of Scientific and Technical Information of China (English)

    张选平; 王旭; 邵利平; 赵仲孟

    2012-01-01

    The characteristics of digital audio such as a large quantity of data, strong correlation and high redundancy, make it difficult to meet real-time security to encrypt digital audio by using conventional encryption algorithms. An audio cryptography scheme based on binary audio is proposed by using the auditory properties of human being. In order to reduce the amount of audio data, the secret audio is transformed into binary audio by halftone as in the image process. The basic array of (k, n) threshold in visual cryptography scheme is used to share the binary audio so that correlations in audio data is removed. The secret audio information can be directly decrypted through human auditory system by synchronously playing any k shares of the binary audio, but synchronously playing any m(maudio by the proposed scheme has high randomicity and security. Comparisons with original secret audio show that although the decrypted secret voice has some distortion, its content is still natural and comprehensible.%数字音频文件数据量大、相关性强、冗余度高,传统加密算法难以满足其实时安全性要求.针对此问题,根据人耳听觉特性提出了一种基于二值音频的可听密码方案.该方案借鉴图像半色调技术对秘密音频数据进行二值化处理,以降低音频信息的数据量;结合可视密码技术的(k,n)门限方案的基本阵对二值数字音频进行分存和加密,从而破坏了原始音频数据之间的相关性.在解密时只需同步播放任意k份秘密音频就可直接通过人耳听觉系统解密,少于k份的任意秘密音频同步播放不会暴露原始音频的任何信息.实验结果表明,加密后的音频具有良好的随机性和安全性.与原始秘密音频相比,恢复后的秘密语音虽有一些失真,但其语音内容易于辨识、可懂.

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

    OpenAIRE

    2012-01-01

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

  8. 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....

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

    OpenAIRE

    Ljusev, Petar; Andersen, Michael Andreas E.

    2005-01-01

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

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

    Science.gov (United States)

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

    1996-01-01

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

  11. 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.

  12. 蓝牙音频网关的实现%Realization of Bluetooth audio gateway

    Institute of Scientific and Technical Information of China (English)

    刘建添

    2013-01-01

      在CSR公司的蓝牙芯片BC352239A上,利用Bluelab3.52开发软件,实现了音频网关(Audio Gateway),可以将普通音频接口转成蓝牙通信。%  In the CSR Bluetooth chip BC352239A, using Bluelab3.52 to develop software, realize the audio gateway (Audio Gateway), can be converted into ordinary audio interface bluetooth communication.

  13. 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

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

    Science.gov (United States)

    Fourer, Dominique; Marchand, Sylvain

    2013-12-01

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

  15. 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.

  16. 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。

  17. 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.

  18. Tools for signal compression applications to speech and audio coding

    CERN Document Server

    Moreau, Nicolas

    2013-01-01

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

  19. 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

  20. DOA Estimation of Audio Sources in Reverberant Environments

    DEFF Research Database (Denmark)

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

    2016-01-01

    estimate the direction-of-arrival (DOA) of both the direct path of the audio source and the early reflections. In these methods, the contribution of the early reflections is essentially subtracted from the signal observations before localization of the direct path component, which may reduce the estimation...... bias. Our simulation results show that we can estimate the DOA of the desired signal more accurately with this procedure compared to state-of-theart estimator in both synthetic and real data experiments with reverberation....

  1. 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; ·     �...

  2. 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.

  3. 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.

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

    Science.gov (United States)

    Feldbauer, Christian; Kubin, Gernot; Kleijn, W. Bastiaan

    2005-12-01

    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.

  5. 基于DWT和DCT的自适应双重音频水印%Adaptive dual audio watermarking based on DWT and DCT

    Institute of Scientific and Technical Information of China (English)

    任克强; 李慧; 谢斌

    2013-01-01

    Aiming at the problem of digital audio copyright protection and content authentication,this article proposed an adaptive dual audio watermarking algorithm for copyright protection and content authentication.The algorithm carried on DWT and DCT for each segment of original audio signal.It extracted low frequency coefficients and middle-low frequency coefficients in hybrid transform domain respectively as watermarking structure and embedding domain,structured zero-watermarking and embedded semi-fragile watermarking adaptively,and realized audio copyright protection and content authentication.The experimental results show that the imperceptibility of the algorithm is very good,it has good robustness against many kinds of attacks,and it is able to locate the malicious tampering.%针对数字音频版权保护和内容认证的问题,提出了一种用于版权保护和内容认证的自适应双重音频水印算法.算法对每段原始音频信号进行离散小波变换和离散余弦变换,分别提取混合变换域的低频系数和中低频系数作为水印的构造和嵌入域,利用音频载体的统计特性自适应地构造零水印和嵌入半脆弱水印,实现音频版权保护和内容认证.实验结果表明,该算法的不可感知性很好,在抵抗多种攻击时具有良好的鲁棒性,并且能够对恶意窜改进行定位.

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

    Science.gov (United States)

    Merry, Stephen; Orsmond, Paul

    2008-01-01

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

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

    Science.gov (United States)

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

    2010-01-01

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

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

    Science.gov (United States)

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

    2013-01-01

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

  10. 76 FR 591 - Determination of Rates and Terms for Preexisting Subscription and Satellite Digital Audio Radio...

    Science.gov (United States)

    2011-01-05

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

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

    Science.gov (United States)

    Sha, Guoquan

    2010-01-01

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

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

    Science.gov (United States)

    Weld, Christopher

    2014-01-01

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

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

    Science.gov (United States)

    Bauer, Sara

    2011-01-01

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

  14. 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 ...

  15. 78 FR 18416 - Sixth Meeting: RTCA Special Committee 226, Audio Systems and Equipment

    Science.gov (United States)

    2013-03-26

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

  16. 77 FR 37733 - Third Meeting: RTCA Special Committee 226, Audio Systems and Equipment

    Science.gov (United States)

    2012-06-22

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

  17. 78 FR 57673 - Eighth Meeting: RTCA Special Committee 226, Audio Systems and Equipment

    Science.gov (United States)

    2013-09-19

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

  18. 78 FR 38093 - Seventh Meeting: RTCA Special Committee 226, Audio Systems and Equipment

    Science.gov (United States)

    2013-06-25

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

  19. 77 FR 58209 - Fourth Meeting: RTCA Special Committee 226, Audio Systems and Equipment

    Science.gov (United States)

    2012-09-19

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

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

    NARCIS (Netherlands)

    Harma, A.S.

    2011-01-01

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