Selective attention modulates the direction of audio-visual temporal recalibration.

Science.gov (United States)

Ikumi, Nara; Soto-Faraco, Salvador

2014-01-01

Temporal recalibration of cross-modal synchrony has been proposed as a mechanism to compensate for timing differences between sensory modalities. However, far from the rich complexity of everyday life sensory environments, most studies to date have examined recalibration on isolated cross-modal pairings. Here, we hypothesize that selective attention might provide an effective filter to help resolve which stimuli are selected when multiple events compete for recalibration. We addressed this question by testing audio-visual recalibration following an adaptation phase where two opposing audio-visual asynchronies were present. The direction of voluntary visual attention, and therefore to one of the two possible asynchronies (flash leading or flash lagging), was manipulated using colour as a selection criterion. We found a shift in the point of subjective audio-visual simultaneity as a function of whether the observer had focused attention to audio-then-flash or to flash-then-audio groupings during the adaptation phase. A baseline adaptation condition revealed that this effect of endogenous attention was only effective toward the lagging flash. This hints at the role of exogenous capture and/or additional endogenous effects producing an asymmetry toward the leading flash. We conclude that selective attention helps promote selected audio-visual pairings to be combined and subsequently adjusted in time but, stimulus organization exerts a strong impact on recalibration. We tentatively hypothesize that the resolution of recalibration in complex scenarios involves the orchestration of top-down selection mechanisms and stimulus-driven processes.

Selective attention modulates the direction of audio-visual temporal recalibration.

Directory of Open Access Journals (Sweden)

Nara Ikumi

Full Text Available Temporal recalibration of cross-modal synchrony has been proposed as a mechanism to compensate for timing differences between sensory modalities. However, far from the rich complexity of everyday life sensory environments, most studies to date have examined recalibration on isolated cross-modal pairings. Here, we hypothesize that selective attention might provide an effective filter to help resolve which stimuli are selected when multiple events compete for recalibration. We addressed this question by testing audio-visual recalibration following an adaptation phase where two opposing audio-visual asynchronies were present. The direction of voluntary visual attention, and therefore to one of the two possible asynchronies (flash leading or flash lagging, was manipulated using colour as a selection criterion. We found a shift in the point of subjective audio-visual simultaneity as a function of whether the observer had focused attention to audio-then-flash or to flash-then-audio groupings during the adaptation phase. A baseline adaptation condition revealed that this effect of endogenous attention was only effective toward the lagging flash. This hints at the role of exogenous capture and/or additional endogenous effects producing an asymmetry toward the leading flash. We conclude that selective attention helps promote selected audio-visual pairings to be combined and subsequently adjusted in time but, stimulus organization exerts a strong impact on recalibration. We tentatively hypothesize that the resolution of recalibration in complex scenarios involves the orchestration of top-down selection mechanisms and stimulus-driven processes.

An Interactive Concert Program Based on Infrared Watermark and Audio Synthesis

Science.gov (United States)

Wang, Hsi-Chun; Lee, Wen-Pin Hope; Liang, Feng-Ju

The objective of this research is to propose a video/audio system which allows the user to listen the typical music notes in the concert program under infrared detection. The system synthesizes audio with different pitches and tempi in accordance with the encoded data in a 2-D barcode embedded in the infrared watermark. The digital halftoning technique has been used to fabricate the infrared watermark composed of halftone dots by both amplitude modulation (AM) and frequency modulation (FM). The results show that this interactive system successfully recognizes the barcode and synthesizes audio under infrared detection of a concert program which is also valid for human observation of the contents. This interactive video/audio system has greatly expanded the capability of the printout paper to audio display and also has many potential value-added applications.

Self-oscillating modulators for direct energy conversion audio power amplifiers

DEFF Research Database (Denmark)

Ljusev, Petar; Andersen, Michael Andreas E.

2005-01-01

Direct energy conversion audio power amplifier represents total integration of switching-mode power supply and Class D audio power amplifier into one compact stage, achieving high efficiency, high level of integration, low component count and eventually low cost. This paper presents how self-oscillating...

Spread Spectrum Modulation by Using Asymmetric-Carrier Random PWM

DEFF Research Database (Denmark)

Mathe, Laszlo; Lungeanu, Florin; Sera, Dezso

2012-01-01

is very effective and is independent from the modulation index. The flat motor current spectrum generates an acoustical noise close to the white noise, which improves the acoustical performance of the drive. The new carrier wave is easy to implement digitally, without employing any external circuits...

Perancangan Sistem Audio Mobil Berbasiskan Sistem Pakar dan Web

Directory of Open Access Journals (Sweden)

Djunaidi Santoso

2011-12-01

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

Nonspeech audio in user interfaces for TV

NARCIS (Netherlands)

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

1997-01-01

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

Modeling of carrier transport in multi-quantum-well p-i-n modulators

DEFF Research Database (Denmark)

Højfeldt, Sune; Mørk, Jesper

2002-01-01

The dynamical properties of InGaAsP multi-quantum-well electroabsorption modulators are investigated using a comprehensive numerical device model. We calculate the time-dependent sweep-out of photo-generated carriers and the corresponding time-dependent absorption change. The sweep-out is influen......The dynamical properties of InGaAsP multi-quantum-well electroabsorption modulators are investigated using a comprehensive numerical device model. We calculate the time-dependent sweep-out of photo-generated carriers and the corresponding time-dependent absorption change. The sweep......-out is influenced by carriers being recaptured into subsequent wells as they move towards the contacts. This process drastically increases the sweep-out time in our ten-well structure (similar to25 ps) compared to the pure drift-time (similar to1 ps). We also compare the saturation properties of two components...

Double-Carrier Phase-Disposition Pulse Width Modulation Method for Modular Multilevel Converters

DEFF Research Database (Denmark)

Zhou, Fayun; Luo, An; Li, Yan

2017-01-01

Modular multilevel converters (MMCs) have become one of the most attractive topologies for high-voltage and high-power applications. A double-carrier phase disposition pulse width modulation (DCPDPWM) method for MMCs is proposed in this paper. Only double triangular carriers with displacement ang......, the proposed method and theoretical analysis are verified by simulation and experimental results. View Full-Text...

Large Modulation of Charge Carrier Mobility in Doped Nanoporous Organic Transistors

Energy Technology Data Exchange (ETDEWEB)

Zhang, Fengjiao [Department of Chemical and Biomolecular Engineering, University of Illinois Urbana-Champaign, 600 S. Mathews Ave. Urbana IL 61801 USA; Dai, Xiaojuan [Beijing National Laboratory for Molecular Sciences, Institute of Chemistry, Chinese Academy of Sciences, Beijing 100190 P. R. China; Zhu, Weikun [Department of Chemical and Biomolecular Engineering, University of Illinois Urbana-Champaign, 600 S. Mathews Ave. Urbana IL 61801 USA; Chung, Hyunjoong [Department of Chemical and Biomolecular Engineering, University of Illinois Urbana-Champaign, 600 S. Mathews Ave. Urbana IL 61801 USA; Diao, Ying [Department of Chemical and Biomolecular Engineering, University of Illinois Urbana-Champaign, 600 S. Mathews Ave. Urbana IL 61801 USA

2017-05-10

Molecular doping of organic electronics has shown promise to sensitively modulate important device metrics. One critical challenge is the disruption of structure order upon doping of highly crystalline organic semiconductors, which significantly reduces the charge carrier mobility. This paper demonstrates a new method to achieve large modulation of charge carrier mobility via channel doping without disrupting the molecular ordering. Central to the method is the introduction of nanopores into the organic semiconductor thin films via a simple and robust templated meniscus-guided coating method. Using this method, the charge carrier mobility of C8-benzothieno[3,2-b]benzothiophene transistors is boosted by almost sevenfold. This paper further demonstrates enhanced electron transport by close to an order of magnitude in a diketopyrrolopyrrole-based donor–acceptor polymer. Combining spectroscopic measurements, density functional theory calculations, and electrical characterizations, the doping mechanism is identified as partial-charge-transfer induced trap filling. The nanopores serve to enhance the dopant/organic semiconductor charge transfer reaction by exposing the π-electrons to the pore wall.

An assessment of envelope-based demodulation in case of proximity of carrier and modulation frequencies

Science.gov (United States)

Shahriar, Md Rifat; Borghesani, Pietro; Randall, R. B.; Tan, Andy C. C.

2017-11-01

Demodulation is a necessary step in the field of diagnostics to reveal faults whose signatures appear as an amplitude and/or frequency modulation. The Hilbert transform has conventionally been used for the calculation of the analytic signal required in the demodulation process. However, the carrier and modulation frequencies must meet the conditions set by the Bedrosian identity for the Hilbert transform to be applicable for demodulation. This condition, basically requiring the carrier frequency to be sufficiently higher than the frequency of the modulation harmonics, is usually satisfied in many traditional diagnostic applications (e.g. vibration analysis of gear and bearing faults) due to the order-of-magnitude ratio between the carrier and modulation frequency. However, the diversification of the diagnostic approaches and applications shows cases (e.g. electrical signature analysis-based diagnostics) where the carrier frequency is in close proximity to the modulation frequency, thus challenging the applicability of the Bedrosian theorem. This work presents an analytic study to quantify the error introduced by the Hilbert transform-based demodulation when the Bedrosian identity is not satisfied and proposes a mitigation strategy to combat the error. An experimental study is also carried out to verify the analytical results. The outcome of the error analysis sets a confidence limit on the estimated modulation (both shape and magnitude) achieved through the Hilbert transform-based demodulation in case of violated Bedrosian theorem. However, the proposed mitigation strategy is found effective in combating the demodulation error aroused in this scenario, thus extending applicability of the Hilbert transform-based demodulation.

Dual-Modulation, Dual-Wavelength, Optical Polarimetry System for Glucose Monitoring

Science.gov (United States)

2016-08-26

bandwidth transimpedance amplifier (CVI Melles Griot, Albuquerque, NM, USA) and fed into two lock-in amplifiers (Stanford Research Systems, Sunnyvale...were modulated using synchronous sinusoidal signals and linear amplifiers (carrier frequencies: 1 2=45 82c cf kHz f kHz， ). The sinusoidal signals...ferrite core was powered with an audio amplifier (Radio Shack, Fort Worth, TX, USA) connected in series with a 0.18 µF capacitor in order to achieve

Electrical system for pulse-width modulated control of a power inverter using phase-shifted carrier signals and related operating methods

Science.gov (United States)

Welchko, Brian A [Torrance, CA

2012-02-14

Systems and methods are provided for pulse-width modulated control of power inverter using phase-shifted carrier signals. An electrical system comprises an energy source and a motor. The motor has a first set of windings and a second set of windings, which are electrically isolated from each other. An inverter module is coupled between the energy source and the motor and comprises a first set of phase legs coupled to the first set of windings and a second set of phase legs coupled to the second set of windings. A controller is coupled to the inverter module and is configured to achieve a desired power flow between the energy source and the motor by modulating the first set of phase legs using a first carrier signal and modulating the second set of phase legs using a second carrier signal. The second carrier signal is phase-shifted relative to the first carrier signal.

A second-order class-D audio amplifier

OpenAIRE

Cox, Stephen M.; Tan, M.T.; Yu, J.

2011-01-01

Class-D audio amplifiers are particularly efficient, and this efficiency has led to their ubiquity in a wide range of modern electronic appliances. Their output takes the form of a high-frequency square wave whose duty cycle (ratio of on-time to off-time) is modulated at low frequency according to the audio signal. A mathematical model is developed here for a second-order class-D amplifier design (i.e., containing one second-order integrator) with negative feedback. We derive exact expression...

Comparison of Simple Self-Oscillating PWM Modulators

DEFF Research Database (Denmark)

Dahl, Nicolai J.; Iversen, Niels Elkjær; Knott, Arnold

2016-01-01

Switch-mode power amplifiers has become the conventional choice for audio applications due to their superior efficiency and excellent audio performance. These amplifiers rely on high frequency modulation of the audio input. Conventional modulators use a fixed high frequency for modulation. Self......-oscillating modulators do not have a fixed modulation frequency and can provide good audio performance with very simple circuitry. This paper proposes a new type of self-oscillating modulator. The proposed modulator is compared to an already existing modulator of similar type and their performances are compared both...... theoretically and experimentally. The result shows that the proposed modulator provides a higher degree of linearity resulting in around 2% lower Total Harmonic Distortion (THD)....

12.5 Gb/s carrier-injection silicon Mach—Zehnder optical modulator

International Nuclear Information System (INIS)

Chen Hongtao; Ding Jianfeng; Yang Lin

2012-01-01

We demonstrate a 12.5 Gb/s carrier-injection silicon Mach—Zehnder optical modulator. Under a nonreturn-zero (NRZ) pre-emphasized electrical drive signal with voltage swing of 6.3 V and forward bias of 0.7 V, the eye is clearly opened with an extinction ratio of 8.4 dB. The device exhibits high modulation efficiency, with a figure of merit V π L of 0.036 V·mm. (semiconductor devices)

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

DEFF Research Database (Denmark)

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

2007-01-01

In general, the temporal structure of stimuli must be considered to account for certain observations made in detection and masking experiments in the audio-frequency domain. Two such phenomena are (1) a heightened sensitivity to amplitude increments with a temporal fringe compared to gated level......-frequency domain. Pure-tone carrier amplitude-modulation (AM) depth-discrimination thresholds were found to be similar using both traditional gated stimuli and using a temporally modulated fringe for a fixed standard depth (ms=0.25) and a range of AM frequencies (4-64 Hz). In a second experiment, masked sinusoidal...... AM detection thresholds were compared in conditions with and without slow and regular fluctuations imposed on the instantaneous masker AM depth. Release from masking was obtained only for very slow masker fluctuations (less than 2 Hz). A physiologically motivated model that effectively acts...

PWM pulse pattern optimization method using carrier frequency modulation. Carrier shuhasu hencho ni yoru PWM pulse pattern saitekikaho

Energy Technology Data Exchange (ETDEWEB)

Iwaji, Y.; Fukuda, S. (Hokkaido University, Sapporo (Japan))

1991-07-15

Sinusoidal inverters are getting more widely used keeping pace with the development of semiconductor switching elements. This paper discusses optimizing a PWM pulse pattern at an inverter output to drive an induction motor, proposes methods for improving distortion and torque ripples using a carrier frequency modulation (CFM), and describes a method for realizing the improvement through use of a single-chip microcomputer. The method defines evaluation parameters corresponding to the distortion and torque ripples, and optimizes the CFM depth to the parameters. The PWM pulse pattern has its voltage vector and time width so selected that the time integrated space vector of a three-phase voltage approaches a circular locus. Furthermore, the carrier frequency, that is the sampling frequency of the inverter, is also adjusted so that the above evaluation parameters are minimized. The addition of a new variable called the frequency modulation provides freedom in selecting an output characteristic as called for by the purpose. 12 refs., 18 figs.

Perceptual interaction between carrier periodicity and amplitude modulation in broadband stimuli: A comparison of the autocorrelation and modulation-filterbank model

DEFF Research Database (Denmark)

Stein, A.; Ewert, Stephan; Wiegrebe, L.

2005-01-01

, autocorrelation is applied. Considering the large overlap in pitch and modulation perception, this is not parsimonious. Two experiments are presented to investigate the interaction between carrier periodicity, which produces strong pitch sensations, and envelope periodicity using broadband stimuli. Results show......Recent temporal models of pitch and amplitude modulation perception converge on a relatively realistic implementation of cochlear processing followed by a temporal analysis of periodicity. However, for modulation perception, a modulation filterbank is applied whereas for pitch perception...

Design and analysis of ultrasonic monaural audio guiding device for the visually impaired.

Science.gov (United States)

Kim, Keonwook; Kim, Hyunjai; Yun, Gihun; Kim, Myungsoo

2009-01-01

The novel Audio Guiding Device (AGD) based on the ultrasonic, which is named as SonicID, has been developed in order to localize point of interest for the visually impaired. The SonicID requires the infrastructure of the transmitters for broadcasting the location information over the ultrasonic carrier. The user with ultrasonic headset receives the information with variable amplitude upon the location and direction of the user due to the ultrasonic characteristic and modulation method. This paper proposes the monaural headset form factor of the SonicID which improves the daily life of the beneficiary compare to the previous version which uses the both ears. Experimental results from SonicID, Bluetooth, and audible sound show that the SonicID demonstrates comparable localization performance to the audible sound with silence to others.

Wavelet-based audio embedding and audio/video compression

Science.gov (United States)

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

2001-12-01

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

Carrier Modulation Layer-Enhanced Organic Light-Emitting Diodes

Directory of Open Access Journals (Sweden)

Jwo-Huei Jou

2015-07-01

Full Text Available Organic light-emitting diode (OLED-based display products have already emerged in the market and their efficiencies and lifetimes are sound at the comparatively low required luminance. To realize OLED for lighting application sooner, higher light quality and better power efficiency at elevated luminance are still demanded. This review reveals the advantages of incorporating a nano-scale carrier modulation layer (CML, also known as a spacer, carrier-regulating layer, or interlayer, among other terms, to tune the chromaticity and color temperature as well as to markedly improve the device efficiency and color rendering index (CRI for numerous OLED devices. The functions of the CML can be enhanced as multiple layers and blend structures are employed. At proper thickness, the employment of CML enables the device to balance the distribution of carriers in the two emissive zones and achieve high device efficiencies and long operational lifetime while maintaining very high CRI. Moreover, we have also reviewed the effect of using CML on the most significant characteristics of OLEDs, namely: efficiency, luminance, life-time, CRI, SRI, chromaticity, and the color temperature, and see how the thickness tuning and selection of proper CML are crucial to effectively control the OLED device performance.

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

DEFF Research Database (Denmark)

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

2013-01-01

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

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

Science.gov (United States)

Hage, Steffen R; Nieder, Andreas

2015-05-06

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

Field Effect Optoelectronic Modulation of Quantum-Confined Carriers in Black Phosphorus.

Science.gov (United States)

Whitney, William S; Sherrott, Michelle C; Jariwala, Deep; Lin, Wei-Hsiang; Bechtel, Hans A; Rossman, George R; Atwater, Harry A

2017-01-11

We report measurements of the infrared optical response of thin black phosphorus under field-effect modulation. We interpret the observed spectral changes as a combination of an ambipolar Burstein-Moss (BM) shift of the absorption edge due to band-filling under gate control, and a quantum confined Franz-Keldysh (QCFK) effect, phenomena that have been proposed theoretically to occur for black phosphorus under an applied electric field. Distinct optical responses are observed depending on the flake thickness and starting carrier concentration. Transmission extinction modulation amplitudes of more than two percent are observed, suggesting the potential for use of black phosphorus as an active material in mid-infrared optoelectronic modulator applications.

Multi-Gigahertz radar range processing of baseband and RF carrier modulated signals in Tm:YAG

International Nuclear Information System (INIS)

Merkel, K.D.; Krishna Mohan, R.; Cole, Z.; Chang, T.; Olson, A.; Babbitt, W.R.

2004-01-01

An optical device is described and demonstrated that uses a spatial-spectral holographic material to perform coherent signal processing operations on analog, high-bandwidth optical signals with large time-bandwidth-products. Signal processing is performed as the material records the coherent spectral interference (or cross-power spectrum) of modulated optical signals as a spatial-spectral population grating between electronic transition states. Multiple exposures of processing pulse sequences are integrated with increasing grating strength. The device, coined as the Spatial-Spectral Coherent Holographic Integrating Processor (or S 2 -CHIP), is described as currently envisioned for a broadband, mid-to-high pulse repetition frequency range-Doppler radar signal processing system. Experiments were performed in Tm:YAG (0.1 at% at 5 K) to demonstrate time delay variation, integration dynamics, and effects of coding as applied to a radar range processor. These demonstrations used baseband modulation with a 1 gigabit per second (GPBS) bit rate and code length of 512 bits (512 ns), where delays up to 1.0 μs were resolved with greater than a 40 dB peak to RMS sidelobe ratio after 800 processing shots. Multi-GHz processing was demonstrated using a bit rate of 2.5 GBPS (baseband modulation) and code length of 2048 bits (819.2 ns). Processing of double-sideband modulated signals on a radio frequency (RF) carrier was demonstrated, where 512 bit, 1.0 GBPS codes were modulated on a 1.75 GHz carrier and then modulated on the optical carrier

Single-Carrier Modulation for Neutral-Point-Clamped Inverters in Three-Phase Transformerless Photovoltaic Systems

DEFF Research Database (Denmark)

Guo, Xiaoqiang; Cavalcanti, Marcelo C.; Farias, Alexandre M.

2013-01-01

Modulation strategy is one of the most important issues for three-level neutral-point-clamped inverters in three-phase transformerless photovoltaic systems. A challenge for modulation is how to keep the common-mode voltages constant to reduce the leakage currents. A single-carrier modulation...... strategy is proposed. It has a very simple structure, and the common-mode voltages can be kept constant with no need of complex space-vector modulation or multicarrier pulsewidth modulation. Experimental results verify the theoretical analysis and the effectiveness of the presented method....

Spectrotemporal modulation sensitivity for hearing-impaired listeners: dependence on carrier center frequency and the relationship to speech intelligibility.

Science.gov (United States)

Mehraei, Golbarg; Gallun, Frederick J; Leek, Marjorie R; Bernstein, Joshua G W

2014-07-01

Poor speech understanding in noise by hearing-impaired (HI) listeners is only partly explained by elevated audiometric thresholds. Suprathreshold-processing impairments such as reduced temporal or spectral resolution or temporal fine-structure (TFS) processing ability might also contribute. Although speech contains dynamic combinations of temporal and spectral modulation and TFS content, these capabilities are often treated separately. Modulation-depth detection thresholds for spectrotemporal modulation (STM) applied to octave-band noise were measured for normal-hearing and HI listeners as a function of temporal modulation rate (4-32 Hz), spectral ripple density [0.5-4 cycles/octave (c/o)] and carrier center frequency (500-4000 Hz). STM sensitivity was worse than normal for HI listeners only for a low-frequency carrier (1000 Hz) at low temporal modulation rates (4-12 Hz) and a spectral ripple density of 2 c/o, and for a high-frequency carrier (4000 Hz) at a high spectral ripple density (4 c/o). STM sensitivity for the 4-Hz, 4-c/o condition for a 4000-Hz carrier and for the 4-Hz, 2-c/o condition for a 1000-Hz carrier were correlated with speech-recognition performance in noise after partialling out the audiogram-based speech-intelligibility index. Poor speech-reception and STM-detection performance for HI listeners may be related to a combination of reduced frequency selectivity and a TFS-processing deficit limiting the ability to track spectral-peak movements.

Visualizing Carrier Transport in Metal Halide Perovskite Nanoplates via Electric Field Modulated Photoluminescence Imaging.

Science.gov (United States)

Hu, Xuelu; Wang, Xiao; Fan, Peng; Li, Yunyun; Zhang, Xuehong; Liu, Qingbo; Zheng, Weihao; Xu, Gengzhao; Wang, Xiaoxia; Zhu, Xiaoli; Pan, Anlian

2018-05-09

Metal halide perovskite nanostructures have recently been the focus of intense research due to their exceptional optoelectronic properties and potential applications in integrated photonics devices. Charge transport in perovskite nanostructure is a crucial process that defines efficiency of optoelectronic devices but still requires a deep understanding. Herein, we report the study of the charge transport, particularly the drift of minority carrier in both all-inorganic CsPbBr 3 and organic-inorganic hybrid CH 3 NH 3 PbBr 3 perovskite nanoplates by electric field modulated photoluminescence (PL) imaging. Bias voltage dependent elongated PL emission patterns were observed due to the carrier drift at external electric fields. By fitting the drift length as a function of electric field, we obtained the carrier mobility of about 28 cm 2 V -1 S -1 in the CsPbBr 3 perovskite nanoplate. The result is consistent with the spatially resolved PL dynamics measurement, confirming the feasibility of the method. Furthermore, the electric field modulated PL imaging is successfully applied to the study of temperature-dependent carrier mobility in CsPbBr 3 nanoplates. This work not only offers insights for the mobile carrier in metal halide perovskite nanostructures, which is essential for optimizing device design and performance prediction, but also provides a novel and simple method to investigate charge transport in many other optoelectronic materials.

Audio Papers

DEFF Research Database (Denmark)

Groth, Sanne Krogh; Samson, Kristine

2016-01-01

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

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

DEFF Research Database (Denmark)

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

2012-01-01

This paper deals with a design of a digital interpolation filter for a 3rd order multi-bit ΣΔ modulator with over-sampling ratio OSR = 64. The interpolation filter and the ΣΔ modulator are part of the back-end of an audio signal processing system in a hearing-aid application. The aim in this paper...... is to compare this design to designs presented in other state-of-the-art works ranging from hi-fi audio to hearing-aids. By performing comparison, trends and tradeoffs in interpolation filter design are indentified and hearing-aid specifications are derived. The possibilities for hardware reduction...... in the interpolation filter are investigated. Proposed design simplifications presented here result in the least hardware demanding combination of oversampling ratio, number of stages and number of filter taps among a number of filters reported for audio applications....

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

Directory of Open Access Journals (Sweden)

Jose Juan Garcia-Hernandez

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

Reconfigurable digital receiver for 8PSK subcarrier multiplexed and 16QAM single carrier phase‐modulated radio over fiber links

DEFF Research Database (Denmark)

Guerrero Gonzalez, Neil; Zibar, Darko; Yu, Xianbin

2011-01-01

A reconfigurable digital receiver based on the k‐means algorithm is proposed for phase‐modulated subcarrier multiplexed (SCM) and quadrature amplitude‐modulated single carrier, phase‐modulated radio‐over‐fiber links. We report successful demodulation after 40 km single mode fiber transmission wit...... with three 50 Mbaud 8PSK SCM signals and a 312.5 Mbaud 16QAM single carrier. © 2011 Wiley Periodicals, Inc. Microwave Opt Technol Lett 53:1015–1018, 2011; View this article online at wileyonlinelibrary.com. DOI 10.1002/mop.25905...

Generation of tunable, high repetition rate frequency combs with equalized spectra using carrier injection based silicon modulators

Science.gov (United States)

Nagarjun, K. P.; Selvaraja, Shankar Kumar; Supradeepa, V. R.

2016-03-01

High repetition-rate frequency combs with tunable repetition rate and carrier frequency are extensively used in areas like Optical communications, Microwave Photonics and Metrology. A common technique for their generation is strong phase modulation of a CW-laser. This is commonly implemented using Lithium-Niobate based modulators. With phase modulation alone, the combs have poor spectral flatness and significant number of missing lines. To overcome this, a complex cascade of multiple intensity and phase modulators are used. A comb generator on Silicon based on these principles is desirable to enable on-chip integration with other functionalities while reducing power consumption and footprint. In this work, we analyse frequency comb generation in carrier injection based Silicon modulators. We observe an interesting effect in these comb generators. Enhanced absorption accompanying carrier injection, an undesirable effect in data modulators, shapes the amplitude here to enable high quality combs from a single modulator. Thus, along with reduced power consumption to generate a specific number of lines, the complexity has also been significantly reduced. We use a drift-diffusion solver and mode solver (Silvaco TCAD) along with Soref-Bennett relations to calculate the variations in refractive indices and absorption of an optimized Silicon PIN - waveguide modulator driven by an unbiased high frequency (10 Ghz) voltage signal. Our simulations demonstrate that with a device length of 1 cm, a driving voltage of 2V and minor shaping with a passive ring-resonator filter, we obtain 37 lines with a flatness better than 5-dB across the band and power consumption an order of magnitude smaller than Lithium-Niobate modulators.

Intelligent audio analysis

CERN Document Server

Schuller, Björn W

2013-01-01

This book provides the reader with the knowledge necessary for comprehension of the field of Intelligent Audio Analysis. It firstly introduces standard methods and discusses the typical Intelligent Audio Analysis chain going from audio data to audio features to audio recognition.  Further, an introduction to audio source separation, and enhancement and robustness are given. After the introductory parts, the book shows several applications for the three types of audio: speech, music, and general sound. Each task is shortly introduced, followed by a description of the specific data and methods applied, experiments and results, and a conclusion for this specific task. The books provides benchmark results and standardized test-beds for a broader range of audio analysis tasks. The main focus thereby lies on the parallel advancement of realism in audio analysis, as too often today’s results are overly optimistic owing to idealized testing conditions, and it serves to stimulate synergies arising from transfer of ...

Low-complexity Joint Sub-carrier Phase Noise Compensation for Digital Multi-carrier Systems

DEFF Research Database (Denmark)

Yankov, Metodi Plamenov; Barletta, Luca; Zibar, Darko

2017-01-01

Joint sub-carrier phase noise processing is proposed which recovers the SNR penalty related to decreased sub-carrier baudrate w.r.t. single carrier systems. The method enables digital sub-banding to be safely employed for nonlinear mitigation for modulation formats of up to 256-QAM.......Joint sub-carrier phase noise processing is proposed which recovers the SNR penalty related to decreased sub-carrier baudrate w.r.t. single carrier systems. The method enables digital sub-banding to be safely employed for nonlinear mitigation for modulation formats of up to 256-QAM....

Modulated phase matching and high-order harmonic enhancement mediated by the carrier-envelope phase

International Nuclear Information System (INIS)

Faccio, Daniele; Serrat, Carles; Cela, Jose M.; Farres, Albert; Di Trapani, Paolo; Biegert, Jens

2010-01-01

The process of high-order harmonic generation in gases is numerically investigated in the presence of a few-cycle pulsed-Bessel-beam pump, featuring a periodic modulation in the peak intensity due to large carrier-envelope-phase mismatch. A two-decade enhancement in the conversion efficiency is observed and interpreted as the consequence of a mechanism known as a nonlinearly induced modulation in the phase mismatch.

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.

Gigascale Silicon Photonic Transmitters Integrating HBT-based Carrier-injection Electroabsorption Modulator Structures

Science.gov (United States)

Fu, Enjin

Demand for more bandwidth is rapidly increasing, which is driven by data intensive applications such as high-definition (HD) video streaming, cloud storage, and terascale computing applications. Next-generation high-performance computing systems require power efficient chip-to-chip and intra-chip interconnect yielding densities on the order of 1Tbps/cm2. The performance requirements of such system are the driving force behind the development of silicon integrated optical interconnect, providing a cost-effective solution for fully integrated optical interconnect systems on a single substrate. Compared to conventional electrical interconnect, optical interconnects have several advantages, including frequency independent insertion loss resulting in ultra wide bandwidth and link latency reduction. For high-speed optical transmitter modules, the optical modulator is a key component of the optical I/O channel. This thesis presents a silicon integrated optical transmitter module design based on a novel silicon HBT-based carrier injection electroabsorption modulator (EAM), which has the merits of wide optical bandwidth, high speed, low power, low drive voltage, small footprint, and high modulation efficiency. The structure, mechanism, and fabrication of the modulator structure will be discussed which is followed by the electrical modeling of the post-processed modulator device. The design and realization of a 10Gbps monolithic optical transmitter module integrating the driver circuit architecture and the HBT-based EAM device in a 130nm BiCMOS process is discussed. For high power efficiency, a 6Gbps ultra-low power driver IC implemented in a 130nm BiCMOS process is presented. The driver IC incorporates an integrated 27-1 pseudo-random bit sequence (PRBS) generator for reliable high-speed testing, and a driver circuit featuring digitally-tuned pre-emphasis signal strength. With outstanding drive capability, the driver module can be applied to a wide range of carrier

Surface/Interface Carrier-Transport Modulation for Constructing Photon-Alternative Ultraviolet Detectors Based on Self-Bending-Assembled ZnO Nanowires.

Science.gov (United States)

Guo, Zhen; Zhou, Lianqun; Tang, Yuguo; Li, Lin; Zhang, Zhiqi; Yang, Hongbo; Ma, Hanbin; Nathan, Arokia; Zhao, Dongxu

2017-09-13

Surface/interface charge-carrier generation, diffusion, and recombination/transport modulation are especially important in the construction of photodetectors with high efficiency in the field of nanoscience. In the paper, a kind of ultraviolet (UV) detector is designed based on ZnO nanostructures considering photon-trapping, surface plasmonic resonance (SPR), piezophototronic effects, interface carrier-trapping/transport control, and collection. Through carefully optimized surface/interface carrier-transport modulation, a designed device with detectivity as high as 1.69 × 10 16 /1.71 × 10 16 cm·Hz 1/2 /W irradiating with 380 nm photons under ultralow bias of 0.2 V is realized by alternating nanoparticle/nanowire active layers, respectively, and the designed UV photodetectors show fast and slow recovery processes of 0.27 and 4.52 ms, respectively, which well-satisfy practical needs. Further, it is observed that UV photodetection could be performed within an alternative response by varying correlated key parameters, through efficient surface/interface carrier-transport modulation, spectrally resolved photoresponse of the detector revealing controlled detection in the UV region based on the ZnO nanomaterial, photodetection allowed or limited by varying the active layers, irradiation distance from one of the electrodes, standing states, or electric field. The detailed carrier generation, diffusion, and recombination/transport processes are well illustrated to explain charge-carrier dynamics contributing to the photoresponse behavior.

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

Efficient and Robust Detection of GFSK Signals under Dispersive Channel, Modulation Index, and Carrier Frequency Offset Conditions

Directory of Open Access Journals (Sweden)

Stephan Weiss

2005-09-01

Full Text Available Gaussian frequency shift keying is the modulation scheme specified for Bluetooth. Signal adversities typical in Bluetooth networks include AWGN, multipath propagation, carrier frequency, and modulation index offsets. In our effort to realise a robust but efficient Bluetooth receiver, we adopt a high-performance matched-filter-based detector, which is near optimal in AWGN, but requires a prohibitively costly filter bank for processing of K bits worth of the received signal. However, through filtering over a single bit period and performing phase propagation of intermediate results over successive single-bit stages, we eliminate redundancy involved in providing the matched filter outputs and reduce its complexity by up to 90% (for K=9. The constant modulus signal characteristic and the potential for carrier frequency offsets make the constant modulus algorithm (CMA suitable for channel equalisation, and we demonstrate its effectiveness in this paper. We also introduce a stochastic gradient-based algorithm for carrier frequency offset correction, and show that the relative rotation between successive intermediate filter outputs enables us to detect and correct offsets in modulation index.

A novel amplitude modulated triangular carrier gain linearization technique for SPWM inverter

Directory of Open Access Journals (Sweden)

Ramkumar Subburam

2009-01-01

Full Text Available This paper presents a new method to extend the linearity of the sinusoidal pulse width modulation (SPWM to full range of the pulse dropping region. The proposed amplitude modulated triangular carrier PWM method (AMTCPWM increases the dynamic range of the SPWM control and eliminates the need of nonlinear modulation in the pulse dropping region to reach the square wave boundary. The novel method combines the spectral quality of SPWM with the efficient single-mode linear control. A simple analytical characterization of the exact method is presented and its effectiveness is demonstrated using simulation for the basic single-phase H-bridge inverter circuit. The hardware results of the designed prototype inverter are presented to validate the betterment of the novel scheme. .

Study on a low complexity adaptive modulation algorithm in OFDM-ROF system with sub-carrier grouping technology

Science.gov (United States)

Liu, Chong-xin; Liu, Bo; Zhang, Li-jia; Xin, Xiang-jun; Tian, Qing-hua; Tian, Feng; Wang, Yong-jun; Rao, Lan; Mao, Yaya; Li, Deng-ao

2018-01-01

During the last decade, the orthogonal frequency division multiplexing radio-over-fiber (OFDM-ROF) system with adaptive modulation technology is of great interest due to its capability of raising the spectral efficiency dramatically, reducing the effects of fiber link or wireless channel, and improving the communication quality. In this study, according to theoretical analysis of nonlinear distortion and frequency selective fading on the transmitted signal, a low-complexity adaptive modulation algorithm is proposed in combination with sub-carrier grouping technology. This algorithm achieves the optimal performance of the system by calculating the average combined signal-to-noise ratio of each group and dynamically adjusting the origination modulation format according to the preset threshold and user's requirements. At the same time, this algorithm takes the sub-carrier group as the smallest unit in the initial bit allocation and the subsequent bit adjustment. So, the algorithm complexity is only 1 /M (M is the number of sub-carriers in each group) of Fischer algorithm, which is much smaller than many classic adaptive modulation algorithms, such as Hughes-Hartogs algorithm, Chow algorithm, and is in line with the development direction of green and high speed communication. Simulation results show that the performance of OFDM-ROF system with the improved algorithm is much better than those without adaptive modulation, and the BER of the former achieves 10e1 to 10e2 times lower than the latter when SNR values gets larger. We can obtain that this low complexity adaptive modulation algorithm is extremely useful for the OFDM-ROF system.

Probing exotic phases of interacting two-dimensional carriers using one-dimensional density modulation

Science.gov (United States)

Mueed, M. A.

In this Thesis, we present low-temperature magnetotransport studies of two-dimensional (2D) electron and hole systems confined to GaAs quantum wells and subjected to a one-dimensional, periodic density modulation. The modulation is achieved through the piezo-electric effect in GaAs as we fabricate a periodic, strain-inducing superlattice on the sample surface. Under varying perpendicular magnetic field, whenever the carriers' cyclotron orbit becomes commensurate with the modulation period, the magnetoresistance exhibits a minimum value. The resulting oscillations, known as the commensurability oscillations, directly measure the carriers' Fermi wave vector. Imposing a density modulation thus allows us to study the Fermi contour properties of 2D electrons and holes near zero field, and composite fermions (CFs) near the half filling of the lowest Landau level, i.e., filling factor nu=1/2. The application of a parallel magnetic field (B||) also features extensively in the Thesis. First, we use commensurability oscillations to capture the B||-induced deformation and the eventual splitting of the Fermi contour of 2D electrons. We also deduce the scattering time anisotropy of hole-flux CFs whose Fermi contour is rendered anisotropic by B||. Moreover, we study the anisotropic (warped) Fermi contour of 2D holes and hole-flux CFs in wide quantum well samples at B||=0. The results provide evidence that CFs inherit Fermi contour properties from their zero-field counterparts. We further investigate the fate of CFs near the bilayer quantum Hall states at nu=1 and 1/2 induced by a large B||. We observe that the commensurability features of CFs near nu=1 are consistent with half the total carrier density, implying that CFs prefer to stay in separate layers and show a two-component behavior. In contrast, close to nu=1/2, CFs appear single-layer-like (single-component) as their commensurability features correspond to the total density. This finding sheds light on the different

A Novel Robust Audio Watermarking Algorithm by Modifying the Average Amplitude in Transform Domain

Directory of Open Access Journals (Sweden)

Qiuling Wu

2018-05-01

Full Text Available In order to improve the robustness and imperceptibility in practical application, a novel audio watermarking algorithm with strong robustness is proposed by exploring the multi-resolution characteristic of discrete wavelet transform (DWT and the energy compaction capability of discrete cosine transform (DCT. The human auditory system is insensitive to the minor changes in the frequency components of the audio signal, so the watermarks can be embedded by slightly modifying the frequency components of the audio signal. The audio fragments segmented from the cover audio signal are decomposed by DWT to obtain several groups of wavelet coefficients with different frequency bands, and then the fourth level detail coefficient is selected to be divided into the former packet and the latter packet, which are executed for DCT to get two sets of transform domain coefficients (TDC respectively. Finally, the average amplitudes of the two sets of TDC are modified to embed the binary image watermark according to the special embedding rule. The watermark extraction is blind without the carrier audio signal. Experimental results confirm that the proposed algorithm has good imperceptibility, large payload capacity and strong robustness when resisting against various attacks such as MP3 compression, low-pass filtering, re-sampling, re-quantization, amplitude scaling, echo addition and noise corruption.

Analysis of carrier transport and carrier trapping in organic diodes with polyimide-6,13-Bis(triisopropylsilylethynyl)pentacene double-layer by charge modulation spectroscopy and optical second harmonic generation measurement

Energy Technology Data Exchange (ETDEWEB)

Lim, Eunju, E-mail: elim@dankook.ac.kr, E-mail: taguchi.d.aa@m.titech.ac.jp, E-mail: iwamoto@pe.titech.ac.jp [Department of Applied Physics, Institute of Nanosensor and Biotechnology, Dankook University, Jukjeon-dong, Gyeonggi-do 448-701 (Korea, Republic of); Taguchi, Dai, E-mail: elim@dankook.ac.kr, E-mail: taguchi.d.aa@m.titech.ac.jp, E-mail: iwamoto@pe.titech.ac.jp; Iwamoto, Mitsumasa, E-mail: elim@dankook.ac.kr, E-mail: taguchi.d.aa@m.titech.ac.jp, E-mail: iwamoto@pe.titech.ac.jp [Department of Physical Electronics, Tokyo Institute of Technology 2-12-1, O-okayama, Meguro-ku, Tokyo 152-8552 (Japan)

2014-08-18

We studied the carrier transport and carrier trapping in indium tin oxide/polyimide (PI)/6,13-Bis(triisopropylsilylethynyl)pentacene (TIPS-pentacene)/Au diodes by using charge modulation spectroscopy (CMS) and time-resolved electric field induced optical second harmonic generation (TR-EFISHG) measurements. TR-EFISHG directly probes the spatial carrier behaviors in the diodes, and CMS is useful in explaining the carrier motion with respect to energy. The results clearly indicate that the injected carriers move across TIPS-pentacene thorough the molecular energy states of TIPS-pentacene and accumulate at the PI/TIPS-pentacene interface. However, some carriers are trapped in the PI layers. These findings take into account the capacitance-voltage and current-voltage characteristics of the diodes.

Intensity position modulation for free-space laser communication system

Science.gov (United States)

Jangjoo, Alireza; Faghihi, F.

2004-12-01

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

Effective information rates of single-carrier and multi-carrier modulation schemes for bandwidth constrained IM/DD systems

KAUST Repository

Mazahir, Sana

2017-07-31

Information-theoretic and signal processing aspects of some modulation schemes designed for intensity modulation/direct detection (IM/DD) optical wireless communication (OWC) systems are studied. Due to the constraints of IM/DD signals (non-negative real and baseband signals), the construction of these signals along with their time and frequency characteristics differ from their RF counterparts. This necessitates a careful study of such schemes under practical constraints. Three schemes are studied in this paper, namely, single carrier pulse amplitude modulation (SC-PAM), asymmetrically clipped optical OFDM (ACO-OFDM), and DC biased optical OFDM (DCO-OFDM). Our aim is to carry out a comparative study of these schemes in the presence of identical constraints on bandwidth and average optical power. The study reveals that the clipping operation required in ACO-OFDM significantly reduces its information rate, and as a result, it is outperformed by SC-PAM. Such a limitation does not apply to DCO-OFDM which has a higher information rate, even though part of the available optical power is expended in the non-information-bearing DC bias.

Audio Conferencing Enhancements

OpenAIRE

VESTERINEN, LEENA

2006-01-01

Audio conferencing allows multiple people in distant locations to interact in a single voice call. Whilst it can be very useful service it also has several key disadvantages. This thesis study investigated the options for improving the user experience of the mobile teleconferencing applications. In particular, the use of 3D, spatial audio and visualinteractive functionality was investigated as the means of improving the intelligibility and audio perception during the audio...

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

Directory of Open Access Journals (Sweden)

Jussi Kuutti

2014-02-01

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

A physico-genetic module for the polarisation of auxin efflux carriers PIN-FORMED (PIN)

Science.gov (United States)

Hernández-Hernández, Valeria; Barrio, Rafael A.; Benítez, Mariana; Nakayama, Naomi; Romero-Arias, José Roberto; Villarreal, Carlos

2018-05-01

Intracellular polarisation of auxin efflux carriers is crucial for understanding how auxin gradients form in plants. The polarisation dynamics of auxin efflux carriers PIN-FORMED (PIN) depends on both biomechanical forces as well as chemical, molecular and genetic factors. Biomechanical forces have shown to affect the localisation of PIN transporters to the plasma membrane. We propose a physico-genetic module of PIN polarisation that integrates biomechanical, molecular, and cellular processes as well as their non-linear interactions. The module was implemented as a discrete Boolean model and then approximated to a continuous dynamic system, in order to explore the relative contribution of the factors mediating PIN polarisation at the scale of single cell. Our models recovered qualitative behaviours that have been experimentally observed and enable us to predict that, in the context of PIN polarisation, the effects of the mechanical forces can predominate over the activity of molecular factors such as the GTPase ROP6 and the ROP-INTERACTIVE CRIB MOTIF-CONTAINING PROTEIN RIC1.

A 115dB-DR Audio DAC with –61dBFS out-of-band noise

NARCIS (Netherlands)

Westerveld, H.J.; Schinkel, Daniel; van Tuijl, Adrianus Johannes Maria

2015-01-01

Out-of-band noise (OBN) is troublesome in analog circuits that process the output of a noise-shaping audio DAC. It causes slewing in amplifiers and aliasing in sampling circuits like ADCs and class-D amplifiers. Nonlinearity in these circuits also causes cross-modulation of the OBN into the audio

An Introduction to Boiler Water Chemistry for the Marine Engineer: A Text of Audio-Tutorial Instruction.

Science.gov (United States)

Schlenker, Richard M.; And Others

Presented is a manuscript for an introductory boiler water chemistry course for marine engineer education. The course is modular, self-paced, audio-tutorial, contract graded and combined lecture-laboratory instructed. Lectures are presented to students individually via audio-tapes and 35 mm slides. The course consists of a total of 17 modules -…

Simulation Performance of Multiple-Input Multiple-Output Systems Employing Single-Carrier Modulation and Orthogonal Frequency Division Multiplexing

National Research Council Canada - National Science Library

Saglam, Halil D

2004-01-01

...) systems utilizing Alamouti-based space-time block coding (STBC) technique. The MIMO communication systems using STBC technique employing both single-carrier modulation and orthogonal frequency division multiplexing (OFDM...

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

Science.gov (United States)

2010-10-01

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

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

Science.gov (United States)

2010-10-01

... 47 Telecommunication 2 2010-10-01 2010-10-01 false Licensing provisions for the 2.3 GHz satellite digital audio radio service. 25.144 Section 25.144 Telecommunication FEDERAL COMMUNICATIONS COMMISSION (CONTINUED) COMMON CARRIER SERVICES SATELLITE COMMUNICATIONS Applications and Licenses Space Stations § 25...

Semantics and the multisensory brain: how meaning modulates processes of audio-visual integration.

Science.gov (United States)

Doehrmann, Oliver; Naumer, Marcus J

2008-11-25

By using meaningful stimuli, multisensory research has recently started to investigate the impact of stimulus content on crossmodal integration. Variations in this respect have often been termed as "semantic". In this paper we will review work related to the question for which tasks the influence of semantic factors has been found and which cortical networks are most likely to mediate these effects. More specifically, the focus of this paper will be on processing of object stimuli presented in the auditory and visual sensory modalities. Furthermore, we will investigate which cortical regions are particularly responsive to experimental variations of content by comparing semantically matching ("congruent") and mismatching ("incongruent") experimental conditions. In this context, recent neuroimaging studies point toward a possible functional differentiation of temporal and frontal cortical regions, with the former being more responsive to semantically congruent and the latter to semantically incongruent audio-visual (AV) stimulation. To account for these differential effects, we will suggest in the final section of this paper a possible synthesis of these data on semantic modulation of AV integration with findings from neuroimaging studies and theoretical accounts of semantic memory.

Audio Twister

DEFF Research Database (Denmark)

Cermak, Daniel; Moreno Garcia, Rodrigo; Monastiridis, Stefanos

2015-01-01

Daniel Cermak-Sassenrath, Rodrigo Moreno Garcia, Stefanos Monastiridis. Audio Twister. Installation. P-Hack Copenhagen 2015, Copenhagen, DK, Apr 24, 2015.......Daniel Cermak-Sassenrath, Rodrigo Moreno Garcia, Stefanos Monastiridis. Audio Twister. Installation. P-Hack Copenhagen 2015, Copenhagen, DK, Apr 24, 2015....

Back to basics audio

CERN Document Server

Nathan, Julian

1998-01-01

Back to Basics Audio is a thorough, yet approachable handbook on audio electronics theory and equipment. The first part of the book discusses electrical and audio principles. Those principles form a basis for understanding the operation of equipment and systems, covered in the second section. Finally, the author addresses planning and installation of a home audio system.Julian Nathan joined the audio service and manufacturing industry in 1954 and moved into motion picture engineering and production in 1960. He installed and operated recording theaters in Sydney, Austra

Capacity upgrade in short-reach optical fibre networks: simultaneous 4-PAM 20 Gbps data and polarization-modulated PPS clock signal using a single VCSEL carrier

Science.gov (United States)

Isoe, G. M.; Wassin, S.; Gamatham, R. R. G.; Leitch, A. W. R.; Gibbon, T. B.

2017-11-01

In this work, a four-level pulse amplitude modulation (4-PAM) format with a polarization-modulated pulse per second (PPS) clock signal using a single vertical cavity surface emitting laser (VCSEL) carrier is for the first time experimentally demonstrated. We propose uncomplex alternative technique for increasing capacity and flexibility in short-reach optical communication links through multi-signal modulation onto a single VCSEL carrier. A 20 Gbps 4-PAM data signal is directly modulated onto a single mode 10 GHz bandwidth VCSEL carrier at 1310 nm, therefore, doubling the network bit rate. Carrier spectral efficiency is further maximized by exploiting the inherent orthogonal polarization switching of the VCSEL carrier with changing bias in transmission of a PPS clock signal. We, therefore, simultaneously transmit a 20 Gbps 4-PAM data signal and a polarization-based PPS clock signal using a single VCSEL carrier. It is the first time a signal VCSEL carrier is reported to simultaneously transmit a directly modulated 20 Gbps 4-PAM data signal and a polarization-based PPS clock signal. We further demonstrate on the design of a software-defined digital signal processing (DSP)-assisted receiver as an alternative to costly receiver hardware. Experimental results show that a 3.21 km fibre transmission with simultaneous 20 Gbps 4-PAM data signal and polarization-based PPS clock signal introduced a penalty of 3.76 dB. The contribution of polarization-based PPS clock signal to this penalty was found out to be 0.41 dB. Simultaneous distribution of data and timing clock signals over shared network infrastructure significantly increases the aggregated data rate at different optical network units (ONUs), without costly investment.

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

International Nuclear Information System (INIS)

Luo Shumeng; Li Dongmei

2013-01-01

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

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

International Nuclear Information System (INIS)

George, Rohini; Chung, Theodore D.; Vedam, Sastry S.; Ramakrishnan, Viswanathan; Mohan, Radhe; Weiss, Elisabeth; Keall, Paul J.

2006-01-01

Purpose: Respiratory gating is a commercially available technology for reducing the deleterious effects of motion during imaging and treatment. The efficacy of gating is dependent on the reproducibility within and between respiratory cycles during imaging and treatment. The aim of this study was to determine whether audio-visual biofeedback can improve respiratory reproducibility by decreasing residual motion and therefore increasing the accuracy of gated radiotherapy. Methods and Materials: A total of 331 respiratory traces were collected from 24 lung cancer patients. The protocol consisted of five breathing training sessions spaced about a week apart. Within each session the patients initially breathed without any instruction (free breathing), with audio instructions and with audio-visual biofeedback. Residual motion was quantified by the standard deviation of the respiratory signal within the gating window. Results: Audio-visual biofeedback significantly reduced residual motion compared with free breathing and audio instruction. Displacement-based gating has lower residual motion than phase-based gating. Little reduction in residual motion was found for duty cycles less than 30%; for duty cycles above 50% there was a sharp increase in residual motion. Conclusions: The efficiency and reproducibility of gating can be improved by: incorporating audio-visual biofeedback, using a 30-50% duty cycle, gating during exhalation, and using displacement-based gating

Highly efficient low color temperature organic LED using blend carrier modulation layer

Science.gov (United States)

Hsieh, Yao-Ching; Chen, Szu-Hao; Shen, Shih-Ming; Wang, Ching-Chiun; Chen, Chien-Chih; Jou, Jwo-Huei

2012-10-01

Color temperature (CT) of light has great effect on human physiology and psychology, and low CT light, minimizing melatonin suppression and decreasing the risk of breast, colorectal, and prostate cancer. We demonstrates the incorporation of a blend carrier modulation interlayer (CML) between emissive layers to improve the device performance of low CT organic light emitting diodes, which exhibits an external quantum efficiency of 22.7% and 36 lm W-1 (54 cd A-1) with 1880 K at 100 cd m-2, or 20.8% and 29 lm W-1 (50 cd A-1) with 1940 K at 1000 cd m-2. The result shows a CT much lower than that of incandescent bulbs, which is 2500 K with 15 lmW-1 efficiency, and even as low as that of candles, which is 2000 K with 0.1 lmW-1. The high efficiency of the proposed device may be attributed to its CML, which helps effectively distribute the entering carriers into the available recombination zones.

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

Energy Technology Data Exchange (ETDEWEB)

NONE

2000-03-01

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

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

Directory of Open Access Journals (Sweden)

O. Schnick

2007-06-01

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

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

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

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

Directory of Open Access Journals (Sweden)

John Mourjopoulos

2007-01-01

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

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

Directory of Open Access Journals (Sweden)

Mourjopoulos John

2007-01-01

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

Integrated unaligned resonant modulator tuning

Energy Technology Data Exchange (ETDEWEB)

Zortman, William A.; Lentine, Anthony L.

2017-10-03

Methods and systems for tuning a resonant modulator are disclosed. One method includes receiving a carrier signal modulated by the resonant modulator with a stream of data having an approximately equal number of high and low bits, determining an average power of the modulated carrier signal, comparing the average power to a predetermined threshold, and operating a tuning device coupled to the resonant modulator based on the comparison of the average power and the predetermined threshold. One system includes an input structure, a plurality of processing elements, and a digital control element. The input structure is configured to receive, from the resonant modulator, a modulated carrier signal. The plurality of processing elements are configured to determine an average power of the modulated carrier signal. The digital control element is configured to operate a tuning device coupled to the resonant modulator based on the average power of the modulated carrier signal.

Categorizing Video Game Audio

DEFF Research Database (Denmark)

Westerberg, Andreas Rytter; Schoenau-Fog, Henrik

2015-01-01

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

Wiener's Loop Filter for PLL-Based Carrier Recovery of OQPSK and MSK-Type Modulations

Directory of Open Access Journals (Sweden)

Arnaldo Spalvieri

2008-01-01

Full Text Available This letter considers carrier recovery for offset quadrature phase shift keying (OQPSK and minimum shift keying-type (MSK-type modulations based on phase-lock loop (PLL. The concern of the letter is the optimization of the loop filter of the PLL. The optimization is worked out in the light of Wiener's theory taking into account the phase noise affecting the incoming carrier, the additive white Gaussian noise that is present on the channel, and the self-noise produced by the phase detector. Delay in the loop, which may affect the numerical implementation of the PLL, is also considered. Closed-form expressions for the loop filter and for the mean-square error are given for the case where the phase noise is characterized as a first-order process.

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.

Amplitude modulation detection with concurrent frequency modulation.

Science.gov (United States)

Nagaraj, Naveen K

2016-09-01

Human speech consists of concomitant temporal modulations in amplitude and frequency that are crucial for speech perception. In this study, amplitude modulation (AM) detection thresholds were measured for 550 and 5000 Hz carriers with and without concurrent frequency modulation (FM), at AM rates crucial for speech perception. Results indicate that adding 40 Hz FM interferes with AM detection, more so for 5000 Hz carrier and for frequency deviations exceeding the critical bandwidth of the carrier frequency. These findings suggest that future cochlear implant processors, encoding speech fine-structures may consider limiting the FM to narrow bandwidth and to low frequencies.

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....

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,

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

Advances in audio source seperation and multisource audio content retrieval

Science.gov (United States)

Vincent, Emmanuel

2012-06-01

Audio source separation aims to extract the signals of individual sound sources from a given recording. In this paper, we review three recent advances which improve the robustness of source separation in real-world challenging scenarios and enable its use for multisource content retrieval tasks, such as automatic speech recognition (ASR) or acoustic event detection (AED) in noisy environments. We present a Flexible Audio Source Separation Toolkit (FASST) and discuss its advantages compared to earlier approaches such as independent component analysis (ICA) and sparse component analysis (SCA). We explain how cues as diverse as harmonicity, spectral envelope, temporal fine structure or spatial location can be jointly exploited by this toolkit. We subsequently present the uncertainty decoding (UD) framework for the integration of audio source separation and audio content retrieval. We show how the uncertainty about the separated source signals can be accurately estimated and propagated to the features. Finally, we explain how this uncertainty can be efficiently exploited by a classifier, both at the training and the decoding stage. We illustrate the resulting performance improvements in terms of speech separation quality and speaker recognition accuracy.

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.

Automatic Modulation Classification Based on Deep Learning for Unmanned Aerial Vehicles.

Science.gov (United States)

Zhang, Duona; Ding, Wenrui; Zhang, Baochang; Xie, Chunyu; Li, Hongguang; Liu, Chunhui; Han, Jungong

2018-03-20

Deep learning has recently attracted much attention due to its excellent performance in processing audio, image, and video data. However, few studies are devoted to the field of automatic modulation classification (AMC). It is one of the most well-known research topics in communication signal recognition and remains challenging for traditional methods due to complex disturbance from other sources. This paper proposes a heterogeneous deep model fusion (HDMF) method to solve the problem in a unified framework. The contributions include the following: (1) a convolutional neural network (CNN) and long short-term memory (LSTM) are combined by two different ways without prior knowledge involved; (2) a large database, including eleven types of single-carrier modulation signals with various noises as well as a fading channel, is collected with various signal-to-noise ratios (SNRs) based on a real geographical environment; and (3) experimental results demonstrate that HDMF is very capable of coping with the AMC problem, and achieves much better performance when compared with the independent network.

Audio collection in the SASA Institute of Musicology

Directory of Open Access Journals (Sweden)

Lajić-Mihajlović Danka

2010-01-01

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

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

Science.gov (United States)

Romero-Fresco, Pablo; Fryer, Louise

2013-01-01

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

A Method to Detect AAC Audio Forgery

Directory of Open Access Journals (Sweden)

Qingzhong Liu

2015-08-01

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

Hiding secret data into a carrier image

Directory of Open Access Journals (Sweden)

Ovidiu COSMA

2012-06-01

Full Text Available The object of steganography is embedding hidden information in an appropriate multimedia carrier, e.g., image, audio, or video. There are several known methods of solving this problem, which operate either in the space domain or in the frequency domain, and are distinguished by the following characteristics: payload, robustness and strength. The payload is the amount of secret data that can be embedded in the carrier without inducing suspicious artefacts, robustness indicates the degree in which the secret data is affected by the normal processing of the carrier e.g., compression, and the strength indicate how easy the presence of hidden data can be detected by steganalysis techniques. This paper presents a new method of hiding secret data into a digital image compressed by a technique based on the Discrete Wavelet Transform (DWT [2] and the Set Partitioning In Hierarchical Trees (SPIHT subband coding algorithm [6]. The proposed method admits huge payloads and has considerable strength.

Location audio simplified capturing your audio and your audience

CERN Document Server

Miles, Dean

2014-01-01

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

Smartphone audio port data collection cookbook

Directory of Open Access Journals (Sweden)

Kyle Forinash

2018-06-01

Full Text Available The audio port of a smartphone is designed to send and receive audio but can be harnessed for portable, economical, and accurate data collection from a variety of sources. While smartphones have internal sensors to measure a number of physical phenomena such as acceleration, magnetism and illumination levels, measurement of other phenomena such as voltage, external temperature, or accurate timing of moving objects are excluded. The audio port cannot be only employed to sense external phenomena. It has the additional advantage of timing precision; because audio is recorded or played at a controlled rate separated from other smartphone activities, timings based on audio can be highly accurate. The following outlines unpublished details of the audio port technical elements for data collection, a general data collection recipe and an example timing application for Android devices.

Functional Imaging of Audio-Visual Selective Attention in Monkeys and Humans: How do Lapses in Monkey Performance Affect Cross-Species Correspondences?

Science.gov (United States)

Rinne, Teemu; Muers, Ross S; Salo, Emma; Slater, Heather; Petkov, Christopher I

2017-06-01

The cross-species correspondences and differences in how attention modulates brain responses in humans and animal models are poorly understood. We trained 2 monkeys to perform an audio-visual selective attention task during functional magnetic resonance imaging (fMRI), rewarding them to attend to stimuli in one modality while ignoring those in the other. Monkey fMRI identified regions strongly modulated by auditory or visual attention. Surprisingly, auditory attention-related modulations were much more restricted in monkeys than humans performing the same tasks during fMRI. Further analyses ruled out trivial explanations, suggesting that labile selective-attention performance was associated with inhomogeneous modulations in wide cortical regions in the monkeys. The findings provide initial insights into how audio-visual selective attention modulates the primate brain, identify sources for "lost" attention effects in monkeys, and carry implications for modeling the neurobiology of human cognition with nonhuman animals. © The Author 2017. Published by Oxford University Press.

Structure Learning in Audio

DEFF Research Database (Denmark)

Nielsen, Andreas Brinch

By having information about the setting a user is in, a computer is able to make decisions proactively to facilitate tasks for the user. Two approaches are taken in this thesis to achieve more information about an audio environment. One approach is that of classifying audio, and a new approach...... investigated. A fast and computationally simple approach that compares recordings and classifies if they are from the same audio environment have been developed, and shows very high accuracy and the ability to synchronize recordings in the case of recording devices which are not connected. A more general model...

A centralized audio presentation manager

Energy Technology Data Exchange (ETDEWEB)

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

1994-05-16

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

Bit rates in audio source coding

NARCIS (Netherlands)

Veldhuis, Raymond N.J.

1992-01-01

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

Implementing Audio-CASI on Windows’ Platforms

Science.gov (United States)

Cooley, Philip C.; Turner, Charles F.

2011-01-01

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

New Modeling and Simulation Platform for Communications Systems:(I Double Sideband Suppressed Carrier AM Modulator DSB-SC

Directory of Open Access Journals (Sweden)

H. A. Ahmed

2012-08-01

Full Text Available The main goal of this paper is to introduce a new platform for the implementation and simulation of communication systems. SCILAB/SCICOS is an open source software for conducting communication system related experiments, aiming to provide an experimentation platform for research on communication theories. Double Sideband Suppressed Carrier (DSB-SC Modulator is modeled and simulated using this platform.

A photovoltaic module

DEFF Research Database (Denmark)

2013-01-01

The present invention relates to a photovoltaic module comprising a carrier substrate, said carrier substrate carrying a purely printed structure comprising printed positive and negative module terminals, a plurality of printed photovoltaic cell units each comprising one or more printed...... photovoltaic cells, wherein the plurality of printed photovoltaic cell units are electrically connected in series between the positive and the negative module terminals such that any two neighbouring photovoltaic cell units are electrically connected by a printed interconnecting electrical conductor....... The carrier substrate comprises a foil and the total thickness of the photovoltaic module is below 500 [mu]m. Moreover, the nominal voltage level between the positive and the negative terminals is at least 5 kV DC....

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.

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.

Comparative evaluation of audio and audio - tactile methods to improve oral hygiene status of visually impaired school children

OpenAIRE

R Krishnakumar; Swarna Swathi Silla; Sugumaran K Durai; Mohan Govindarajan; Syed Shaheed Ahamed; Logeshwari Mathivanan

2016-01-01

Background: Visually impaired children are unable to maintain good oral hygiene, as their tactile abilities are often underdeveloped owing to their visual disturbances. Conventional brushing techniques are often poorly comprehended by these children and hence, it was decided to evaluate the effectiveness of audio and audio-tactile methods in improving the oral hygiene of these children. Objective: To evaluate and compare the effectiveness of audio and audio-tactile methods in improving oral h...

WLAN Technologies for Audio Delivery

Directory of Open Access Journals (Sweden)

Nicolas-Alexander Tatlas

2007-01-01

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

Substitute Energy Carriers from Refinement of Coal using HTR-Module

International Nuclear Information System (INIS)

Barnert, Heiko; Kugeler, Kurt; Will, Michael

2014-01-01

There is a revival of coal refinement in the world: a recent press article in Germany titled “The Renaissance of Coal Refinement”. It reports about a large number of conventional plans and plants for coal refinement in many countries in the world, and in particular in China. Nuclear energy can be of assistance, in particular the High Temperature Reactor-Module, because it offers all needed process energies. The status of the research, development, and demonstration, RDD, of technology is summarized, in particular of the former programs in Germany: The primary energy carriers were hard coal and lignite. The envisaged products were: Substitute Natural Gas, SNG, for the gas market, Hydrogen, H2, for a future H2-market, e.g. for airplane traffic, Liquid Fuels, as Substitute Gasoline, or as Energy Alcohol, e.g. Methanol CH3OH, in mixture with higher alcohols, for the car traffic and for home heating. (author)

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…

[Intermodal timing cues for audio-visual speech recognition].

Science.gov (United States)

Hashimoto, Masahiro; Kumashiro, Masaharu

2004-06-01

The purpose of this study was to investigate the limitations of lip-reading advantages for Japanese young adults by desynchronizing visual and auditory information in speech. In the experiment, audio-visual speech stimuli were presented under the six test conditions: audio-alone, and audio-visually with either 0, 60, 120, 240 or 480 ms of audio delay. The stimuli were the video recordings of a face of a female Japanese speaking long and short Japanese sentences. The intelligibility of the audio-visual stimuli was measured as a function of audio delays in sixteen untrained young subjects. Speech intelligibility under the audio-delay condition of less than 120 ms was significantly better than that under the audio-alone condition. On the other hand, the delay of 120 ms corresponded to the mean mora duration measured for the audio stimuli. The results implied that audio delays of up to 120 ms would not disrupt lip-reading advantage, because visual and auditory information in speech seemed to be integrated on a syllabic time scale. Potential applications of this research include noisy workplace in which a worker must extract relevant speech from all the other competing noises.

Presence and the utility of audio spatialization

DEFF Research Database (Denmark)

Bormann, Karsten

2005-01-01

The primary concern of this paper is whether the utility of audio spatialization, as opposed to the fidelity of audio spatialization, impacts presence. An experiment is reported that investigates the presence-performance relationship by decoupling spatial audio fidelity (realism) from task...... performance by varying the spatial fidelity of the audio independently of its relevance to performance on the search task that subjects were to perform. This was achieved by having conditions in which subjects searched for a music-playing radio (an active sound source) and having conditions in which...... supplied only nonattenuated audio was detrimental to performance. Even so, this group of subjects consistently had the largest increase in presence scores over the baseline experiment. Further, the Witmer and Singer (1998) presence questionnaire was more sensitive to whether the audio source was active...

Modified BTC Algorithm for Audio Signal Coding

Directory of Open Access Journals (Sweden)

TOMIC, S.

2016-11-01

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

Modulating the Surface State of SiC to Control Carrier Transport in Graphene/SiC.

Science.gov (United States)

Jia, Yuping; Sun, Xiaojuan; Shi, Zhiming; Jiang, Ke; Liu, Henan; Ben, Jianwei; Li, Dabing

2018-05-28

Silicon carbide (SiC) with epitaxial graphene (EG/SiC) shows a great potential in the applications of electronic and photoelectric devices. The performance of devices is primarily dependent on the interfacial heterojunction between graphene and SiC. Here, the band structure of the EG/SiC heterojunction is experimentally investigated by Kelvin probe force microscopy. The dependence of the barrier height at the EG/SiC heterojunction to the initial surface state of SiC is revealed. Both the barrier height and band bending tendency of the heterojunction can be modulated by controlling the surface state of SiC, leading to the tuned carrier transport behavior at the EG/SiC interface. The barrier height at the EG/SiC(000-1) interface is almost ten times that of the EG/SiC(0001) interface. As a result, the amount of carrier transport at the EG/SiC(000-1) interface is about ten times that of the EG/SiC(0001) interface. These results offer insights into the carrier transport behavior at the EG/SiC heterojunction by controlling the initial surface state of SiC, and this strategy can be extended in all devices with graphene as the top layer. © 2018 WILEY-VCH Verlag GmbH & Co. KGaA, Weinheim.

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

Science.gov (United States)

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

2014-02-01

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

Audio scene segmentation for video with generic content

Science.gov (United States)

Niu, Feng; Goela, Naveen; Divakaran, Ajay; Abdel-Mottaleb, Mohamed

2008-01-01

In this paper, we present a content-adaptive audio texture based method to segment video into audio scenes. The audio scene is modeled as a semantically consistent chunk of audio data. Our algorithm is based on "semantic audio texture analysis." At first, we train GMM models for basic audio classes such as speech, music, etc. Then we define the semantic audio texture based on those classes. We study and present two types of scene changes, those corresponding to an overall audio texture change and those corresponding to a special "transition marker" used by the content creator, such as a short stretch of music in a sitcom or silence in dramatic content. Unlike prior work using genre specific heuristics, such as some methods presented for detecting commercials, we adaptively find out if such special transition markers are being used and if so, which of the base classes are being used as markers without any prior knowledge about the content. Our experimental results show that our proposed audio scene segmentation works well across a wide variety of broadcast content genres.

Digital audio watermarking fundamentals, techniques and challenges

CERN Document Server

Xiang, Yong; Yan, Bin

2017-01-01

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

Automatic Modulation Classification Based on Deep Learning for Unmanned Aerial Vehicles

Directory of Open Access Journals (Sweden)

Duona Zhang

2018-03-01

Full Text Available Deep learning has recently attracted much attention due to its excellent performance in processing audio, image, and video data. However, few studies are devoted to the field of automatic modulation classification (AMC. It is one of the most well-known research topics in communication signal recognition and remains challenging for traditional methods due to complex disturbance from other sources. This paper proposes a heterogeneous deep model fusion (HDMF method to solve the problem in a unified framework. The contributions include the following: (1 a convolutional neural network (CNN and long short-term memory (LSTM are combined by two different ways without prior knowledge involved; (2 a large database, including eleven types of single-carrier modulation signals with various noises as well as a fading channel, is collected with various signal-to-noise ratios (SNRs based on a real geographical environment; and (3 experimental results demonstrate that HDMF is very capable of coping with the AMC problem, and achieves much better performance when compared with the independent network.

Web Audio/Video Streaming Tool

Science.gov (United States)

Guruvadoo, Eranna K.

2003-01-01

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

Semantic Context Detection Using Audio Event Fusion

Directory of Open Access Journals (Sweden)

Cheng Wen-Huang

2006-01-01

Full Text Available Semantic-level content analysis is a crucial issue in achieving efficient content retrieval and management. We propose a hierarchical approach that models audio events over a time series in order to accomplish semantic context detection. Two levels of modeling, audio event and semantic context modeling, are devised to bridge the gap between physical audio features and semantic concepts. In this work, hidden Markov models (HMMs are used to model four representative audio events, that is, gunshot, explosion, engine, and car braking, in action movies. At the semantic context level, generative (ergodic hidden Markov model and discriminative (support vector machine (SVM approaches are investigated to fuse the characteristics and correlations among audio events, which provide cues for detecting gunplay and car-chasing scenes. The experimental results demonstrate the effectiveness of the proposed approaches and provide a preliminary framework for information mining by using audio characteristics.

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

Detecting double compression of audio signal

Science.gov (United States)

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

2010-01-01

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

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

DEFF Research Database (Denmark)

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

2014-01-01

procedure was used to reduce these phrases into a comprehensive set of attributes. Groups of experienced and inexperienced listeners determined nine and eight attributes, respectively. These attribute sets were combined by the listeners to produce a final set of 12 attributes: masking, calming, distraction......An experiment to determine the perceptual attributes of the experience of listening to a target audio program in the presence of an audio interferer was performed. The first stage was a free elicitation task in which a total of 572 phrases were produced. In the second stage, a consensus vocabulary...

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

DEFF Research Database (Denmark)

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

2017-01-01

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

CERN automatic audio-conference service

CERN Multimedia

Sierra Moral, R

2009-01-01

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

CERN automatic audio-conference service

CERN Document Server

Sierra Moral, R

2010-01-01

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

Debugging of Class-D Audio Power Amplifiers

DEFF Research Database (Denmark)

Crone, Lasse; Pedersen, Jeppe Arnsdorf; Mønster, Jakob Døllner

2012-01-01

Determining and optimizing the performance of a Class-D audio power amplier can be very dicult without knowledge of the use of audio performance measuring equipment and of how the various noise and distortion sources in uence the audio performance. This paper gives an introduction on how to measure...

Plasma protein corona modulates the vascular wall interaction of drug carriers in a material and donor specific manner.

Directory of Open Access Journals (Sweden)

Daniel J Sobczynski

Full Text Available The nanoscale plasma protein interaction with intravenously injected particulate carrier systems is known to modulate their organ distribution and clearance from the bloodstream. However, the role of this plasma protein interaction in prescribing the adhesion of carriers to the vascular wall remains relatively unknown. Here, we show that the adhesion of vascular-targeted poly(lactide-co-glycolic-acid (PLGA spheres to endothelial cells is significantly inhibited in human blood flow, with up to 90% reduction in adhesion observed relative to adhesion in simple buffer flow, depending on the particle size and the magnitude and pattern of blood flow. This reduced PLGA adhesion in blood flow is linked to the adsorption of certain high molecular weight plasma proteins on PLGA and is donor specific, where large reductions in particle adhesion in blood flow (>80% relative to buffer is seen with ∼60% of unique donor bloods while others exhibit moderate to no reductions. The depletion of high molecular weight immunoglobulins from plasma is shown to successfully restore PLGA vascular wall adhesion. The observed plasma protein effect on PLGA is likely due to material characteristics since the effect is not replicated with polystyrene or silica spheres. These particles effectively adhere to the endothelium at a higher level in blood over buffer flow. Overall, understanding how distinct plasma proteins modulate the vascular wall interaction of vascular-targeted carriers of different material characteristics would allow for the design of highly functional delivery vehicles for the treatment of many serious human diseases.

Design of an audio advertisement dataset

Science.gov (United States)

Fu, Yutao; Liu, Jihong; Zhang, Qi; Geng, Yuting

2015-12-01

Since more and more advertisements swarm into radios, it is necessary to establish an audio advertising dataset which could be used to analyze and classify the advertisement. A method of how to establish a complete audio advertising dataset is presented in this paper. The dataset is divided into four different kinds of advertisements. Each advertisement's sample is given in *.wav file format, and annotated with a txt file which contains its file name, sampling frequency, channel number, broadcasting time and its class. The classifying rationality of the advertisements in this dataset is proved by clustering the different advertisements based on Principal Component Analysis (PCA). The experimental results show that this audio advertisement dataset offers a reliable set of samples for correlative audio advertisement experimental studies.

Making the Switch to Digital Audio

Directory of Open Access Journals (Sweden)

Shannon Gwin Mitchell

2004-12-01

Full Text Available In this article, the authors describe the process of converting from analog to digital audio data. They address the step-by-step decisions that they made in selecting hardware and software for recording and converting digital audio, issues of system integration, and cost considerations. The authors present a brief description of how digital audio is being used in their current research project and how it has enhanced the “quality” of their qualitative research.

Efficient Audio Power Amplification - Challenges

DEFF Research Database (Denmark)

Andersen, Michael Andreas E.

2005-01-01

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

Acoustic signal recovery by thermal demodulation

Science.gov (United States)

Boullosa, R. R.; Santillán, Arturo O.

2006-10-01

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

Tunable microwave photonic filter free from baseband and carrier suppression effect not requiring single sideband modulation using a Mach-Zenhder configuration.

Science.gov (United States)

Mora, José; Ortigosa-Blanch, Arturo; Pastor, Daniel; Capmany, José

2006-08-21

We present a full theoretical and experimental analysis of a novel all-optical microwave photonic filter combining a mode-locked fiber laser and a Mach-Zenhder structure in cascade to a 2x1 electro-optic modulator. The filter is free from the carrier suppression effect and thus it does not require single sideband modulation. Positive and negative coefficients are obtained inherently in the system and the tunability is achieved by controlling the optical path difference of the Mach-Zenhder structure.

GUI Application for ATCA-based LLRF Carrier Board Management

CERN Document Server

Wychowaniak, Jan; Predki, Pawel; Napieralski, Andrzej

2011-01-01

The Advanced Telecommunications Computing Architecture (ATCA) standard describes an efficient and powerful platform, implementation of which was adopted to be used as a base for control systems in high energy physics. The ATCA platform is considered to be applied for the X-ray Free Electron Laser (X-FEL), being built at Deutsches Electronen- Synchrotron (DESY) in Hamburg, Germany. The Low Level Radio Frequency (LLRF) control system is composed of a few ATCA Carrier Boards. Carrier Board hosts Intelligent Platform Management Controller (IPMC), which is developed in compliance with the PICMG specifications. IPMC is responsible for management and monitoring of sub-modules installed on Carrier Boards and pluggable Advanced Mezzanine Card (AMC) modules. The ATCA Shelf Manager is the main control unit of a single ATCA crate, responsible for all power and fan modules and Carrier Boards installed in ATCA shelf. The device provides a system administrator with a set of control and diagnostic capabilities regarding the ...

Consequence of audio visual collection in school libraries

OpenAIRE

Kuri, Ramesh

2016-01-01

The collection of Audio-Visual in library plays important role in teaching and learning. The importance of audio visual (AV) technology in education should not be underestimated. If audio-visual collection in library is carefully planned and designed, it can provide a rich learning environment. In this article, an author discussed the consequences of Audio-Visual collection in libraries especially for students of school library

New audio applications of beryllium metal

International Nuclear Information System (INIS)

Sato, M.

1977-01-01

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

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

Towards Structural Analysis of Audio Recordings in the Presence of Musical Variations

Directory of Open Access Journals (Sweden)

Müller Meinard

2007-01-01

Full Text Available One major goal of structural analysis of an audio recording is to automatically extract the repetitive structure or, more generally, the musical form of the underlying piece of music. Recent approaches to this problem work well for music, where the repetitions largely agree with respect to instrumentation and tempo, as is typically the case for popular music. For other classes of music such as Western classical music, however, musically similar audio segments may exhibit significant variations in parameters such as dynamics, timbre, execution of note groups, modulation, articulation, and tempo progression. In this paper, we propose a robust and efficient algorithm for audio structure analysis, which allows to identify musically similar segments even in the presence of large variations in these parameters. To account for such variations, our main idea is to incorporate invariance at various levels simultaneously: we design a new type of statistical features to absorb microvariations, introduce an enhanced local distance measure to account for local variations, and describe a new strategy for structure extraction that can cope with the global variations. Our experimental results with classical and popular music show that our algorithm performs successfully even in the presence of significant musical variations.

Efficient audio power amplification - challenges

Energy Technology Data Exchange (ETDEWEB)

Andersen, Michael A.E.

2005-07-01

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

Detection Of Alterations In Audio Files Using Spectrograph Analysis

Directory of Open Access Journals (Sweden)

Anandha Krishnan G

2015-08-01

Full Text Available The corresponding study was carried out to detect changes in audio file using spectrograph. An audio file format is a file format for storing digital audio data on a computer system. A sound spectrograph is a laboratory instrument that displays a graphical representation of the strengths of the various component frequencies of a sound as time passes. The objectives of the study were to find the changes in spectrograph of audio after altering them to compare altering changes with spectrograph of original files and to check for similarity and difference in mp3 and wav. Five different alterations were carried out on each audio file to analyze the differences between the original and the altered file. For altering the audio file MP3 or WAV by cutcopy the file was opened in Audacity. A different audio was then pasted to the audio file. This new file was analyzed to view the differences. By adjusting the necessary parameters the noise was reduced. The differences between the new file and the original file were analyzed. By adjusting the parameters from the dialog box the necessary changes were made. The edited audio file was opened in the software named spek where after analyzing a graph is obtained of that particular file which is saved for further analysis. The original audio graph received was combined with the edited audio file graph to see the alterations.

Enhanced audio-visual interactions in the auditory cortex of elderly cochlear-implant users.

Science.gov (United States)

Schierholz, Irina; Finke, Mareike; Schulte, Svenja; Hauthal, Nadine; Kantzke, Christoph; Rach, Stefan; Büchner, Andreas; Dengler, Reinhard; Sandmann, Pascale

2015-10-01

Auditory deprivation and the restoration of hearing via a cochlear implant (CI) can induce functional plasticity in auditory cortical areas. How these plastic changes affect the ability to integrate combined auditory (A) and visual (V) information is not yet well understood. In the present study, we used electroencephalography (EEG) to examine whether age, temporary deafness and altered sensory experience with a CI can affect audio-visual (AV) interactions in post-lingually deafened CI users. Young and elderly CI users and age-matched NH listeners performed a speeded response task on basic auditory, visual and audio-visual stimuli. Regarding the behavioral results, a redundant signals effect, that is, faster response times to cross-modal (AV) than to both of the two modality-specific stimuli (A, V), was revealed for all groups of participants. Moreover, in all four groups, we found evidence for audio-visual integration. Regarding event-related responses (ERPs), we observed a more pronounced visual modulation of the cortical auditory response at N1 latency (approximately 100 ms after stimulus onset) in the elderly CI users when compared with young CI users and elderly NH listeners. Thus, elderly CI users showed enhanced audio-visual binding which may be a consequence of compensatory strategies developed due to temporary deafness and/or degraded sensory input after implantation. These results indicate that the combination of aging, sensory deprivation and CI facilitates the coupling between the auditory and the visual modality. We suggest that this enhancement in multisensory interactions could be used to optimize auditory rehabilitation, especially in elderly CI users, by the application of strong audio-visually based rehabilitation strategies after implant switch-on. Copyright © 2015 Elsevier B.V. All rights reserved.

AudioMUD: a multiuser virtual environment for blind people.

Science.gov (United States)

Sánchez, Jaime; Hassler, Tiago

2007-03-01

A number of virtual environments have been developed during the last years. Among them there are some applications for blind people based on different type of audio, from simple sounds to 3-D audio. In this study, we pursued a different approach. We designed AudioMUD by using spoken text to describe the environment, navigation, and interaction. We have also introduced some collaborative features into the interaction between blind users. The core of a multiuser MUD game is a networked textual virtual environment. We developed AudioMUD by adding some collaborative features to the basic idea of a MUD and placed a simulated virtual environment inside the human body. This paper presents the design and usability evaluation of AudioMUD. Blind learners were motivated when interacted with AudioMUD and helped to improve the interaction through audio and interface design elements.

A novel research approach on the dynamic properties of photogenerated charge carriers at Ag{sub 2}S quantum-dots-sensitized TiO{sub 2} films by a frequency-modulated surface photovoltage technology

Energy Technology Data Exchange (ETDEWEB)

Zhang, Yu; Zhang, Wei [Liaoning Key Laboratory for Green Synthesis and Preparative Chemistry of Advanced Materials, College of Chemistry, Liaoning University, Shenyang 110036 (China); Xie, Tengfeng; Wang, Dejun [College of Chemistry, Jilin University, Changchun 130012 (China); Song, Xi-Ming, E-mail: songlab@lnu.edu.cn [Liaoning Key Laboratory for Green Synthesis and Preparative Chemistry of Advanced Materials, College of Chemistry, Liaoning University, Shenyang 110036 (China)

2013-09-01

Graphical abstract: The changed SPV with chopping frequencies indicate the separation speeds of photogenerated charge carriers in different films. - Highlights: • Ag{sub 2}S-sensitized TiO{sub 2} films show good photoelectric responses in visible-light region. • Frequency-modulated SPV give dynamic information and evidence of Ag{sub 2}S QDSSCs’ performance. • Frequency-modulated SPV can supply complementary information in the study of Ag{sub 2}S ODSSCs. - Abstract: Ag{sub 2}S quantum-dots-sensitized TiO{sub 2} films with different amount of Ag{sub 2}S were fabricated by a successive ionic layer adsorption and reaction (SILAR) method. The separation and transport of photogenerated charge carriers at different spectral regions were studied by the frequency-modulated surface photovoltage technology. Some novel dynamic information of photogenerated charge carriers in a wide spectral range is found. The results indicate that the rate and direction of separation (diffusion) for photogenerated charge carriers are closely related to the performance of quantum-dots-sensitized solar cells (QDSSCs) based on the Ag{sub 2}S/TiO{sub 2} nano-structure.

Audio Recording of Children with Dyslalia

OpenAIRE

Stefan Gheorghe Pentiuc; Maria D. Schipor; Ovidiu A. Schipor

2008-01-01

In this paper we present our researches regarding automat parsing of audio recordings. These recordings are obtained from children with dyslalia and are necessary for an accurate identification of speech problems. We develop a software application that helps parsing audio, real time, recordings.

Parametric time-frequency domain spatial audio

CERN Document Server

Delikaris-Manias, Symeon; Politis, Archontis

2018-01-01

This book provides readers with the principles and best practices in spatial audio signal processing. It describes how sound fields and their perceptual attributes are captured and analyzed within the time-frequency domain, how essential representation parameters are coded, and how such signals are efficiently reproduced for practical applications. The book is split into four parts starting with an overview of the fundamentals. It then goes on to explain the reproduction of spatial sound before offering an examination of signal-dependent spatial filtering. The book finishes with coverage of both current and future applications and the direction that spatial audio research is heading in. Parametric Time-frequency Domain Spatial Audio focuses on applications in entertainment audio, including music, home cinema, and gaming--covering the capturing and reproduction of spatial sound as well as its generation, transduction, representation, transmission, and perception. This book will teach readers the tools needed...

Predicting the Overall Spatial Quality of Automotive Audio Systems

Science.gov (United States)

Koya, Daisuke

The spatial quality of automotive audio systems is often compromised due to their unideal listening environments. Automotive audio systems need to be developed quickly due to industry demands. A suitable perceptual model could evaluate the spatial quality of automotive audio systems with similar reliability to formal listening tests but take less time. Such a model is developed in this research project by adapting an existing model of spatial quality for automotive audio use. The requirements for the adaptation were investigated in a literature review. A perceptual model called QESTRAL was reviewed, which predicts the overall spatial quality of domestic multichannel audio systems. It was determined that automotive audio systems are likely to be impaired in terms of the spatial attributes that were not considered in developing the QESTRAL model, but metrics are available that might predict these attributes. To establish whether the QESTRAL model in its current form can accurately predict the overall spatial quality of automotive audio systems, MUSHRA listening tests using headphone auralisation with head tracking were conducted to collect results to be compared against predictions by the model. Based on guideline criteria, the model in its current form could not accurately predict the overall spatial quality of automotive audio systems. To improve prediction performance, the QESTRAL model was recalibrated and modified using existing metrics of the model, those that were proposed from the literature review, and newly developed metrics. The most important metrics for predicting the overall spatial quality of automotive audio systems included those that were interaural cross-correlation (IACC) based, relate to localisation of the frontal audio scene, and account for the perceived scene width in front of the listener. Modifying the model for automotive audio systems did not invalidate its use for domestic audio systems. The resulting model predicts the overall spatial

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

Directory of Open Access Journals (Sweden)

Loizos G.

2014-07-01

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

Editing Audio with Audacity

Directory of Open Access Journals (Sweden)

Brandon Walsh

2016-08-01

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

14 CFR 382.69 - What requirements must carriers meet concerning the accessibility of videos, DVDs, and other...

Science.gov (United States)

2010-01-01

... concerning the accessibility of videos, DVDs, and other audio-visual presentations shown on-aircraft to individuals who are deaf or hard of hearing? 382.69 Section 382.69 Aeronautics and Space OFFICE OF THE... BASIS OF DISABILITY IN AIR TRAVEL Accessibility of Aircraft § 382.69 What requirements must carriers...

Fusion for Audio-Visual Laughter Detection

NARCIS (Netherlands)

Reuderink, B.

2007-01-01

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

AudioPairBank: Towards A Large-Scale Tag-Pair-Based Audio Content Analysis

OpenAIRE

Sager, Sebastian; Elizalde, Benjamin; Borth, Damian; Schulze, Christian; Raj, Bhiksha; Lane, Ian

2016-01-01

Recently, sound recognition has been used to identify sounds, such as car and river. However, sounds have nuances that may be better described by adjective-noun pairs such as slow car, and verb-noun pairs such as flying insects, which are under explored. Therefore, in this work we investigate the relation between audio content and both adjective-noun pairs and verb-noun pairs. Due to the lack of datasets with these kinds of annotations, we collected and processed the AudioPairBank corpus cons...

Tourism research and audio methods

DEFF Research Database (Denmark)

Jensen, Martin Trandberg

2016-01-01

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

Newnes audio and Hi-Fi engineer's pocket book

CERN Document Server

Capel, Vivian

2013-01-01

Newnes Audio and Hi-Fi Engineer's Pocket Book, Second Edition provides concise discussion of several audio topics. The book is comprised of 10 chapters that cover different audio equipment. The coverage of the text includes microphones, gramophones, compact discs, and tape recorders. The book also covers high-quality radio, amplifiers, and loudspeakers. The book then reviews the concepts of sound and acoustics, and presents some facts and formulas relevant to audio. The text will be useful to sound engineers and other professionals whose work involves sound systems.

DAFX Digital Audio Effects

CERN Document Server

Zö

2011-01-01

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

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.

47 CFR 10.520 - Common audio attention signal.

Science.gov (United States)

2010-10-01

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

Audio Recording of Children with Dyslalia

Directory of Open Access Journals (Sweden)

Stefan Gheorghe Pentiuc

2008-01-01

Full Text Available In this paper we present our researches regarding automat parsing of audio recordings. These recordings are obtained from children with dyslalia and are necessary for an accurate identification of speech problems. We develop a software application that helps parsing audio, real time, recordings.

Audio Journal in an ELT Context

Directory of Open Access Journals (Sweden)

Neşe Aysin Siyli

2012-09-01

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

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.

Music Genre Classification Using MIDI and Audio Features

Science.gov (United States)

Cataltepe, Zehra; Yaslan, Yusuf; Sonmez, Abdullah

2007-12-01

We report our findings on using MIDI files and audio features from MIDI, separately and combined together, for MIDI music genre classification. We use McKay and Fujinaga's 3-root and 9-leaf genre data set. In order to compute distances between MIDI pieces, we use normalized compression distance (NCD). NCD uses the compressed length of a string as an approximation to its Kolmogorov complexity and has previously been used for music genre and composer clustering. We convert the MIDI pieces to audio and then use the audio features to train different classifiers. MIDI and audio from MIDI classifiers alone achieve much smaller accuracies than those reported by McKay and Fujinaga who used not NCD but a number of domain-based MIDI features for their classification. Combining MIDI and audio from MIDI classifiers improves accuracy and gets closer to, but still worse, accuracies than McKay and Fujinaga's. The best root genre accuracies achieved using MIDI, audio, and combination of them are 0.75, 0.86, and 0.93, respectively, compared to 0.98 of McKay and Fujinaga. Successful classifier combination requires diversity of the base classifiers. We achieve diversity through using certain number of seconds of the MIDI file, different sample rates and sizes for the audio file, and different classification algorithms.

Suppression of high-frequency perturbations in pulse-width modulation

DEFF Research Database (Denmark)

2008-01-01

A method suppresses high-frequency perturbations in a pulse-width modulated signal. The pulse-width modulation may superpose a carrier signal onto an input signal having a predetermined modulation frequency. The carrier signals may be phase-shifted. The resulting modulated signals may...

Realtime Audio with Garbage Collection

OpenAIRE

Matheussen, Kjetil Svalastog

2010-01-01

Two non-moving concurrent garbage collectors tailored for realtime audio processing are described. Both collectors work on copies of the heap to avoid cache misses and audio-disruptive synchronizations. Both collectors are targeted at multiprocessor personal computers. The first garbage collector works in uncooperative environments, and can replace Hans Boehm's conservative garbage collector for C and C++. The collector does not access the virtual memory system. Neither doe...

Audio-Visual Classification of Sports Types

DEFF Research Database (Denmark)

Gade, Rikke; Abou-Zleikha, Mohamed; Christensen, Mads Græsbøll

2015-01-01

In this work we propose a method for classification of sports types from combined audio and visual features ex- tracted from thermal video. From audio Mel Frequency Cepstral Coefficients (MFCC) are extracted, and PCA are applied to reduce the feature space to 10 dimensions. From the visual modali...

CERN automatic audio-conference service

International Nuclear Information System (INIS)

Sierra Moral, Rodrigo

2010-01-01

Scientists from all over the world need to collaborate with CERN on a daily basis. They must be able to communicate effectively on their joint projects at any time; as a result telephone conferences have become indispensable and widely used. Managed by 6 operators, CERN already has more than 20000 hours and 5700 audio-conferences per year. However, the traditional telephone based audio-conference system needed to be modernized in three ways. Firstly, to provide the participants with more autonomy in the organization of their conferences; secondly, to eliminate the constraints of manual intervention by operators; and thirdly, to integrate the audio-conferences into a collaborative working framework. The large number, and hence cost, of the conferences prohibited externalization and so the CERN telecommunications team drew up a specification to implement a new system. It was decided to use a new commercial collaborative audio-conference solution based on the SIP protocol. The system was tested as the first European pilot and several improvements (such as billing, security, redundancy...) were implemented based on CERN's recommendations. The new automatic conference system has been operational since the second half of 2006. It is very popular for the users and has doubled the number of conferences in the past two years.

CERN automatic audio-conference service

Energy Technology Data Exchange (ETDEWEB)

Sierra Moral, Rodrigo, E-mail: Rodrigo.Sierra@cern.c [CERN, IT Department 1211 Geneva-23 (Switzerland)

2010-04-01

Scientists from all over the world need to collaborate with CERN on a daily basis. They must be able to communicate effectively on their joint projects at any time; as a result telephone conferences have become indispensable and widely used. Managed by 6 operators, CERN already has more than 20000 hours and 5700 audio-conferences per year. However, the traditional telephone based audio-conference system needed to be modernized in three ways. Firstly, to provide the participants with more autonomy in the organization of their conferences; secondly, to eliminate the constraints of manual intervention by operators; and thirdly, to integrate the audio-conferences into a collaborative working framework. The large number, and hence cost, of the conferences prohibited externalization and so the CERN telecommunications team drew up a specification to implement a new system. It was decided to use a new commercial collaborative audio-conference solution based on the SIP protocol. The system was tested as the first European pilot and several improvements (such as billing, security, redundancy...) were implemented based on CERN's recommendations. The new automatic conference system has been operational since the second half of 2006. It is very popular for the users and has doubled the number of conferences in the past two years.

CERN automatic audio-conference service

Science.gov (United States)

Sierra Moral, Rodrigo

2010-04-01

Scientists from all over the world need to collaborate with CERN on a daily basis. They must be able to communicate effectively on their joint projects at any time; as a result telephone conferences have become indispensable and widely used. Managed by 6 operators, CERN already has more than 20000 hours and 5700 audio-conferences per year. However, the traditional telephone based audio-conference system needed to be modernized in three ways. Firstly, to provide the participants with more autonomy in the organization of their conferences; secondly, to eliminate the constraints of manual intervention by operators; and thirdly, to integrate the audio-conferences into a collaborative working framework. The large number, and hence cost, of the conferences prohibited externalization and so the CERN telecommunications team drew up a specification to implement a new system. It was decided to use a new commercial collaborative audio-conference solution based on the SIP protocol. The system was tested as the first European pilot and several improvements (such as billing, security, redundancy...) were implemented based on CERN's recommendations. The new automatic conference system has been operational since the second half of 2006. It is very popular for the users and has doubled the number of conferences in the past two years.

Near-field Localization of Audio

DEFF Research Database (Denmark)

Jensen, Jesper Rindom; Christensen, Mads Græsbøll

2014-01-01

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

Musical Audio Synthesis Using Autoencoding Neural Nets

OpenAIRE

Sarroff, Andy; Casey, Michael A.

2014-01-01

With an optimal network topology and tuning of hyperpa-\\ud rameters, artificial neural networks (ANNs) may be trained\\ud to learn a mapping from low level audio features to one\\ud or more higher-level representations. Such artificial neu-\\ud ral networks are commonly used in classification and re-\\ud gression settings to perform arbitrary tasks. In this work\\ud we suggest repurposing autoencoding neural networks as\\ud musical audio synthesizers. We offer an interactive musi-\\ud cal audio synt...

An Estimation Method for number of carrier frequency

Directory of Open Access Journals (Sweden)

Xiong Peng

2015-01-01

Full Text Available This paper proposes a method that utilizes AR model power spectrum estimation based on Burg algorithm to estimate the number of carrier frequency in single pulse. In the modern electronic and information warfare, the pulse signal form of radar is complex and changeable, among which single pulse with multi-carrier frequencies is the most typical one, such as the frequency shift keying (FSK signal, the frequency shift keying with linear frequency (FSK-LFM hybrid modulation signal and the frequency shift keying with bi-phase shift keying (FSK-BPSK hybrid modulation signal. In view of this kind of single pulse which has multi-carrier frequencies, this paper adopts a method which transforms the complex signal into AR model, then takes power spectrum based on Burg algorithm to show the effect. Experimental results show that the estimation method still can determine the number of carrier frequencies accurately even when the signal noise ratio (SNR is very low.

Spatial audio reproduction with primary ambient extraction

CERN Document Server

He, JianJun

2017-01-01

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

Music Genre Classification Using MIDI and Audio Features

Directory of Open Access Journals (Sweden)

Abdullah Sonmez

2007-01-01

Full Text Available We report our findings on using MIDI files and audio features from MIDI, separately and combined together, for MIDI music genre classification. We use McKay and Fujinaga's 3-root and 9-leaf genre data set. In order to compute distances between MIDI pieces, we use normalized compression distance (NCD. NCD uses the compressed length of a string as an approximation to its Kolmogorov complexity and has previously been used for music genre and composer clustering. We convert the MIDI pieces to audio and then use the audio features to train different classifiers. MIDI and audio from MIDI classifiers alone achieve much smaller accuracies than those reported by McKay and Fujinaga who used not NCD but a number of domain-based MIDI features for their classification. Combining MIDI and audio from MIDI classifiers improves accuracy and gets closer to, but still worse, accuracies than McKay and Fujinaga's. The best root genre accuracies achieved using MIDI, audio, and combination of them are 0.75, 0.86, and 0.93, respectively, compared to 0.98 of McKay and Fujinaga. Successful classifier combination requires diversity of the base classifiers. We achieve diversity through using certain number of seconds of the MIDI file, different sample rates and sizes for the audio file, and different classification algorithms.

Flattened optical frequency-locked multi-carrier generation by cascading one DML and one phase modulator driven by different RF frequency clocks

International Nuclear Information System (INIS)

Li, Xinying; Yu, Jianjun; Zhang, Junwen; Chi, Nan

2013-01-01

We propose a novel scheme for flattened optical frequency-locked multi-carrier generation based on one directly modulated laser (DML) and one phase modulator (PM) in cascade driven by different sinusoidal radio-frequency (RF) clocks. We experimentally demonstrate that when the clock frequencies for the cascaded DML and the PM are respectively 12.5 GHz and 25 GHz, over 24 optical subcarriers can be generated with 12.5-GHz frequency spacing and amplitude fluctuation less than 3 dB. Furthermore, the number of generated optical subcarriers can be further increased when we increase the driving power for the DML. (letter)

Current-Driven Switch-Mode Audio Power Amplifiers

DEFF Research Database (Denmark)

Knott, Arnold; Buhl, Niels Christian; Andersen, Michael A. E.

2012-01-01

The conversion of electrical energy into sound waves by electromechanical transducers is proportional to the current through the coil of the transducer. However virtually all audio power amplifiers provide a controlled voltage through the interface to the transducer. This paper is presenting...... a switch-mode audio power amplifier not only providing controlled current but also being supplied by current. This results in an output filter size reduction by a factor of 6. The implemented prototype shows decent audio performance with THD + N below 0.1 %....

In-band 16-QAM and multi-carrier SCM modulation to label DPSK payload signals for IP packet routing.

Science.gov (United States)

Tafur Monroy, Idelfonso; Vegas Olmos, Juan; Garcia Larrode, Maria; Koonen, Ton; Díaz Jiménez, Cristina

2006-02-06

We present an experimental demonstration of the feasibility of in-band subcarrier multiplexing (SCM) for labeling of differential phase shift keying (DPSK) payload signals. We show that by proper selection of the value of the subcarrier frequency the effect of the superimposed SCM label on the performance of the DPSK signal is minimized. Furthermore, we show experimentally the advantages of using alternative modulation formats such as 16-QAM and multi-carrier SCM for optical labeling of a 10 Gb/s DPSK payload signal.

METHOD AND MODULE FOR OPTICAL SUBCARRIER LABELLING

DEFF Research Database (Denmark)

2004-01-01

The present invention relates to optical labelling in WDM networks, in that it provides a method and a module to be used in subcarrier label generation and switching in network edge nodes and core switch nodes. The methods and modules are typically employed in Optical Subcarrier Multiplexing (OSCM......) transmitters. The payload and the label are encoded independently on optical carrier and subcarrier signals respectively, using electro-optical modulators. The invention applies single or double sideband carrier-suppressed modulation to generate subcarrier signals for encoding of the label. Thereby the payload...... encoded carrier signal and the label encoded subcarrier signal can be coupled directly without prior filtering....

Joint Carrier-Phase Synchronization and LDPC Decoding

Science.gov (United States)

Simon, Marvin; Valles, Esteban

2009-01-01

A method has been proposed to increase the degree of synchronization of a radio receiver with the phase of a suppressed carrier signal modulated with a binary- phase-shift-keying (BPSK) or quaternary- phase-shift-keying (QPSK) signal representing a low-density parity-check (LDPC) code. This method is an extended version of the method described in Using LDPC Code Constraints to Aid Recovery of Symbol Timing (NPO-43112), NASA Tech Briefs, Vol. 32, No. 10 (October 2008), page 54. Both methods and the receiver architectures in which they would be implemented belong to a class of timing- recovery methods and corresponding receiver architectures characterized as pilotless in that they do not require transmission and reception of pilot signals. The proposed method calls for the use of what is known in the art as soft decision feedback to remove the modulation from a replica of the incoming signal prior to feeding this replica to a phase-locked loop (PLL) or other carrier-tracking stage in the receiver. Soft decision feedback refers to suitably processed versions of intermediate results of iterative computations involved in the LDPC decoding process. Unlike a related prior method in which hard decision feedback (the final sequence of decoded symbols) is used to remove the modulation, the proposed method does not require estimation of the decoder error probability. In a basic digital implementation of the proposed method, the incoming signal (having carrier phase theta theta (sub c) plus noise would first be converted to inphase (I) and quadrature (Q) baseband signals by mixing it with I and Q signals at the carrier frequency [wc/(2 pi)] generated by a local oscillator. The resulting demodulated signals would be processed through one-symbol-period integrate and- dump filters, the outputs of which would be sampled and held, then multiplied by a soft-decision version of the baseband modulated signal. The resulting I and Q products consist of terms proportional to the cosine

High-Order Sparse Linear Predictors for Audio Processing

DEFF Research Database (Denmark)

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

2010-01-01

Linear prediction has generally failed to make a breakthrough in audio processing, as it has done in speech processing. This is mostly due to its poor modeling performance, since an audio signal is usually an ensemble of different sources. Nevertheless, linear prediction comes with a whole set...... of interesting features that make the idea of using it in audio processing not far fetched, e.g., the strong ability of modeling the spectral peaks that play a dominant role in perception. In this paper, we provide some preliminary conjectures and experiments on the use of high-order sparse linear predictors...... in audio processing. These predictors, successfully implemented in modeling the short-term and long-term redundancies present in speech signals, will be used to model tonal audio signals, both monophonic and polyphonic. We will show how the sparse predictors are able to model efficiently the different...

Audio Mining with emphasis on Music Genre Classification

DEFF Research Database (Denmark)

Meng, Anders

2004-01-01

Audio is an important part of our daily life, basically it increases our impression of the world around us whether this is communication, music, danger detection etc. Currently the field of Audio Mining, which here includes areas of music genre, music recognition / retrieval, playlist generation...... the world the problem of detecting environments from the input audio is researched as to increase the life quality of hearing-impaired. Basically there is a lot of work within the field of audio mining. The presentation will mainly focus on music genre classification where we have a fixed amount of genres...... to choose from. Basically every audio mining system is more or less consisting of the same stages as for the music genre setting. My research so far has mainly focussed on finding relevant features for music genre classification living at different timescales using early and late information fusion. It has...

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

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.

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

Science.gov (United States)

George, Rohini

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

Fusion of audio and visual cues for laughter detection

NARCIS (Netherlands)

Petridis, Stavros; Pantic, Maja

Past research on automatic laughter detection has focused mainly on audio-based detection. Here we present an audio- visual approach to distinguishing laughter from speech and we show that integrating the information from audio and video channels leads to improved performance over single-modal

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.

Perceptual Audio Hashing Functions

Directory of Open Access Journals (Sweden)

Emin Anarım

2005-07-01

Full Text Available Perceptual hash functions provide a tool for fast and reliable identification of content. We present new audio hash functions based on summarization of the time-frequency spectral characteristics of an audio document. The proposed hash functions are based on the periodicity series of the fundamental frequency and on singular-value description of the cepstral frequencies. They are found, on one hand, to perform very satisfactorily in identification and verification tests, and on the other hand, to be very resilient to a large variety of attacks. Moreover, we address the issue of security of hashes and propose a keying technique, and thereby a key-dependent hash function.

Horatio Audio-Describes Shakespeare's "Hamlet": Blind and Low-Vision Theatre-Goers Evaluate an Unconventional Audio Description Strategy

Science.gov (United States)

Udo, J. P.; Acevedo, B.; Fels, D. I.

2010-01-01

Audio description (AD) has been introduced as one solution for providing people who are blind or have low vision with access to live theatre, film and television content. However, there is little research to inform the process, user preferences and presentation style. We present a study of a single live audio-described performance of Hart House…

String Stretching, Frequency Modulation, and Banjo Clang

OpenAIRE

Politzer, David

2014-01-01

The banjo’s floating bridge, string break angle, and flexible drumhead all contribute to substantial audio range frequency modulation. From the world of electronic music synthesis, it is known that modulating higher frequency sounds with lower acoustic frequencies leads to metallic and bell-like tone. The mechanics of the banjo does just that quite naturally, modulating fundamentals and harmonics with the motion of the bridge. In technical terms, with a floating bridge, string stretching is f...

Audio localization for mobile robots

OpenAIRE

de Guillebon, Thibaut; Grau Saldes, Antoni; Bolea Monte, Yolanda

2009-01-01

The department of the University for which I worked is developing a project based on the interaction with robots in the environment. My work was to define an audio system for the robot. This audio system that I have to realize consists on a mobile head which is able to follow the sound in its environment. This subject was treated as a research problem, with the liberty to find and develop different solutions and make them evolve in the chosen way.

EVALUASI KEPUASAN PENGGUNA TERHADAP APLIKASI AUDIO BOOKS

Directory of Open Access Journals (Sweden)

Raditya Maulana Anuraga

2017-02-01

Full Text Available Listeno is the first application audio books in Indonesia so that the users can get the book in audio form like listen to music, Listeno have problems in a feature request Listeno offline mode that have not been released, a security problem mp3 files that must be considered, and the target Listeno not yet reached 100,000 active users. This research has the objective to evaluate user satisfaction to Audio Books with research method approach, Nielsen. The analysis in this study using Importance Performance Analysis (IPA is combined with the index of User Satisfaction (IKP based on the indicators used are: Benefit (Usefulness, Utility (Utility, Usability (Usability, easy to understand (Learnability, Efficient (efficiency , Easy to remember (Memorability, Error (Error, and satisfaction (satisfaction. The results showed Applications User Satisfaction Audio books are quite satisfied with the results of the calculation IKP 69.58%..

Asymmetric Carrier Random PWM

DEFF Research Database (Denmark)

Mathe, Laszlo; Lungeanu, Florin; Rasmussen, Peter Omand

2010-01-01

index. The flat motor current spectrum generates an acoustical noise close to the white noise, which may improve the acoustical performance of the drive. The new carrier wave is easy to implement digitally, without employing any external circuits. The modulation method can be used in open, as well...

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

Removable Watermarking Sebagai Pengendalian Terhadap Cyber Crime Pada Audio Digital

Directory of Open Access Journals (Sweden)

Reyhani Lian Putri

2017-08-01

Full Text Available Perkembangan teknologi informasi yang pesat menuntut penggunanya untuk lebih berhati-hati seiring semakin meningkatnya cyber crime.Banyak pihak telah mengembangkan berbagai teknik perlindungan data digital, salah satunya adalah watermarking. Teknologi watermarking berfungsi untuk memberikan identitas, melindungi, atau menandai data digital, baik audio, citra, ataupun video, yang mereka miliki. Akan tetapi, teknik tersebut masih dapat diretas oleh oknum-oknum yang tidak bertanggung jawab.Pada penelitian ini, proses watermarking diterapkan pada audio digital dengan menyisipkan watermark yang terdengar jelas oleh indera pendengaran manusia (perceptible pada audio host.Hal ini bertujuan agar data audio dapat terlindungi dan apabila ada pihak lain yang ingin mendapatkan data audio tersebut harus memiliki “kunci” untuk menghilangkan watermark. Proses removable watermarking ini dilakukan pada data watermark yang sudah diketahui metode penyisipannya, agar watermark dapat dihilangkan sehingga kualitas audio menjadi lebih baik. Dengan menggunakan metode ini diperoleh kinerja audio watermarking pada nilai distorsi tertinggi dengan rata-rata nilai SNR sebesar7,834 dB dan rata-rata nilai ODG sebesar -3,77.Kualitas audio meningkat setelah watermark dihilangkan, di mana rata-rata SNR menjadi sebesar 24,986 dB dan rata-rata ODG menjadi sebesar -1,064 serta nilai MOS sebesar 4,40.

A listening test system for automotive audio

DEFF Research Database (Denmark)

Christensen, Flemming; Geoff, Martin; Minnaar, Pauli

2005-01-01

This paper describes a system for simulating automotive audio through headphones for the purposes of conducting listening experiments in the laboratory. The system is based on binaural technology and consists of a component for reproducing the sound of the audio system itself and a component...

Voice activity detection using audio-visual information

DEFF Research Database (Denmark)

Petsatodis, Theodore; Pnevmatikakis, Aristodemos; Boukis, Christos

2009-01-01

An audio-visual voice activity detector that uses sensors positioned distantly from the speaker is presented. Its constituting unimodal detectors are based on the modeling of the temporal variation of audio and visual features using Hidden Markov Models; their outcomes are fused using a post...

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

Directory of Open Access Journals (Sweden)

Petar S. Aleksic

2002-11-01

Full Text Available We describe an audio-visual automatic continuous speech recognition system, which significantly improves speech recognition performance over a wide range of acoustic noise levels, as well as under clean audio conditions. The system utilizes facial animation parameters (FAPs supported by the MPEG-4 standard for the visual representation of speech. We also describe a robust and automatic algorithm we have developed to extract FAPs from visual data, which does not require hand labeling or extensive training procedures. The principal component analysis (PCA was performed on the FAPs in order to decrease the dimensionality of the visual feature vectors, and the derived projection weights were used as visual features in the audio-visual automatic speech recognition (ASR experiments. Both single-stream and multistream hidden Markov models (HMMs were used to model the ASR system, integrate audio and visual information, and perform a relatively large vocabulary (approximately 1000 words speech recognition experiments. The experiments performed use clean audio data and audio data corrupted by stationary white Gaussian noise at various SNRs. The proposed system reduces the word error rate (WER by 20% to 23% relatively to audio-only speech recognition WERs, at various SNRs (0–30 dB with additive white Gaussian noise, and by 19% relatively to audio-only speech recognition WER under clean audio conditions.

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

Science.gov (United States)

Egi, Noritsugu; Hayashi, Takanori; Takahashi, Akira

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

Audio power amplifier design handbook

CERN Document Server

Self, Douglas

2013-01-01

This book is essential for audio power amplifier designers and engineers for one simple reason...it enables you as a professional to develop reliable, high-performance circuits. The Author Douglas Self covers the major issues of distortion and linearity, power supplies, overload, DC-protection and reactive loading. He also tackles unusual forms of compensation and distortion produced by capacitors and fuses. This completely updated fifth edition includes four NEW chapters including one on The XD Principle, invented by the author, and used by Cambridge Audio. Cro

Audio stream classification for multimedia database search

Science.gov (United States)

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

2013-03-01

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

MP3 audio-editing software for the department of radiology

International Nuclear Information System (INIS)

Hong Qingfen; Sun Canhui; Li Ziping; Meng Quanfei; Jiang Li

2006-01-01

Objective: To evaluate the MP3 audio-editing software in the daily work in the department of radiology. Methods: The audio content of daily consultation seminar, held in the department of radiology every morning, was recorded and converted into MP3 audio format by a computer integrated recording device. The audio data were edited, archived, and eventually saved in the computer memory storage media, which was experimentally replayed and applied in the research or teaching. Results: MP3 audio-editing was a simple process and convenient for saving and searching the data. The record could be easily replayed. Conclusion: MP3 audio-editing perfectly records and saves the contents of consultation seminar, and has replaced the conventional hand writing notes. It is a valuable tool in both research and teaching in the department. (authors)

Automated processing of massive audio/video content using FFmpeg

Directory of Open Access Journals (Sweden)

Kia Siang Hock

2014-01-01

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

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...

Tune in the Net with RealAudio.

Science.gov (United States)

Buchanan, Larry

1997-01-01

Describes how to connect to the RealAudio Web site to download a player that provides sound from Web pages to the computer through streaming technology. Explains hardware and software requirements and provides addresses for other RealAudio Web sites are provided, including weather information and current news. (LRW)

Car audio using DSP for active sound control. DSP ni yoru active seigyo wo mochiita audio

Energy Technology Data Exchange (ETDEWEB)

Yamada, K.; Asano, S.; Furukawa, N. (Mitsubishi Motor Corp., Tokyo (Japan))

1993-06-01

In the automobile cabin, there are some unique problems which spoil the quality of sound reproduction from audio equipment, such as the narrow space and/or the background noise. The audio signal processing by using DSP (digital signal processor) makes enable a solution to these problems. A car audio with a high amenity has been successfully made by the active sound control using DSP. The DSP consists of an adder, coefficient multiplier, delay unit, and connections. For the actual processing by DSP, are used functions, such as sound field correction, response and processing of noises during driving, surround reproduction, graphic equalizer processing, etc. High effectiveness of the method was confirmed through the actual driving evaluation test. The present paper describes the actual method of sound control technology using DSP. Especially, the dynamic processing of the noise during driving is discussed in detail. 1 ref., 12 figs., 1 tab.

Portable audio electronics for impedance-based measurements in microfluidics

International Nuclear Information System (INIS)

Wood, Paul; Sinton, David

2010-01-01

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

Electrical power inverter having a phase modulated, twin-inverter, high frequency link and an energy storage module

Science.gov (United States)

Pitel, I.J.

1987-02-03

The present invention provides an electrical power inverter method and apparatus, which includes a high frequency link, for converting DC power into AC power. Generally stated, the apparatus includes a first high frequency module which produces an AC voltage at a first output frequency, and a second high frequency inverter module which produces an AC voltage at a second output frequency that is substantially the same as the first output frequency. The second AC voltage is out of phase with the first AC voltage by a selected angular phase displacement. A mixer mixes the first and second output voltages to produce a high frequency carrier which has a selected base frequency impressed on the sidebands thereof. A rectifier rectifies the carrier, and a filter filters the rectified carrier. An output inverter inverts the filtered carrier to produce an AC line voltage at the selected base frequency. A phase modulator adjusts the relative angular phase displacement between the outputs of the first and second high frequency modules to control the base frequency and magnitude of the AC line voltage. 19 figs.

Electrical power inverter having a phase modulated, twin-inverter, high frequency link and an energy storage module

Science.gov (United States)

Pitel, Ira J.

1987-02-03

The present invention provides an electrical power inverter method and apparatus, which includes a high frequency link, for converting DC power into AC power. Generally stated, the apparatus includes a first high frequency module which produces an AC voltage at a first output frequency, and a second high frequency inverter module which produces an AC voltage at a second output frequency that is substantially the same as the first output frequency. The second AC voltage is out of phase with the first AC voltage by a selected angular phase displacement. A mixer mixes the first and second output voltages to produce a high frequency carrier which has a selected base frequency impressed on the sidebands thereof. A rectifier rectifies the carrier, and a filter filters the rectified carrier. An output inverter inverts the filtered carrier to produce an AC line voltage at the selected base frequency. A phase modulator adjusts the relative angular phase displacement between the outputs of the first and second high frequency modules to control the base frequency and magnitude of the AC line voltage.

Perceived Audio Quality Analysis in Digital Audio Broadcasting Plus System Based on PEAQ

Directory of Open Access Journals (Sweden)

K. Ulovec

2018-04-01

Full Text Available Broadcasters need to decide on bitrates of the services in the multiplex transmitted via Digital Audio Broadcasting Plus system. The bitrate should be set as low as possible for maximal number of services, but with high quality, not lower than in conventional analog systems. In this paper, the objective method Perceptual Evaluation of Audio Quality is used to analyze the perceived audio quality for appropriate codecs --- MP2 and AAC offering three profiles. The main aim is to determine dependencies on the type of signal --- music and speech, the number of channels --- stereo and mono, and the bitrate. Results indicate that only MP2 codec and AAC Low Complexity profile reach imperceptible quality loss. The MP2 codec needs higher bitrate than AAC Low Complexity profile for the same quality. For the both versions of AAC High-Efficiency profiles, the limit bitrates are determined above which less complex profiles outperform the more complex ones and higher bitrates above these limits are not worth using. It is shown that stereo music has worse quality than stereo speech generally, whereas for mono, the dependencies vary upon the codec/profile. Furthermore, numbers of services satisfying various quality criteria are presented.

Wide-band analog frequency modulation of optic signals using indirect techniques

Science.gov (United States)

Fitzmartin, D. J.; Balboni, E. J.; Gels, R. G.

1991-01-01

The wideband frequency modulation (FM) of an optical carrier by a radio frequency (RF) or microwave signal can be accomplished independent of laser type when indirect modulation is employed. Indirect modulators exploit the integral relation of phase to frequency so that phase modulators can be used to impress frequency modulation on an optical carrier. The use of integrated optics phase modulators, which are highly linear, enables the generation of optical wideband FM signals with very low intermodulation distortion. This modulator can be used as part of an optical wideband FM link for RF and microwave signals. Experimental results from the test of an indirect frequency modulator for an optical carrier are discussed.

Auditory and audio-visual processing in patients with cochlear, auditory brainstem, and auditory midbrain implants: An EEG study.

Science.gov (United States)

Schierholz, Irina; Finke, Mareike; Kral, Andrej; Büchner, Andreas; Rach, Stefan; Lenarz, Thomas; Dengler, Reinhard; Sandmann, Pascale

2017-04-01

There is substantial variability in speech recognition ability across patients with cochlear implants (CIs), auditory brainstem implants (ABIs), and auditory midbrain implants (AMIs). To better understand how this variability is related to central processing differences, the current electroencephalography (EEG) study compared hearing abilities and auditory-cortex activation in patients with electrical stimulation at different sites of the auditory pathway. Three different groups of patients with auditory implants (Hannover Medical School; ABI: n = 6, CI: n = 6; AMI: n = 2) performed a speeded response task and a speech recognition test with auditory, visual, and audio-visual stimuli. Behavioral performance and cortical processing of auditory and audio-visual stimuli were compared between groups. ABI and AMI patients showed prolonged response times on auditory and audio-visual stimuli compared with NH listeners and CI patients. This was confirmed by prolonged N1 latencies and reduced N1 amplitudes in ABI and AMI patients. However, patients with central auditory implants showed a remarkable gain in performance when visual and auditory input was combined, in both speech and non-speech conditions, which was reflected by a strong visual modulation of auditory-cortex activation in these individuals. In sum, the results suggest that the behavioral improvement for audio-visual conditions in central auditory implant patients is based on enhanced audio-visual interactions in the auditory cortex. Their findings may provide important implications for the optimization of electrical stimulation and rehabilitation strategies in patients with central auditory prostheses. Hum Brain Mapp 38:2206-2225, 2017. © 2017 Wiley Periodicals, Inc. © 2017 Wiley Periodicals, Inc.

Modified DCTNet for audio signals classification

Science.gov (United States)

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

2016-10-01

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

Carrier introduction to moire pattern for automatic fringe-order distinguishing

International Nuclear Information System (INIS)

Fang, J.; Laermann, K.H.

1992-01-01

This paper presents an automatic procedure of pseudo-colour encoding of moire fringe orders. A carrier consisting of parallel fringes is introduced before the specimen deforms. The carrier pattern is captured by a camera and then stored in computer as a standard image. The space of the carrier fringes is distored by the strains on the specimen as it is loaded. On a certain condition, the orders of the frequency-modulated carrier still vary monotonically so that they can be easyly distinguished. Both the standard fringe-carrier and the frequency-modulated fringe pattern are transformed into two digital images, of which every fringe is encoded by one of the pseudo-colour codes corresponding to the monotonical fringe orders. At each pixel, the difference between the colour sequences of two images is calculated to obtain the fringe order of pure deformation. The moire pattern of the in-plane displacement is restored as a pseudo-colour image by whose colour-change the variation of the fringe orders is displayed. (orig.)

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. 

Efficiency Optimization in Class-D Audio Amplifiers

DEFF Research Database (Denmark)

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

2015-01-01

This paper presents a new power efficiency optimization routine for designing Class-D audio amplifiers. The proposed optimization procedure finds design parameters for the power stage and the output filter, and the optimum switching frequency such that the weighted power losses are minimized under...... the given constraints. The optimization routine is applied to minimize the power losses in a 130 W class-D audio amplifier based on consumer behavior investigations, where the amplifier operates at idle and low power levels most of the time. Experimental results demonstrate that the optimization method can...... lead to around 30 % of efficiency improvement at 1.3 W output power without significant effects on both audio performance and the efficiency at high power levels....

DOA Estimation of Audio Sources in Reverberant Environments

DEFF Research Database (Denmark)

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

2016-01-01

Reverberation is well-known to have a detrimental impact on many localization methods for audio sources. We address this problem by imposing a model for the early reflections as well as a model for the audio source itself. Using these models, we propose two iterative localization methods...... that estimate the direction-of-arrival (DOA) of both the direct path of the audio source and the early reflections. In these methods, the contribution of the early reflections is essentially subtracted from the signal observations before localization of the direct path component, which may reduce the estimation...

Portable Audio Design

DEFF Research Database (Denmark)

Groth, Sanne Krogh

2014-01-01

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

Effects of Carrier Frequency Offset, Timing Offset, and Channel Spread Factor on the Performance of Hexagonal Multicarrier Modulation Systems

Directory of Open Access Journals (Sweden)

Kui Xu

2009-01-01

Full Text Available Hexagonal multicarrier modulation (HMM system is the technique of choice to overcome the impact of time-frequency dispersive transmission channel. This paper examines the effects of insufficient synchronization (carrier frequency offset, timing offset on the amplitude and phase of the demodulated symbol by using a projection receiver in hexagonal multicarrier modulation systems. Furthermore, effects of CFO, TO, and channel spread factor on the performance of signal-to-interference-plus-noise ratio (SINR in hexagonal multicarrier modulation systems are further discussed. The exact SINR expression versus insufficient synchronization and channel spread factor is derived. Theoretical analysis shows that similar degradation on symbol amplitude and phase caused by insufficient synchronization is incurred as in traditional cyclic prefix orthogonal frequency-division multiplexing (CP-OFDM transmission. Our theoretical analysis is confirmed by numerical simulations in a doubly dispersive (DD channel with exponential delay power profile and U-shape Doppler power spectrum, showing that HMM systems outperform traditional CP-OFDM systems with respect to SINR against ISI/ICI caused by insufficient synchronization and doubly dispersive channel.

Frequency-Diversity Reception for Phase Modulation

Science.gov (United States)

Brockman, M. H.

1984-01-01

Signal-to-noise ratio improved. System receives phase modulation transmitted simultaneously on different carrier frequencies. Used for carriers received through different antennas or through same antenna.

Subcarrier intensity modulation for MIMO visible light communications

Science.gov (United States)

Celik, Yasin; Akan, Aydin

2018-04-01

In this paper, subcarrier intensity modulation (SIM) is investigated for multiple-input multiple-output (MIMO) visible light communication (VLC) systems. A new modulation scheme called DC-aid SIM (DCA-SIM) is proposed for the spatial modulation (SM) transmission plan. Then, DCA-SIM is extended for multiple subcarrier case which is called DC-aid Multiple Subcarrier Modulation (DCA-MSM). Bit error rate (BER) performances of the considered system are analyzed for different MIMO schemes. The power efficiencies of DCA-SIM and DCA-MSM are shown in correlated MIMO VLC channels. The upper bound BER performances of the proposed models are obtained analytically for PSK and QAM modulation types in order to validate the simulation results. Additionally, the effect of power imbalance method on the performance of SIM is studied and remarkable power gains are obtained compared to the non-power imbalanced cases. In this work, Pulse amplitude modulation (PAM) and MSM-Index are used as benchmarks for single carrier and multiple carrier cases, respectively. And the results show that the proposed schemes outperform PAM and MSM-Index for considered single carrier and multiple carrier communication scenarios.

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

Directory of Open Access Journals (Sweden)

Smita Paira

2016-09-01

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

Differential carrier lifetime and transport effects in electrically injected III-nitride light-emitting diodes

Science.gov (United States)

Rashidi, A.; Nami, M.; Monavarian, M.; Aragon, A.; DaVico, K.; Ayoub, F.; Mishkat-Ul-Masabih, S.; Rishinaramangalam, A.; Feezell, D.

2017-07-01

This work describes a small-signal microwave method for determining the differential carrier lifetime and transport effects in electrically injected InGaN/GaN light-emitting diodes (LEDs). By considering the carrier diffusion, capture, thermionic escape, and recombination, the rate equations are used to derive an equivalent small-signal electrical circuit for the LEDs, from which expressions for the input impedance and modulation response are obtained. The expressions are simultaneously fit to the experimental data for the input impedance and modulation response for nonpolar InGaN/GaN micro-LEDs on free-standing GaN substrates. The fittings are used to extract the transport related circuit parameters and differential carrier lifetimes. The dependence of the parameters on the device diameter and current density is reported. We also derive approximations for the modulation response under low and high injection levels and show that the transport of carriers affects the modulation response of the device, especially at low injection levels. The methods presented are relevant to the design of high-speed LEDs for visible-light communication.

Adaptive DCTNet for Audio Signal Classification

OpenAIRE

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

2016-01-01

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

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.

Tuning of Human Modulation Filters Is Carrier-Frequency Dependent

Science.gov (United States)

Simpson, Andrew J. R.; Reiss, Joshua D.; McAlpine, David

2013-01-01

Recent studies employing speech stimuli to investigate ‘cocktail-party’ listening have focused on entrainment of cortical activity to modulations at syllabic (5 Hz) and phonemic (20 Hz) rates. The data suggest that cortical modulation filters (CMFs) are dependent on the sound-frequency channel in which modulations are conveyed, potentially underpinning a strategy for separating speech from background noise. Here, we characterize modulation filters in human listeners using a novel behavioral method. Within an ‘inverted’ adaptive forced-choice increment detection task, listening level was varied whilst contrast was held constant for ramped increments with effective modulation rates between 0.5 and 33 Hz. Our data suggest that modulation filters are tonotopically organized (i.e., vary along the primary, frequency-organized, dimension). This suggests that the human auditory system is optimized to track rapid (phonemic) modulations at high sound-frequencies and slow (prosodic/syllabic) modulations at low frequencies. PMID:24009759

Audio-Tutorial Instruction: A Strategy For Teaching Introductory College Geology.

Science.gov (United States)

Fenner, Peter; Andrews, Ted F.

The rationale of audio-tutorial instruction is discussed, and the history and development of the audio-tutorial botany program at Purdue University is described. Audio-tutorial programs in geology at eleven colleges and one school are described, illustrating several ways in which programs have been developed and integrated into courses. Programs…

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.

Assessing associations between the AURKA-HMMR-TPX2-TUBG1 functional module and breast cancer risk in BRCA1/2 mutation carriers.

Directory of Open Access Journals (Sweden)

Ignacio Blanco

Full Text Available While interplay between BRCA1 and AURKA-RHAMM-TPX2-TUBG1 regulates mammary epithelial polarization, common genetic variation in HMMR (gene product RHAMM may be associated with risk of breast cancer in BRCA1 mutation carriers. Following on these observations, we further assessed the link between the AURKA-HMMR-TPX2-TUBG1 functional module and risk of breast cancer in BRCA1 or BRCA2 mutation carriers. Forty-one single nucleotide polymorphisms (SNPs were genotyped in 15,252 BRCA1 and 8,211 BRCA2 mutation carriers and subsequently analyzed using a retrospective likelihood approach. The association of HMMR rs299290 with breast cancer risk in BRCA1 mutation carriers was confirmed: per-allele hazard ratio (HR = 1.10, 95% confidence interval (CI 1.04-1.15, p = 1.9 x 10(-4 (false discovery rate (FDR-adjusted p = 0.043. Variation in CSTF1, located next to AURKA, was also found to be associated with breast cancer risk in BRCA2 mutation carriers: rs2426618 per-allele HR = 1.10, 95% CI 1.03-1.16, p = 0.005 (FDR-adjusted p = 0.045. Assessment of pairwise interactions provided suggestions (FDR-adjusted pinteraction values > 0.05 for deviations from the multiplicative model for rs299290 and CSTF1 rs6064391, and rs299290 and TUBG1 rs11649877 in both BRCA1 and BRCA2 mutation carriers. Following these suggestions, the expression of HMMR and AURKA or TUBG1 in sporadic breast tumors was found to potentially interact, influencing patients' survival. Together, the results of this study support the hypothesis of a causative link between altered function of AURKA-HMMR-TPX2-TUBG1 and breast carcinogenesis in BRCA1/2 mutation carriers.

Space Experiment Module (SEM)

Science.gov (United States)

Brodell, Charles L.

1999-01-01

The Space Experiment Module (SEM) Program is an education initiative sponsored by the National Aeronautics and Space Administration (NASA) Shuttle Small Payloads Project. The program provides nationwide educational access to space for Kindergarten through University level students. The SEM program focuses on the science of zero-gravity and microgravity. Within the program, NASA provides small containers or "modules" for students to fly experiments on the Space Shuttle. The experiments are created, designed, built, and implemented by students with teacher and/or mentor guidance. Student experiment modules are flown in a "carrier" which resides in the cargo bay of the Space Shuttle. The carrier supplies power to, and the means to control and collect data from each experiment.

Audio Description as a Pedagogical Tool

Directory of Open Access Journals (Sweden)

Georgina Kleege

2015-05-01

Full Text Available Audio description is the process of translating visual information into words for people who are blind or have low vision. Typically such description has focused on films, museum exhibitions, images and video on the internet, and live theater. Because it allows people with visual impairments to experience a variety of cultural and educational texts that would otherwise be inaccessible, audio description is a mandated aspect of disability inclusion, although it remains markedly underdeveloped and underutilized in our classrooms and in society in general. Along with increasing awareness of disability, audio description pushes students to practice close reading of visual material, deepen their analysis, and engage in critical discussions around the methodology, standards and values, language, and role of interpretation in a variety of academic disciplines. We outline a few pedagogical interventions that can be customized to different contexts to develop students' writing and critical thinking skills through guided description of visual material.

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...

Improvements of ModalMax High-Fidelity Piezoelectric Audio Device

Science.gov (United States)

Woodard, Stanley E.

2005-01-01

ModalMax audio speakers have been enhanced by innovative means of tailoring the vibration response of thin piezoelectric plates to produce a high-fidelity audio response. The ModalMax audio speakers are 1 mm in thickness. The device completely supplants the need to have a separate driver and speaker cone. ModalMax speakers can perform the same applications of cone speakers, but unlike cone speakers, ModalMax speakers can function in harsh environments such as high humidity or extreme wetness. New design features allow the speakers to be completely submersed in salt water, making them well suited for maritime applications. The sound produced from the ModalMax audio speakers has sound spatial resolution that is readily discernable for headset users.

Aurally Aided Visual Search Performance Comparing Virtual Audio Systems

DEFF Research Database (Denmark)

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

2014-01-01

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

Aurally Aided Visual Search Performance Comparing Virtual Audio Systems

DEFF Research Database (Denmark)

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

2014-01-01

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

Efficiency in audio processing : filter banks and transcoding

NARCIS (Netherlands)

Lee, Jun Wei

2007-01-01

Audio transcoding is the conversion of digital audio from one compressed form A to another compressed form B, where A and B have different compression properties, such as a different bit-rate, sampling frequency or compression method. This is typically achieved by decoding A to an intermediate

Decision-level fusion for audio-visual laughter detection

NARCIS (Netherlands)

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

2008-01-01

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

Decision-Level Fusion for Audio-Visual Laughter Detection

NARCIS (Netherlands)

Reuderink, B.; Poel, Mannes; Truong, Khiet Phuong; Poppe, Ronald Walter; Pantic, Maja; Popescu-Belis, Andrei; Stiefelhagen, Rainer

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

Content Discovery from Composite Audio : An unsupervised approach

NARCIS (Netherlands)

Lu, L.

2009-01-01

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

Audio-tactile integration and the influence of musical training.

Science.gov (United States)

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

2014-01-01

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

Audio-tactile integration and the influence of musical training.

Directory of Open Access Journals (Sweden)

Anja Kuchenbuch

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

Paper-Based Textbooks with Audio Support for Print-Disabled Students.

Science.gov (United States)

Fujiyoshi, Akio; Ohsawa, Akiko; Takaira, Takuya; Tani, Yoshiaki; Fujiyoshi, Mamoru; Ota, Yuko

2015-01-01

Utilizing invisible 2-dimensional codes and digital audio players with a 2-dimensional code scanner, we developed paper-based textbooks with audio support for students with print disabilities, called "multimodal textbooks." Multimodal textbooks can be read with the combination of the two modes: "reading printed text" and "listening to the speech of the text from a digital audio player with a 2-dimensional code scanner." Since multimodal textbooks look the same as regular textbooks and the price of a digital audio player is reasonable (about 30 euro), we think multimodal textbooks are suitable for students with print disabilities in ordinary classrooms.

Huffman coding in advanced audio coding standard

Science.gov (United States)

Brzuchalski, Grzegorz

2012-05-01

This article presents several hardware architectures of Advanced Audio Coding (AAC) Huffman noiseless encoder, its optimisations and working implementation. Much attention has been paid to optimise the demand of hardware resources especially memory size. The aim of design was to get as short binary stream as possible in this standard. The Huffman encoder with whole audio-video system has been implemented in FPGA devices.

Sounding ruins: reflections on the production of an ‘audio drift’

Science.gov (United States)

Gallagher, Michael

2014-01-01

This article is about the use of audio media in researching places, which I term ‘audio geography’. The article narrates some episodes from the production of an ‘audio drift’, an experimental environmental sound work designed to be listened to on a portable MP3 player whilst walking in a ruinous landscape. Reflecting on how this work functions, I argue that, as well as representing places, audio geography can shape listeners’ attention and bodily movements, thereby reworking places, albeit temporarily. I suggest that audio geography is particularly apt for amplifying the haunted and uncanny qualities of places. I discuss some of the issues raised for research ethics, epistemology and spectral geographies. PMID:29708107

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...

Photonics-based microwave frequency measurement using a double-sideband suppressed-carrier modulation and an InP integrated ring-assisted Mach-Zehnder interferometer filter.

Science.gov (United States)

Fandiño, Javier S; Muñoz, Pascual

2013-11-01

A photonic system capable of estimating the unknown frequency of a CW microwave tone is presented. The core of the system is a complementary optical filter monolithically integrated in InP, consisting of a ring-assisted Mach-Zehnder interferometer with a second-order elliptic response. By simultaneously measuring the different optical powers produced by a double-sideband suppressed-carrier modulation at the outputs of the photonic integrated circuit, an amplitude comparison function that depends on the input tone frequency is obtained. Using this technique, a frequency measurement range of 10 GHz (5-15 GHz) with a root mean square value of frequency error lower than 200 MHz is experimentally demonstrated. Moreover, simulations showing the impact of a residual optical carrier on system performance are also provided.

Personalized Audio Systems - a Bayesian Approach

DEFF Research Database (Denmark)

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

2013-01-01

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

IELTS speaking instruction through audio/voice conferencing

Directory of Open Access Journals (Sweden)

Hamed Ghaemi

2012-02-01

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

A high efficiency PWM CMOS class-D audio power amplifier

Energy Technology Data Exchange (ETDEWEB)

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

2009-02-15

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

A high efficiency PWM CMOS class-D audio power amplifier

International Nuclear Information System (INIS)

Zhu Zhangming; Liu Lianxi; Yang Yintang; Lei Han

2009-01-01

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

A high efficiency PWM CMOS class-D audio power amplifier

Science.gov (United States)

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

2009-02-01

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

Audio-Visual Fusion for Sound Source Localization and Improved Attention

Energy Technology Data Exchange (ETDEWEB)

Lee, Byoung Gi; Choi, Jong Suk; Yoon, Sang Suk; Choi, Mun Taek; Kim, Mun Sang [Korea Institute of Science and Technology, Daejeon (Korea, Republic of); Kim, Dai Jin [Pohang University of Science and Technology, Pohang (Korea, Republic of)

2011-07-15

Service robots are equipped with various sensors such as vision camera, sonar sensor, laser scanner, and microphones. Although these sensors have their own functions, some of them can be made to work together and perform more complicated functions. AudioFvisual fusion is a typical and powerful combination of audio and video sensors, because audio information is complementary to visual information and vice versa. Human beings also mainly depend on visual and auditory information in their daily life. In this paper, we conduct two studies using audioFvision fusion: one is on enhancing the performance of sound localization, and the other is on improving robot attention through sound localization and face detection.

Audio-Visual Fusion for Sound Source Localization and Improved Attention

International Nuclear Information System (INIS)

Lee, Byoung Gi; Choi, Jong Suk; Yoon, Sang Suk; Choi, Mun Taek; Kim, Mun Sang; Kim, Dai Jin

2011-01-01

Service robots are equipped with various sensors such as vision camera, sonar sensor, laser scanner, and microphones. Although these sensors have their own functions, some of them can be made to work together and perform more complicated functions. AudioFvisual fusion is a typical and powerful combination of audio and video sensors, because audio information is complementary to visual information and vice versa. Human beings also mainly depend on visual and auditory information in their daily life. In this paper, we conduct two studies using audioFvision fusion: one is on enhancing the performance of sound localization, and the other is on improving robot attention through sound localization and face detection

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

DEFF Research Database (Denmark)

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

2016-01-01

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

A review of lossless audio compression standards and algorithms

Science.gov (United States)

Muin, Fathiah Abdul; Gunawan, Teddy Surya; Kartiwi, Mira; Elsheikh, Elsheikh M. A.

2017-09-01

Over the years, lossless audio compression has gained popularity as researchers and businesses has become more aware of the need for better quality and higher storage demand. This paper will analyse various lossless audio coding algorithm and standards that are used and available in the market focusing on Linear Predictive Coding (LPC) specifically due to its popularity and robustness in audio compression, nevertheless other prediction methods are compared to verify this. Advanced representation of LPC such as LSP decomposition techniques are also discussed within this paper.

WebGL and web audio software lightweight components for multimedia education

Science.gov (United States)

Chang, Xin; Yuksel, Kivanc; Skarbek, Władysław

2017-08-01

The paper presents the results of our recent work on development of contemporary computing platform DC2 for multimedia education usingWebGL andWeb Audio { the W3C standards. Using literate programming paradigm the WEBSA educational tools were developed. It offers for a user (student), the access to expandable collection of WEBGL Shaders and web Audio scripts. The unique feature of DC2 is the option of literate programming, offered for both, the author and the reader in order to improve interactivity to lightweightWebGL andWeb Audio components. For instance users can define: source audio nodes including synthetic sources, destination audio nodes, and nodes for audio processing such as: sound wave shaping, spectral band filtering, convolution based modification, etc. In case of WebGL beside of classic graphics effects based on mesh and fractal definitions, the novel image processing analysis by shaders is offered like nonlinear filtering, histogram of gradients, and Bayesian classifiers.

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.

47 CFR 80.207 - Classes of emission.

Science.gov (United States)

2010-10-01

... communications employing a modulated carrier the carrier must be keyed and modulated by an audio frequency. (c..., 16, 17 A1A, J2A, J2B, J2D. DSC 6 F1B, J2B. NB-DP 14, 16 F1B, J2B, J2D. Facsimile F1C, F3C, J2C, J3C. 156-162 MHz 2 F1B, F2B, F2C, F3C, F1D, F2D. DSC G2B. 216-220 MHz 3 F1B, F2B, F2C, F3C. 1626.5-1646.5...

Class D audio amplifiers for high voltage capacitive transducers

DEFF Research Database (Denmark)

Nielsen, Dennis

of high volume, weight, and cost. High efficient class D amplifiers are now widely available offering power densities, that their linear counterparts can not match. Unlike the technology of audio amplifiers, the loudspeaker is still based on the traditional electrodynamic transducer invented by C.W. Rice......Audio reproduction systems contains two key components, the amplifier and the loudspeaker. In the last 20 – 30 years the technology of audio amplifiers have performed a fundamental shift of paradigm. Class D audio amplifiers have replaced the linear amplifiers, suffering from the well-known issues...... with the low level of acoustical output power and complex amplifier requirements, have limited the commercial success of the technology. Horn or compression drivers are typically favoured, when high acoustic output power is required, this is however at the expense of significant distortion combined...

Can audio recording improve patients' recall of outpatient consultations?

DEFF Research Database (Denmark)

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

Introduction In order to give patients possibility to listen to their consultation again, we have designed a system which gives the patients access to digital audio recordings of their consultations. An Interactive Voice Response platform enables the audio recording and gives the patients access...... and those who have not (control).The audio recordings and the interviews are coded according to six themes: Test results, Treatment, Risks, Future tests, Advice and Plan. Afterwards the extent of patients recall is assessed by comparing the accuracy of the patient’s statements (interview...

Isolated PWM DC-AC SICAM with an active capacitive voltage clamp[Pulse Density Modulated; Pulse Width Modulation

Energy Technology Data Exchange (ETDEWEB)

Ljusev, P.

2004-03-15

In this report an isolated PWM DC-AC SICAM with an active capacitive voltage clamp is presented. AC-DC power supply is implemented in its simplest form: diode rectifier followed by a medium-size charge-storage capacitors and possibly with an EMC filter on the mains entrance. Isolation from the AC mains is achieved using a high frequency (HF) transformer, whose voltages are not audio-modulated. The latter simplifies the design and is expected to have many advantages over the approach where the transformer voltages are modulated in regards to the audio signal reference. Input stage is built as a DC-AC inverter (push-pull, half-bridge or a full-bridge) and operated with 50% duty cycle, with all the challenges to avoid transformer saturation and obtain symmetrical operation. On the secondary side the output section is implemented as rectifier+inverter AC-AC stage, i.e. a true bidirectional bridge, which operation is aimed towards amplification of the audio signal. In order to solve the problem with the commutation of the load current, a dead time between the incoming and outgoing bidirectional switch is implemented, while a capacitive voltage clamp is used to keep the induced overvoltage to reasonable levels. The energy stored in the clamping capacitor is not wasted as in the dissipative clamps, but is rather transferred back to the primary side for further processing using an auxiliary isolated single-switch converter, i.e. an active clamping technique is used. (au)

Music information retrieval in compressed audio files: a survey

Science.gov (United States)

Zampoglou, Markos; Malamos, Athanasios G.

2014-07-01

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

StirMark Benchmark: audio watermarking attacks based on lossy compression

Science.gov (United States)

Steinebach, Martin; Lang, Andreas; Dittmann, Jana

2002-04-01

StirMark Benchmark is a well-known evaluation tool for watermarking robustness. Additional attacks are added to it continuously. To enable application based evaluation, in our paper we address attacks against audio watermarks based on lossy audio compression algorithms to be included in the test environment. We discuss the effect of different lossy compression algorithms like MPEG-2 audio Layer 3, Ogg or VQF on a selection of audio test data. Our focus is on changes regarding the basic characteristics of the audio data like spectrum or average power and on removal of embedded watermarks. Furthermore we compare results of different watermarking algorithms and show that lossy compression is still a challenge for most of them. There are two strategies for adding evaluation of robustness against lossy compression to StirMark Benchmark: (a) use of existing free compression algorithms (b) implementation of a generic lossy compression simulation. We discuss how such a model can be implemented based on the results of our tests. This method is less complex, as no real psycho acoustic model has to be applied. Our model can be used for audio watermarking evaluation of numerous application fields. As an example, we describe its importance for e-commerce applications with watermarking security.

AKTIVITAS SEKUNDER AUDIO UNTUK MENJAGA KEWASPADAAN PENGEMUDI MOBIL INDONESIA

Directory of Open Access Journals (Sweden)

Iftikar Zahedi Sutalaksana

2013-03-01

Full Text Available Tingkat kecelakaan lalu lintas yang melibatkan mobil di Indonesia semakin mengkhawatirkan. Tingginya peran faktor manusia sebagai penyebab utama kejadian kecelakaan patut diperhatikan. Penurunan kewaspadaan saat mengemudi akibat kantuk atau kelelahan merupakan salah satu kondisi yang mendorong terjadinya kecelakaan. Tulisan ini memaparkan aplikasi audio response test sebagai aktivitas sekunder dalam mengemudikan mobil. Response test yang dimaksud merupakan seperangkat aplikasi pada dashboard mobil yang menuntut respon pengemudi setiap stimulus suara bekerja. Audio response test ini diusulkan sebagai pemantau tingkat kewaspadaan pengemudi selama berkendara. Kewaspadaan pengemudi merupakan kondisi selama berkendara yang terjaga, awas, dan mampu memproses semua stimulus dengan baik. Hasil studi ini menghasilkan suatu bentuk audio response test yang terintegrasi dengan sistem berkendara di dalam mobil. Sumber bunyi diperdengarkan dengan intensitas konstan antara 80-85 dB. Bunyi akan berhenti jika pengemudi memberikan respon atas stimulus suara tersebut. Response test ini dirancang untuk mampu memantau tingkat kewaspadaan pengemudi selama berkendara. Penerapannya diharapkan mampu membantu menekan tingkat kecelakaan lalu lintas di Indonesia. Kata kunci: mengemudi, aktivitas sekunder, audio, kewaspadaan, response test   Abstract   The level of traffic accidents involving cars in Indonesia increasingly alarming. The high role of the human factor as the main cause of accident noteworthy. Decreased alertness while driving due to sleepiness or fatigue is one of the conditions that led to the accident. This paper describes an audio application response test as a secondary activity of driving a car. Response test is a set of applications on the dashboard of a car that demands a response driver each stimulus voice work. Audio response was proposed as test monitors the driver's level of alertness while driving. Vigilance driver was driving conditions during

Audio Networking in the Music Industry

Directory of Open Access Journals (Sweden)

Glebs Kuzmics

2018-01-01

Full Text Available This paper surveys the rôle of computer networking technologies in the music industry. A comparison of their relevant technologies, their defining advantages and disadvantages; analyses and discussion of the situation in the market of network enabled audio products followed by a discussion of different devices are presented. The idea of replacing a proprietary solution with open-source and freeware software programs has been chosen as the fundamental concept of this research. The technologies covered include: native IEEE AVnu Alliance Audio Video Bridging (AVB, CobraNet®, Audinate Dante™ and Harman BLU Link.

Frequency Hopping Method for Audio Watermarking

Directory of Open Access Journals (Sweden)

A. Anastasijević

2012-11-01

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

Four-quadrant flyback converter for direct audio power amplification

DEFF Research Database (Denmark)

Ljusev, Petar; Andersen, Michael Andreas E.

2005-01-01

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

Sigma-delta modulator modeling analysis and design

International Nuclear Information System (INIS)

Ge Binjie; Wang Xin'an; Zhang Xing; Feng Xiaoxing; Wang Qingqin

2010-01-01

This paper introduces a new method for SC sigma-delta modulator modeling. It studies the integrator's different equivalent circuits in the integrating and sampling phases. This model uses the OP-AMP input pair's tail current (I 0 ) and overdrive voltage (v on ) as variables. The modulator's static and dynamic errors are analyzed. A group of optimized I 0 and v on for maximum SNR and power x area ratio can be obtained through this model. As examples, a MASH21 modulator for digital audio and a second order modulator for RFID baseband are implemented and tested, and they can achieve 91 dB and 72 dB respectively, which verifies the modeling and design criteria. (semiconductor integrated circuits)

Audio Technology and Mobile Human Computer Interaction

DEFF Research Database (Denmark)

Chamberlain, Alan; Bødker, Mads; Hazzard, Adrian

2017-01-01

Audio-based mobile technology is opening up a range of new interactive possibilities. This paper brings some of those possibilities to light by offering a range of perspectives based in this area. It is not only the technical systems that are developing, but novel approaches to the design...... and understanding of audio-based mobile systems are evolving to offer new perspectives on interaction and design and support such systems to be applied in areas, such as the humanities....

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

Directory of Open Access Journals (Sweden)

Maxime eAmbard

2015-10-01

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

Spatial Terahertz Modulator

Science.gov (United States)

Xie, Zhenwei; Wang, Xinke; Ye, Jiasheng; Feng, Shengfei; Sun, Wenfeng; Akalin, Tahsin; Zhang, Yan

2013-11-01

Terahertz (THz) technology is a developing and promising candidate for biological imaging, security inspection and communications, due to the low photon energy, the high transparency and the broad band properties of the THz radiation. However, a major encountered bottleneck is lack of efficient devices to manipulate the THz wave, especially to modulate the THz wave front. A wave front modulator should allow the optical or electrical control of the spatial transmission (or reflection) of an input THz wave and hence the ability to encode the information in a wave front. Here we propose a spatial THz modulator (STM) to dynamically control the THz wave front with photo-generated carriers. A computer generated THz hologram is projected onto a silicon wafer by a conventional spatial light modulator (SLM). The corresponding photo-generated carrier spatial distribution will be induced, which forms an amplitude hologram to modulate the wave front of the input THz beam. Some special intensity patterns and vortex beams are generated by using this method. This all-optical controllable STM is structure free, high resolution and broadband. It is expected to be widely used in future THz imaging and communication systems.

Unsupervised topic modelling on South African parliament audio data

CSIR Research Space (South Africa)

Kleynhans, N

2014-11-01

Full Text Available Using a speech recognition system to convert spoken audio to text can enable the structuring of large collections of spoken audio data. A convenient means to summarise or cluster spoken data is to identify the topic under discussion. There are many...

Classifying laughter and speech using audio-visual feature prediction

NARCIS (Netherlands)

Petridis, Stavros; Asghar, Ali; Pantic, Maja

2010-01-01

In this study, a system that discriminates laughter from speech by modelling the relationship between audio and visual features is presented. The underlying assumption is that this relationship is different between speech and laughter. Neural networks are trained which learn the audio-to-visual and

Sistema de adquisición y procesamiento de audio

OpenAIRE

Pérez Segurado, Rubén

2015-01-01

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

Analytical Features: A Knowledge-Based Approach to Audio Feature Generation

Directory of Open Access Journals (Sweden)

Pachet François

2009-01-01

Full Text Available We present a feature generation system designed to create audio features for supervised classification tasks. The main contribution to feature generation studies is the notion of analytical features (AFs, a construct designed to support the representation of knowledge about audio signal processing. We describe the most important aspects of AFs, in particular their dimensional type system, on which are based pattern-based random generators, heuristics, and rewriting rules. We show how AFs generalize or improve previous approaches used in feature generation. We report on several projects using AFs for difficult audio classification tasks, demonstrating their advantage over standard audio features. More generally, we propose analytical features as a paradigm to bring raw signals into the world of symbolic computation.

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.

A novel five-level optimized carrier multilevel PWM quad-inverter six-phase AC drive

DEFF Research Database (Denmark)

Sanjeevikumar, P.; Blaabjerg, F.; Wheeler, Pat

2016-01-01

A novel single carrier pulse-width modulation (PWM) for a new quad-inverter configuration for multilevel six-phase asymmetrical open-winding ac converter is proposed in this article. Modularity of the circuit consist of four standard two-level voltage source inverters (VSIs) with slight modificat......A novel single carrier pulse-width modulation (PWM) for a new quad-inverter configuration for multilevel six-phase asymmetrical open-winding ac converter is proposed in this article. Modularity of the circuit consist of four standard two-level voltage source inverters (VSIs) with slight...... modifications, i.e. one additional bi-direction switch (MOSFET/IGBT) in each phase and a link to neutral with two capacitors to generate increased output levels. Furthermore, original optimal single carrier zero-shifted five-level modulation (SCZSFM) algorithm is developed for each VSI to behave as equivalent...

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...

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.

Design of batch audio/video conversion platform based on JavaEE

Science.gov (United States)

Cui, Yansong; Jiang, Lianpin

2018-03-01

With the rapid development of digital publishing industry, the direction of audio / video publishing shows the diversity of coding standards for audio and video files, massive data and other significant features. Faced with massive and diverse data, how to quickly and efficiently convert to a unified code format has brought great difficulties to the digital publishing organization. In view of this demand and present situation in this paper, basing on the development architecture of Sptring+SpringMVC+Mybatis, and combined with the open source FFMPEG format conversion tool, a distributed online audio and video format conversion platform with a B/S structure is proposed. Based on the Java language, the key technologies and strategies designed in the design of platform architecture are analyzed emphatically in this paper, designing and developing a efficient audio and video format conversion system, which is composed of “Front display system”, "core scheduling server " and " conversion server ". The test results show that, compared with the ordinary audio and video conversion scheme, the use of batch audio and video format conversion platform can effectively improve the conversion efficiency of audio and video files, and reduce the complexity of the work. Practice has proved that the key technology discussed in this paper can be applied in the field of large batch file processing, and has certain practical application value.

Class-D audio amplifiers with negative feedback

OpenAIRE

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

2006-01-01

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

Documentary management of the sport audio-visual information in the generalist televisions

OpenAIRE

Jorge Caldera Serrano; Felipe Alonso

2007-01-01

The management of the sport audio-visual documentation of the Information Systems of the state, zonal and local chains is analyzed within the framework. For it it is made makes a route by the documentary chain that makes the sport audio-visual information with the purpose of being analyzing each one of the parameters, showing therefore a series of recommendations and norms for the preparation of the sport audio-visual registry. Evidently the audio-visual sport documentation difference i...

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...

A Psychoacoustic-Based Multiple Audio Object Coding Approach via Intra-Object Sparsity

Directory of Open Access Journals (Sweden)

Maoshen Jia

2017-12-01

Full Text Available Rendering spatial sound scenes via audio objects has become popular in recent years, since it can provide more flexibility for different auditory scenarios, such as 3D movies, spatial audio communication and virtual classrooms. To facilitate high-quality bitrate-efficient distribution for spatial audio objects, an encoding scheme based on intra-object sparsity (approximate k-sparsity of the audio object itself is proposed in this paper. The statistical analysis is presented to validate the notion that the audio object has a stronger sparseness in the Modified Discrete Cosine Transform (MDCT domain than in the Short Time Fourier Transform (STFT domain. By exploiting intra-object sparsity in the MDCT domain, multiple simultaneously occurring audio objects are compressed into a mono downmix signal with side information. To ensure a balanced perception quality of audio objects, a Psychoacoustic-based time-frequency instants sorting algorithm and an energy equalized Number of Preserved Time-Frequency Bins (NPTF allocation strategy are proposed, which are employed in the underlying compression framework. The downmix signal can be further encoded via Scalar Quantized Vector Huffman Coding (SQVH technique at a desirable bitrate, and the side information is transmitted in a lossless manner. Both objective and subjective evaluations show that the proposed encoding scheme outperforms the Sparsity Analysis (SPA approach and Spatial Audio Object Coding (SAOC in cases where eight objects were jointly encoded.

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"--

BAT: An open-source, web-based audio events annotation tool

OpenAIRE

Blai Meléndez-Catalan, Emilio Molina, Emilia Gómez

2017-01-01

In this paper we present BAT (BMAT Annotation Tool), an open-source, web-based tool for the manual annotation of events in audio recordings developed at BMAT (Barcelona Music and Audio Technologies). The main feature of the tool is that it provides an easy way to annotate the salience of simultaneous sound sources. Additionally, it allows to define multiple ontologies to adapt to multiple tasks and offers the possibility to cross-annotate audio data. Moreover, it is easy to install and deploy...

On the Use of Memory Models in Audio Features

DEFF Research Database (Denmark)

Jensen, Karl Kristoffer

2011-01-01

Audio feature estimation is potentially improved by including higher- level models. One such model is the Short Term Memory (STM) model. A new paradigm of audio feature estimation is obtained by adding the influence of notes in the STM. These notes are identified when the perceptual spectral flux...

Audio Teleconferencing: Low Cost Technology for External Studies Networking.

Science.gov (United States)

Robertson, Bill

1987-01-01

This discussion of the benefits of audio teleconferencing for distance education programs and for business and government applications focuses on the recent experience of Canadian educational users. Four successful operating models and their costs are reviewed, and it is concluded that audio teleconferencing is cost efficient and educationally…

Automatic Organisation and Quality Analysis of User-Generated Content with Audio Fingerprinting

OpenAIRE

Cavaco, Sofia; Magalhaes, Joao; Mordido, Gonçalo

2018-01-01

The increase of the quantity of user-generated content experienced in social media has boosted the importance of analysing and organising the content by its quality. Here, we propose a method that uses audio fingerprinting to organise and infer the quality of user-generated audio content. The proposed method detects the overlapping segments between different audio clips to organise and cluster the data according to events, and to infer the audio quality of the samples. A test setup with conce...

Real-Time Audio Processing on the T-CREST Multicore Platform

DEFF Research Database (Denmark)

Ausin, Daniel Sanz; Pezzarossa, Luca; Schoeberl, Martin

2017-01-01

of the audio signal. This paper presents a real-time multicore audio processing system based on the T-CREST platform. T-CREST is a time-predictable multicore processor for real-time embedded systems. Multiple audio effect tasks have been implemented, which can be connected together in different configurations...... forming sequential and parallel effect chains, and using a network-onchip for intercommunication between processors. The evaluation of the system shows that real-time processing of multiple effect configurations is possible, and that the estimation and control of latency ensures real-time behavior.......Multicore platforms are nowadays widely used for audio processing applications, due to the improvement of computational power that they provide. However, some of these systems are not optimized for temporally constrained environments, which often leads to an undesired increase in the latency...

Haptic and Audio-visual Stimuli: Enhancing Experiences and Interaction

NARCIS (Netherlands)

Nijholt, Antinus; Dijk, Esko O.; Lemmens, Paul M.C.; Luitjens, S.B.

2010-01-01

The intention of the symposium on Haptic and Audio-visual stimuli at the EuroHaptics 2010 conference is to deepen the understanding of the effect of combined Haptic and Audio-visual stimuli. The knowledge gained will be used to enhance experiences and interactions in daily life. To this end, a

The Effect of Audio and Animation in Multimedia Instruction

Science.gov (United States)

Koroghlanian, Carol; Klein, James D.

2004-01-01

This study investigated the effects of audio, animation, and spatial ability in a multimedia computer program for high school biology. Participants completed a multimedia program that presented content by way of text or audio with lean text. In addition, several instructional sequences were presented either with static illustrations or animations.…

Selected Audio-Visual Materials for Consumer Education. [New Version.

Science.gov (United States)

Johnston, William L.

Ninety-two films, filmstrips, multi-media kits, slides, and audio cassettes, produced between 1964 and 1974, are listed in this selective annotated bibliography on consumer education. The major portion of the bibliography is devoted to films and filmstrips. The main topics of the audio-visual materials include purchasing, advertising, money…

Audio-visual temporal recalibration can be constrained by content cues regardless of spatial overlap

Directory of Open Access Journals (Sweden)

Warrick eRoseboom

2013-04-01

Full Text Available It has now been well established that the point of subjective synchrony for audio and visual events can be shifted following exposure to asynchronous audio-visual presentations, an effect often referred to as temporal recalibration. Recently it was further demonstrated that it is possible to concurrently maintain two such recalibrated, and opposing, estimates of audio-visual temporal synchrony. However, it remains unclear precisely what defines a given audio-visual pair such that it is possible to maintain a temporal relationship distinct from other pairs. It has been suggested that spatial separation of the different audio-visual pairs is necessary to achieve multiple distinct audio-visual synchrony estimates. Here we investigated if this was necessarily true. Specifically, we examined whether it is possible to obtain two distinct temporal recalibrations for stimuli that differed only in featural content. Using both complex (audio visual speech; Experiment 1 and simple stimuli (high and low pitch audio matched with either vertically or horizontally oriented Gabors; Experiment 2 we found concurrent, and opposite, recalibrations despite there being no spatial difference in presentation location at any point throughout the experiment. This result supports the notion that the content of an audio-visual pair can be used to constrain distinct audio-visual synchrony estimates regardless of spatial overlap.

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

International Nuclear Information System (INIS)

Nakamura, Mitsuhiro; Narita, Yuichiro; Matsuo, Yukinori; Narabayashi, Masaru; Nakata, Manabu; Sawada, Akira; Mizowaki, Takashi; Nagata, Yasushi; Hiraoka, Masahiro

2009-01-01

Purpose: To assess the effect of audio coaching on the time-dependent behavior of the correlation between abdominal motion and lung tumor motion and the corresponding lung tumor position mismatches. Methods and Materials: Six patients who had a lung tumor with a motion range >8 mm were enrolled in the present study. Breathing-synchronized fluoroscopy was performed initially without audio coaching, followed by fluoroscopy with recorded audio coaching for multiple days. Two different measurements, anteroposterior abdominal displacement using the real-time positioning management system and superoinferior (SI) lung tumor motion by X-ray fluoroscopy, were performed simultaneously. Their sequential images were recorded using one display system. The lung tumor position was automatically detected with a template matching technique. The relationship between the abdominal and lung tumor motion was analyzed with and without audio coaching. Results: The mean SI tumor displacement was 10.4 mm without audio coaching and increased to 23.0 mm with audio coaching (p < .01). The correlation coefficients ranged from 0.89 to 0.97 with free breathing. Applying audio coaching, the correlation coefficients improved significantly (range, 0.93-0.99; p < .01), and the SI lung tumor position mismatches became larger in 75% of all sessions. Conclusion: Audio coaching served to increase the degree of correlation and make it more reproducible. In addition, the phase shifts between tumor motion and abdominal displacement were improved; however, all patients breathed more deeply, and the SI lung tumor position mismatches became slightly larger with audio coaching than without audio coaching.

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

Directory of Open Access Journals (Sweden)

Bekti Wulandari

2016-01-01

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

Coherent Detection of Wavelength Division Multiplexed Phase-Modulated Radio-over-Fibre Signals

DEFF Research Database (Denmark)

Zibar, Darko; Yu, Xianbin; Peucheret, Christophe

2008-01-01

A WDM phase-modulated Radio-over-Fibre link using digital coherent detection is experimentally demonstrated. 3 times 50 Mb/s WDM transmission of a BPSK modulated 5 GHz RF carrier is achieved over 25 km.......A WDM phase-modulated Radio-over-Fibre link using digital coherent detection is experimentally demonstrated. 3 times 50 Mb/s WDM transmission of a BPSK modulated 5 GHz RF carrier is achieved over 25 km....

Four-quadrant flyback converter for direct audio power amplification

OpenAIRE

Ljusev, Petar; Andersen, Michael Andreas E.

2005-01-01

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

Semantic Analysis of Multimedial Information Usign Both Audio and Visual Clues

Directory of Open Access Journals (Sweden)

Andrej Lukac

2008-01-01

Full Text Available Nowadays, there is a lot of information in databases (text, audio/video form, etc.. It is important to be able to describe this data for better orientation in them. It is necessary to apply audio/video properties, which are used for metadata management, segmenting the document into semantically meaningful units, classifying each unit into a predefined scene type, indexing, summarizing the document for efficient retrieval and browsing. Data can be used for system that automatically searches for a specific person in a sequence also for special video sequences. Audio/video properties are presented by descriptors and description schemes. There are many features that can be used to characterize multimedial signals. We can analyze audio and video sequences jointly or considered them completely separately. Our aim is oriented to possibilities of combining multimedial features. Focus is direct into discussion programs, because there are more decisions how to combine audio features with video sequences.

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

DEFF Research Database (Denmark)

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

2015-01-01

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

Balancing Audio

DEFF Research Database (Denmark)

Walther-Hansen, Mads

2016-01-01

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

Four-quadrant flyback converter for direct audio power amplification

Energy Technology Data Exchange (ETDEWEB)

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

2005-07-01

This paper presents a bidirectional, four-quadrant yback converter for use in direct audio power amplication. When compared to the standard Class-D switching-mode audio power amplier with separate power supply, the proposed four-quadrant flyback converter provides simple and compact solution with high efciency, higher level of integration, lower component count, less board space and eventually lower cost. Both peak and average current-mode control for use with 4Q flyback power converters are described and compared. Integrated magnetics is presented which simplies the construction of the auxiliary power supplies for control biasing and isolated gate drives. The feasibility of the approach is proven on audio power amplier prototype for subwoofer applications. (au)

Animation, audio, and spatial ability: Optimizing multimedia for scientific explanations

Science.gov (United States)

Koroghlanian, Carol May

This study investigated the effects of audio, animation and spatial ability in a computer based instructional program for biology. The program presented instructional material via text or audio with lean text and included eight instructional sequences presented either via static illustrations or animations. High school students enrolled in a biology course were blocked by spatial ability and randomly assigned to one of four treatments (Text-Static Illustration Audio-Static Illustration, Text-Animation, Audio-Animation). The study examined the effects of instructional mode (Text vs. Audio), illustration mode (Static Illustration vs. Animation) and spatial ability (Low vs. High) on practice and posttest achievement, attitude and time. Results for practice achievement indicated that high spatial ability participants achieved more than low spatial ability participants. Similar results for posttest achievement and spatial ability were not found. Participants in the Static Illustration treatments achieved the same as participants in the Animation treatments on both the practice and posttest. Likewise, participants in the Text treatments achieved the same as participants in the Audio treatments on both the practice and posttest. In terms of attitude, participants responded favorably to the computer based instructional program. They found the program interesting, felt the static illustrations or animations made the explanations easier to understand and concentrated on learning the material. Furthermore, participants in the Animation treatments felt the information was easier to understand than participants in the Static Illustration treatments. However, no difference for any attitude item was found for participants in the Text as compared to those in the Audio treatments. Significant differences were found by Spatial Ability for three attitude items concerning concentration and interest. In all three items, the low spatial ability participants responded more positively

Femtosecond time-resolved hot carrier energy distributions of photoexcited semiconductor quantum dots

International Nuclear Information System (INIS)

Chuang, Chi-Hung; Burda, Clemens; Chen, Xiaobo

2013-01-01

Using femtosecond transient absorption spectroscopy, we investigated hot carrier distributions in semiconductor cadmium selenide quantum dots. The relaxation processes represent the behavior of an ensemble of QDs. This concept is applied for analysis with the Fermi-Dirac distribution and relaxation processes among different electron-hole pair states. By extracting the experimental hot carrier distribution and fitting with the Fermi-Dirac function, we resolved the rapid thermalization processes, such as carrier-carrier and carrier-phonon interactions was resolved within one picosecond upon photoexcitation. The analysis, using the Fermi-Dirac distribution modulated by the density of states, provides a general route to understanding the carrier cooling and heat dissipation processes in quantum dot-based systems. (copyright 2012 by WILEY-VCH Verlag GmbH and Co. KGaA, Weinheim)

Parametric Audio Based Decoder and Music Synthesizer for Mobile Applications

NARCIS (Netherlands)

Oomen, A.W.J.; Szczerba, M.Z.; Therssen, D.

2011-01-01

This paper reviews parametric audio coders and discusses novel technologies introduced in a low-complexity, low-power consumption audiodecoder and music synthesizer platform developed by the authors. Thedecoder uses parametric coding scheme based on the MPEG-4 Parametric Audio standard. In order to

Automated Speech and Audio Analysis for Semantic Access to Multimedia

NARCIS (Netherlands)

Jong, F.M.G. de; Ordelman, R.; Huijbregts, M.

2006-01-01

The deployment and integration of audio processing tools can enhance the semantic annotation of multimedia content, and as a consequence, improve the effectiveness of conceptual access tools. This paper overviews the various ways in which automatic speech and audio analysis can contribute to

Automated speech and audio analysis for semantic access to multimedia

NARCIS (Netherlands)

de Jong, Franciska M.G.; Ordelman, Roeland J.F.; Huijbregts, M.A.H.; Avrithis, Y.; Kompatsiaris, Y.; Staab, S.; O' Connor, N.E.

2006-01-01

The deployment and integration of audio processing tools can enhance the semantic annotation of multimedia content, and as a consequence, improve the effectiveness of conceptual access tools. This paper overviews the various ways in which automatic speech and audio analysis can contribute to

Audio Feedback -- Better Feedback?

Science.gov (United States)

Voelkel, Susanne; Mello, Luciane V.

2014-01-01

National Student Survey (NSS) results show that many students are dissatisfied with the amount and quality of feedback they get for their work. This study reports on two case studies in which we tried to address these issues by introducing audio feedback to one undergraduate (UG) and one postgraduate (PG) class, respectively. In case study one…

Emotion-based Music Rretrieval on a Well-reduced Audio Feature Space

DEFF Research Database (Denmark)

Ruxanda, Maria Magdalena; Chua, Bee Yong; Nanopoulos, Alexandros

2009-01-01

-emotion. However, the real-time systems that retrieve music over large music databases, can achieve order of magnitude performance increase, if applying multidimensional indexing over a dimensionally reduced audio feature space. To meet this performance achievement, in this paper, extensive studies are conducted......Music expresses emotion. A number of audio extracted features have influence on the perceived emotional expression of music. These audio features generate a high-dimensional space, on which music similarity retrieval can be performed effectively, with respect to human perception of the music...... on a number of dimensionality reduction algorithms, including both classic and novel approaches. The paper clearly envisages which dimensionality reduction techniques on the considered audio feature space, can preserve in average the accuracy of the emotion-based music retrieval....

News video story segmentation method using fusion of audio-visual features

Science.gov (United States)

Wen, Jun; Wu, Ling-da; Zeng, Pu; Luan, Xi-dao; Xie, Yu-xiang

2007-11-01

News story segmentation is an important aspect for news video analysis. This paper presents a method for news video story segmentation. Different form prior works, which base on visual features transform, the proposed technique uses audio features as baseline and fuses visual features with it to refine the results. At first, it selects silence clips as audio features candidate points, and selects shot boundaries and anchor shots as two kinds of visual features candidate points. Then this paper selects audio feature candidates as cues and develops different fusion method, which effectively using diverse type visual candidates to refine audio candidates, to get story boundaries. Experiment results show that this method has high efficiency and adaptability to different kinds of news video.

Improving audio chord transcription by exploiting harmonic and metric knowledge

NARCIS (Netherlands)

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

2012-01-01

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

Comprehension and Motivation Levels in Conjunction with the Use of eBooks with Audio: A Quasi-Experimental Study of Post-Secondary Remedial Reading Students

Science.gov (United States)

Wheeler, Kimberly W.

2014-01-01

This quasi-experimental pretest, posttest nonequivalent control group study investigated the comprehension scores and motivation levels of post-secondary remedial reading students in a two-year technical college in Northwest Georgia using an eBook, an eBook with audio, and a print book. After reading a module on Purpose and Tone in the three book…

Extraction, Mapping, and Evaluation of Expressive Acoustic Features for Adaptive Digital Audio Effects

DEFF Research Database (Denmark)

Holfelt, Jonas; Csapo, Gergely; Andersson, Nikolaj Schwab

2017-01-01

This paper describes the design and implementation of a real-time adaptive digital audio effect with an emphasis on using expressive audio features that control effect param- eters. Research in adaptive digital audio effects is cov- ered along with studies about expressivity and important...

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

Science.gov (United States)

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

2018-04-01

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

Linear combination of auditory steady-state responses evoked by co-modulated tones

DEFF Research Database (Denmark)

Guérit, François; Marozeau, Jeremy; Epp, Bastian

2017-01-01

Up to medium intensities and in the 80–100-Hz region, the auditory steady-state response (ASSR) to a multi-tone carrier is commonly considered to be a linear sum of the dipoles from each tone specific ASSR generator. Here, this hypothesis was investigated when a unique modulation frequency is used...... for all carrier components. Listeners were presented with a co-modulated dual-frequency carrier (1 and 4 kHz), from which the modulator starting phase Ui of the 1-kHz component was systematically varied. The results support the hypothesis of a linear superposition of the dipoles originating from different...

Let Their Voices Be Heard! Building a Multicultural Audio Collection.

Science.gov (United States)

Tucker, Judith Cook

1992-01-01

Discusses building a multicultural audio collection for a library. Gives some guidelines about selecting materials that really represent different cultures. Audio materials that are considered fall roughly into the categories of children's stories, didactic materials, oral histories, poetry and folktales, and music. The goal is an authentic…

Carrier phase synchronization system for improved amplitude modulation and television broadcast reception

Science.gov (United States)

Smith, Stephen F [Loudon, TN; Moore, James A [Powell, TN

2009-09-08

Systems and methods are described for carrier phase synchronization for improved AM and TV broadcast reception. A method includes synchronizing the phase of a carrier frequency of a broadcast signal with the phase of a remote reference frequency. An apparatus includes a receiver to detect the phase of a reference signal; a phase comparator coupled to the reference signal-phase receiver; a voltage controlled oscillator coupled to the phase comparator; and a phase-controlled radio frequency output coupled to the voltage controlled oscillator.

Sigma-delta modulator modeling analysis and design

Energy Technology Data Exchange (ETDEWEB)

Ge Binjie; Wang Xin' an; Zhang Xing; Feng Xiaoxing; Wang Qingqin, E-mail: wangxa@szpku.edu.c [Key Laboratory of Integrated Microsystem Science and Engineering Applications, Shenzhen Graduate School of Peking University, Shenzhen 518055 (China)

2010-09-15

This paper introduces a new method for SC sigma-delta modulator modeling. It studies the integrator's different equivalent circuits in the integrating and sampling phases. This model uses the OP-AMP input pair's tail current (I{sub 0}) and overdrive voltage (v{sub on}) as variables. The modulator's static and dynamic errors are analyzed. A group of optimized I{sub 0} and v{sub on} for maximum SNR and power x area ratio can be obtained through this model. As examples, a MASH21 modulator for digital audio and a second order modulator for RFID baseband are implemented and tested, and they can achieve 91 dB and 72 dB respectively, which verifies the modeling and design criteria. (semiconductor integrated circuits)

Multilevel inverter based class D audio amplifier for capacitive transducers

DEFF Research Database (Denmark)

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

2014-01-01

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

The Use of Audio and Animation in Computer Based Instruction.

Science.gov (United States)

Koroghlanian, Carol; Klein, James D.

This study investigated the effects of audio, animation, and spatial ability in a computer-based instructional program for biology. The program presented instructional material via test or audio with lean text and included eight instructional sequences presented either via static illustrations or animations. High school students enrolled in a…

Audio-Visual Speaker Diarization Based on Spatiotemporal Bayesian Fusion.

Science.gov (United States)

Gebru, Israel D; Ba, Sileye; Li, Xiaofei; Horaud, Radu

2018-05-01

Speaker diarization consists of assigning speech signals to people engaged in a dialogue. An audio-visual spatiotemporal diarization model is proposed. The model is well suited for challenging scenarios that consist of several participants engaged in multi-party interaction while they move around and turn their heads towards the other participants rather than facing the cameras and the microphones. Multiple-person visual tracking is combined with multiple speech-source localization in order to tackle the speech-to-person association problem. The latter is solved within a novel audio-visual fusion method on the following grounds: binaural spectral features are first extracted from a microphone pair, then a supervised audio-visual alignment technique maps these features onto an image, and finally a semi-supervised clustering method assigns binaural spectral features to visible persons. The main advantage of this method over previous work is that it processes in a principled way speech signals uttered simultaneously by multiple persons. The diarization itself is cast into a latent-variable temporal graphical model that infers speaker identities and speech turns, based on the output of an audio-visual association process, executed at each time slice, and on the dynamics of the diarization variable itself. The proposed formulation yields an efficient exact inference procedure. A novel dataset, that contains audio-visual training data as well as a number of scenarios involving several participants engaged in formal and informal dialogue, is introduced. The proposed method is thoroughly tested and benchmarked with respect to several state-of-the art diarization algorithms.

El Digital Audio Tape Recorder. Contra autores y creadores

Directory of Open Access Journals (Sweden)

Jun Ono

2015-01-01

Full Text Available La llamada "DAT" (abreviatura por "digital audio tape recorder" / grabadora digital de audio ha recibido cobertura durante mucho tiempo en los medios masivos de Japón y otros países, como un producto acústico electrónico nuevo y controversial de la industria japonesa de artefactos electrónicos. ¿Qué ha pasado con el objeto de esta controversia?

MAP-Motivated Carrier Synchronization of GMSK Based on the Laurent AMP Representation

Science.gov (United States)

Simon, M. K.

1998-01-01

Using the MAP estimation approach to carrier synchronization of digital modulations containing ISI together with a two pulse stream AMP representation of GMSK, it is possible to obtain an optimum closed loop configuration in the same manner as has been previously proposed for other conventional modulations with ISI.

Structural basis for phosphopantetheinyl carrier domain interactions in the terminal module of nonribosomal peptide synthetases

Science.gov (United States)

Liu, Ye; Zheng, Tengfei; Bruner, Steven D.

2011-01-01

Summary Phosphopantetheine-modified carrier domains play a central role in the template-directed, biosynthesis of several classes of primary and secondary metabolites. Fatty acids, polyketides and nonribosomal peptides are constructed on multidomain enzyme assemblies using phosphopantetheinyl thioester-linked carrier domains to traffic and activate building blocks. The carrier domain is a dynamic component of the process, shuttling pathway intermediates to sequential enzyme active sites. Here we report an approach to structurally fix carrier domain/enzyme constructs suitable for X-ray crystallographic analysis. The structure of a two-domain construct of E. coli EntF was determined with a conjugated phosphopantetheinyl-based inhibitor. The didomain structure is locked in an active orientation relevant to the chemistry of nonribosomal peptide biosynthesis. This structure provides details into the interaction of phosphopantetheine arm with the carrier domain and the active site of the thioesterase domain. PMID:22118682

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...

Photonic Implementation of 4-QAM/QPSK Electrical Modulation at Millimeter-Wave Frequency

DEFF Research Database (Denmark)

Yu, Xianbin; Jensen, Jesper Bevensee; Tafur Monroy, Idelfonso

2008-01-01

We propose a photonic method for generating millimeter-wave 4-QAM/QPSK modulated signals. The method is based on optical phase modulation by multilevel electrical signals and optical carrier-suppression. Simulation results are presented for 2.5 Gsymbol/s 4-QAM and QPSK signals at a 36 GHz carrier...

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

Directory of Open Access Journals (Sweden)

Warrick Roseboom

2011-04-01

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

Electroabsorption optical modulator

Energy Technology Data Exchange (ETDEWEB)

Skogen, Erik J.

2017-11-21

An electroabsorption modulator incorporates waveguiding regions along the length of the modulator that include quantum wells where at least two of the regions have quantum wells with different bandgaps. In one embodiment of the invention, the regions are arranged such that the quantum wells have bandgaps with decreasing bandgap energy along the length of the modulator from the modulator's input to its output. The bandgap energy of the quantum wells may be decreased in discrete steps or continuously. Advantageously, such an arrangement better distributes the optical absorption as well as the carrier density along the length of the modulator. Further advantageously, the modulator may handle increased optical power as compared with prior art modulators of similar dimensions, which allows for improved link gain when the optical modulator is used in an analog optical communication link.

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…

Measuring high-frequency responses of an electro-optic phase modulator based on dispersion induced phase modulation to intensity modulation conversion

Science.gov (United States)

Zhang, Shangjian; Wang, Heng; Wang, Yani; Zou, Xinhai; Zhang, Yali; Liu, Shuang; Liu, Yong

2014-11-01

We investigate the phase modulation to intensity modulation conversion in dispersive fibers for measuring frequency responses of electro-optic phase modulators, and demonstrate two typical measurements with cascade path and fold-back path. The measured results achieve an uncertainty of less than 2.8% within 20 GHz. Our measurements show stable and repeatable results because the optical carrier and its phase-modulated sidebands are affected by the same fiber impairments. The proposed method requires only dispersive fibers and works without any small-signal assumption, which is applicable for swept frequency measurement at different driving levels and operating wavelengths.

A conceptual framework for audio-visual museum media

DEFF Research Database (Denmark)

Kirkedahl Lysholm Nielsen, Mikkel

2017-01-01

In today's history museums, the past is communicated through many other means than original artefacts. This interdisciplinary and theoretical article suggests a new approach to studying the use of audio-visual media, such as film, video and related media types, in a museum context. The centre...... and museum studies, existing case studies, and real life observations, the suggested framework instead stress particular characteristics of contextual use of audio-visual media in history museums, such as authenticity, virtuality, interativity, social context and spatial attributes of the communication...

The Single- and Multichannel Audio Recordings Database (SMARD)

DEFF Research Database (Denmark)

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

2014-01-01

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

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.

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…

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

OpenAIRE

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

2013-01-01

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

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

DEFF Research Database (Denmark)

Turchet, Luca; Serafin, Stefania; Dimitrov, Smilen

2010-01-01

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

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

Science.gov (United States)

2016-03-01

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

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.

Radioactive Decay: Audio Data Collection

Science.gov (United States)

Struthers, Allan

2009-01-01

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

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

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.

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.

A Perceptual Model for Sinusoidal Audio Coding Based on Spectral Integration

NARCIS (Netherlands)

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

2005-01-01

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

A perceptual model for sinusoidal audio coding based on spectral integration

NARCIS (Netherlands)

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

2005-01-01

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

Research on a Micro-Nano Si/SiGe/Si Double Heterojunction Electro-Optic Modulation Structure

Directory of Open Access Journals (Sweden)

Song Feng

2018-01-01

Full Text Available The electro-optic modulator is a very important device in silicon photonics, which is responsible for the conversion of optical signals and electrical signals. For the electro-optic modulator, the carrier density of waveguide region is one of the key parameters. The traditional method of increasing carrier density is to increase the external modulation voltage, but this way will increase the modulation loss and also is not conducive to photonics integration. This paper presents a micro-nano Si/SiGe/Si double heterojunction electro-optic modulation structure. Based on the band theory of single heterojunction, the barrier heights are quantitatively calculated, and the carrier concentrations of heterojunction barrier are analyzed. The band and carrier injection characteristics of the double heterostructure structure are simulated, respectively, and the correctness of the theoretical analysis is demonstrated. The micro-nano Si/SiGe/Si double heterojunction electro-optic modulation is designed and tested, and comparison of testing results between the micro-nano Si/SiGe/Si double heterojunction micro-ring electro-optic modulation and the micro-nano Silicon-On-Insulator (SOI micro-ring electro-optic modulation, Free Spectrum Range, 3 dB Bandwidth, Q value, extinction ratio, and other parameters of the micro-nano Si/SiGe/Si double heterojunction micro-ring electro-optic modulation are better than others, and the modulation voltage and the modulation loss are lower.

Ferrite bead effect on Class-D amplifier audio quality

OpenAIRE

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

2014-01-01

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

Nanostructured lipid carriers system: recent advances in drug delivery.

Science.gov (United States)

Iqbal, Md Asif; Md, Shadab; Sahni, Jasjeet Kaur; Baboota, Sanjula; Dang, Shweta; Ali, Javed

2012-12-01

Nanostructured lipid carrier (NLC) is second generation smarter drug carrier system having solid matrix at room temperature. This carrier system is made up of physiological, biodegradable and biocompatible lipid materials and surfactants and is accepted by regulatory authorities for application in different drug delivery systems. The availability of many products in the market in short span of time reveals the success story of this delivery system. Since the introduction of the first product, around 30 NLC preparations are commercially available. NLC exhibit superior advantages over other colloidal carriers viz., nanoemulsions, polymeric nanoparticles, liposomes, SLN etc. and thus, have been explored to more extent in pharmaceutical technology. The whole set of unique advantages such as enhanced drug loading capacity, prevention of drug expulsion, leads to more flexibility for modulation of drug release and makes NLC versatile delivery system for various routes of administration. The present review gives insights on the definitions and characterization of NLC as colloidal carriers including the production techniques and suitable formulations. This review paper also highlights the importance of NLC in pharmaceutical applications for the various routes of drug delivery viz., topical, oral, pulmonary, ocular and parenteral administration and its future perspective as a pharmaceutical carrier.

Phase-Modulated Optical Communication Systems

CERN Document Server

Ho, Keang-Po

2005-01-01

Fiber-optic communication systems have revolutionized our telecommunication infrastructures – currently, almost all telephone land-line, cellular, and internet communications must travel via some form of optical fibers. In these transmission systems, neither the phase nor frequency of the optical signal carries information – only the intensity of the signal is used. To transmit more information in a single optical carrier, the phase of the optical carrier must be explored. As a result, there is renewed interest in phase-modulated optical communications, mainly in direct-detection DPSK signals for long-haul optical communication systems. When optical amplifiers are used to maintain certain signal level among the fiber link, the system is limited by amplifier noises and fiber nonlinearities. Phase-Modulated Optical Communication Systems surveys this newly popular area, covering the following topics: The transmitter and receiver for phase-modulated coherent lightwave systems Method for performance analysis o...

High modulation bandwidth of a light-emitting diode with surface plasmon coupling (Conference Presentation)

Science.gov (United States)

Lin, Chun-Han; Tu, Charng-Gan; Yao, Yu-Feng; Chen, Sheng-Hung; Su, Chia-Ying; Chen, Hao-Tsung; Kiang, Yean-Woei; Yang, Chih-Chung

2017-02-01

Besides lighting, LEDs can be used for indoor data transmission. Therefore, a large modulation bandwidth becomes an important target in the development of visible LED. In this regard, enhancing the radiative recombination rate of carriers in the quantum wells of an LED is a useful method since the modulation bandwidth of an LED is related to the carrier decay rate besides the device RC time constant To increase the carrier decay rate in an LED without sacrificing its output power, the technique of surface plasmon (SP) coupling in an LED is useful. In this paper, the increases of modulation bandwidth by reducing mesa size, decreasing active layer thickness, and inducing SP coupling in blue- and green-emitting LEDs are illustrated. The results are demonstrated by comparing three different LED surface structures, including bare p-type surface, GaZnO current spreading layer, and Ag nanoparticles (NPs) for inducing SP coupling. In a single-quantum-well, blue-emitting LED with a circular mesa of 10 microns in radius, SP coupling results in a modulation bandwidth of 528.8 MHz, which is believed to be the record-high level. A smaller RC time constant can lead to a higher modulation bandwidth. However, when the RC time constant is smaller than 0.2 ns, its effect on modulation bandwidth saturates. The dependencies of modulation bandwidth on injected current density and carrier decay time confirm that the modulation bandwidth is essentially inversely proportional to a time constant, which is inversely proportional to the square-root of carrier decay rate and injected current density.

Analysis of small-signal intensity modulation of semiconductor ...

Indian Academy of Sciences (India)

Computer simulation of the model is applied to 1.55-µm ... Semiconductor laser; small-signal modulation; modulation response; gain suppression. ... originates from intraband relaxation processes of charge carriers that extend for times as ...

Sparse coding of the modulation spectrum for noise-robust automatic speech recognition

NARCIS (Netherlands)

Ahmadi, S.; Ahadi, S.M.; Cranen, B.; Boves, L.W.J.

2014-01-01

The full modulation spectrum is a high-dimensional representation of one-dimensional audio signals. Most previous research in automatic speech recognition converted this very rich representation into the equivalent of a sequence of short-time power spectra, mainly to simplify the computation of the

Advances in coherent optical modems and 16-QAM transmission with feedforward carrier recovery

Science.gov (United States)

Noé, Reinhold; Hoffmann, Sebastian; Wördehoff, Christian; Al-Bermani, Ali; El-Darawy, Mohamed

2011-01-01

Polarization multiplexing and quadrature phase shift keying (QPSK) both double spectral efficiency. Combined with synchronous coherent polarization diverse intradyne receivers this modulation format is ultra-robust and cost-efficient. A feedforward carrier recovery is required in order to tolerate phase noise of normal DFB lasers. Signal processing in the digital domain permits compensation of at least chromatic and polarization mode dispersion. Some companies have products on the market, others are working on them. For 100 GbE transmission, 50 GHz channel spacing is sufficient. 16ary quadrature amplitude modulation (16-QAM) is attractive to double capacity once more, possibly in a modulation format flexible transponder which is switched down to QPSK only if system margin is too low. For 16-QAM the phase noise problem is sharply increased. However, also here a feedforward carrier recovery has been implemented. A number of carrier phase angles is tested in parallel, and the recovered data is selected for that phase angle where squared distance of recovered data to the nearest constellation point, averaged over a number of symbols, is minimum. An intradyne/selfhomodyne synchronous coherent 16-QAM experiment (2.5 Gb/s, 81 km) is presented.

Audio production principles practical studio applications

CERN Document Server

Elmosnino, Stephane

2018-01-01

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

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

Directory of Open Access Journals (Sweden)

Giovany Vega

2016-06-01

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

Securing Digital Audio using Complex Quadratic Map

Science.gov (United States)

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

2018-03-01

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

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

Directory of Open Access Journals (Sweden)

Soraya Ferreira Alves

2016-12-01

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

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

Energy Technology Data Exchange (ETDEWEB)

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

2005-07-01

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

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....

An Exploratory Evaluation of User Interfaces for 3D Audio Mixing

DEFF Research Database (Denmark)

Gelineck, Steven; Korsgaard, Dannie Michael

2015-01-01

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

A 240W Monolithic Class-D Audio Amplifier Output Stage

DEFF Research Database (Denmark)

Nyboe, Flemming; Kaya, Cetin; Risbo, Lars

2006-01-01

A single-channel class-D audio amplifier output stage outputs 240W undipped into 4Omega 0.1% open-loop THD+N allows using the device in a fully-digital audio signal path with no feedback. The output current capability is plusmn18A and the part is fabricated in a 0.4mum/1.8mum high-voltage Bi...

Precision Scaling of Neural Networks for Efficient Audio Processing

OpenAIRE

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

2017-01-01

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

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...

Pulse width modulation inverter with battery charger

Science.gov (United States)

Slicker, James M.

1985-01-01

An inverter is connected between a source of DC power and a three-phase AC induction motor, and a microprocessor-based circuit controls the inverter using pulse width modulation techniques. In the disclosed method of pulse width modulation, both edges of each pulse of a carrier pulse train are equally modulated by a time proportional to sin .theta., where .theta. is the angular displacement of the pulse center at the motor stator frequency from a fixed reference point on the carrier waveform. The carrier waveform frequency is a multiple of the motor stator frequency. The modulated pulse train is then applied to each of the motor phase inputs with respective phase shifts of 120.degree. at the stator frequency. Switching control commands for electronic switches in the inverter are stored in a random access memory (RAM) and the locations of the RAM are successively read out in a cyclic manner, each bit of a given RAM location controlling a respective phase input of the motor. The DC power source preferably comprises rechargeable batteries and all but one of the electronic switches in the inverter can be disabled, the remaining electronic switch being part of a "flyback" DC-DC converter circuit for recharging the battery.

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…

Revealing the ecological content of long-duration audio-recordings of the environment through clustering and visualisation.

Science.gov (United States)

Phillips, Yvonne F; Towsey, Michael; Roe, Paul

2018-01-01

Audio recordings of the environment are an increasingly important technique to monitor biodiversity and ecosystem function. While the acquisition of long-duration recordings is becoming easier and cheaper, the analysis and interpretation of that audio remains a significant research area. The issue addressed in this paper is the automated reduction of environmental audio data to facilitate ecological investigations. We describe a method that first reduces environmental audio to vectors of acoustic indices, which are then clustered. This can reduce the audio data by six to eight orders of magnitude yet retain useful ecological information. We describe techniques to visualise sequences of cluster occurrence (using for example, diel plots, rose plots) that assist interpretation of environmental audio. Colour coding acoustic clusters allows months and years of audio data to be visualised in a single image. These techniques are useful in identifying and indexing the contents of long-duration audio recordings. They could also play an important role in monitoring long-term changes in species abundance brought about by habitat degradation and/or restoration.

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

Science.gov (United States)

Dickinson Public Schools, ND. Instructional Media Center.

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

Audio Restoration

Science.gov (United States)

Esquef, Paulo A. A.

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

Carrier-based modulation schemes for various three-level matrix converters

DEFF Research Database (Denmark)

Blaabjerg, Frede; Loh, P.C.; Rong, R.C.

2008-01-01

different performance merits. To avoid confusion and hence fasten the converter applications in the industry, it would surely be better for modulation schemes to be developed from a common set of modulation principles that unfortunately has not yet been thoroughly defined. Contributing to that area...... a limited set of switching vectors because of its lower semiconductor count. Through simulation and experimental testing, all the evaluated matrix converters are shown to produce satisfactory sinusoidal input and output quantities using the same set of generic modulation principles, which can conveniently...

Design And Construction Of 300W Audio Power Amplifier For Classroom

Directory of Open Access Journals (Sweden)

Shune Lei Aung

2015-07-01

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

Créer des ressources audio pour le cours de FLE

Directory of Open Access Journals (Sweden)

Florence Gérard Lojacono

2010-01-01

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

Perceptual Coding of Audio Signals Using Adaptive Time-Frequency Transform

Directory of Open Access Journals (Sweden)

Umapathy Karthikeyan

2007-01-01

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

Laser interferometric method for determining the carrier diffusion length in semiconductors

Energy Technology Data Exchange (ETDEWEB)

Manukhov, V. V. [Saint Petersburg State University (Russian Federation); Fedortsov, A. B.; Ivanov, A. S., E-mail: ivaleks58@gmail.com [Saint Petersburg Mining University (Russian Federation)

2015-09-15

A new laser interferometric method for measuring the carrier diffusion length in semiconductors is proposed. The method is based on the interference–absorption interaction of two laser radiations in a semiconductor. Injected radiation generates additional carriers in a semiconductor, which causes a change in the material’s optical constants and modulation of the probing radiation passed through the sample. When changing the distance between carrier generation and probing points, a decrease in the carrier concentration, which depends on the diffusion length, is recorded. The diffusion length is determined by comparing the experimental and theoretical dependences of the probe signal on the divergence of the injector and probe beams. The method is successfully tested on semiconductor samples with different thicknesses and surface states and can be used in scientific research and the electronics industry.

"Is It Worth Knowing?" Focus Group Participants' Perceived Utility of Genomic Preconception Carrier Screening.

Science.gov (United States)

Schneider, Jennifer L; Goddard, Katrina A B; Davis, James; Wilfond, Benjamin; Kauffman, Tia L; Reiss, Jacob A; Gilmore, Marian; Himes, Patricia; Lynch, Frances L; Leo, Michael C; McMullen, Carmit

2016-02-01

As genome sequencing technology advances, research is needed to guide decision-making about what results can or should be offered to patients in different clinical settings. We conducted three focus groups with individuals who had prior preconception genetic testing experience to explore perceived advantages and disadvantages of genome sequencing for preconception carrier screening, compared to usual care. Using a discussion guide, a trained qualitative moderator facilitated the audio-recorded focus groups. Sixteen individuals participated. Thematic analysis of transcripts started with a grounded approach and subsequently focused on participants' perceptions of the value of genetic information. Analysis uncovered two orientations toward genomic preconception carrier screening: "certain" individuals desiring all possible screening information; and "hesitant" individuals who were more cautious about its value. Participants revealed valuable information about barriers to screening: fear/anxiety about results; concerns about the method of returning results; concerns about screening necessity; and concerns about partner participation. All participants recommended offering choice to patients to enhance the value of screening and reduce barriers. Overall, two groups of likely users of genome sequencing for preconception carrier screening demonstrated different perceptions of the advantages or disadvantages of screening, suggesting tailored approaches to education, consent, and counseling may be warranted with each group.

Modulation detection as a function of carrier frequency and level

NARCIS (Netherlands)

Fassel, R.; Kohlrausch, A.G.

1995-01-01

This paper describes recent experiments investigating temporal processing in the auditory system. Subjects had to discriminate sinusoidal signals with a flat temporal envelope from those with a sinusoidal amplitude modulation. The modulation depth at threshold was measured for a wide range of

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

Directory of Open Access Journals (Sweden)

Rita Kumalasari

2017-05-01

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

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

Science.gov (United States)

Vigil, Kathleen Murray

2017-01-01

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

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...

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

NARCIS (Netherlands)

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

2015-01-01

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

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

NARCIS (Netherlands)

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

2010-01-01

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

Recognition of Activities of Daily Living Based on Environmental Analyses Using Audio Fingerprinting Techniques: A Systematic Review

Directory of Open Access Journals (Sweden)

Ivan Miguel Pires

2018-01-01

Full Text Available An increase in the accuracy of identification of Activities of Daily Living (ADL is very important for different goals of Enhanced Living Environments and for Ambient Assisted Living (AAL tasks. This increase may be achieved through identification of the surrounding environment. Although this is usually used to identify the location, ADL recognition can be improved with the identification of the sound in that particular environment. This paper reviews audio fingerprinting techniques that can be used with the acoustic data acquired from mobile devices. A comprehensive literature search was conducted in order to identify relevant English language works aimed at the identification of the environment of ADLs using data acquired with mobile devices, published between 2002 and 2017. In total, 40 studies were analyzed and selected from 115 citations. The results highlight several audio fingerprinting techniques, including Modified discrete cosine transform (MDCT, Mel-frequency cepstrum coefficients (MFCC, Principal Component Analysis (PCA, Fast Fourier Transform (FFT, Gaussian mixture models (GMM, likelihood estimation, logarithmic moduled complex lapped transform (LMCLT, support vector machine (SVM, constant Q transform (CQT, symmetric pairwise boosting (SPB, Philips robust hash (PRH, linear discriminant analysis (LDA and discrete cosine transform (DCT.

Pressure dependence of excited-state charge-carrier dynamics in organolead tribromide perovskites

Science.gov (United States)

Liu, X. C.; Han, J. H.; Zhao, H. F.; Yan, H. C.; Shi, Y.; Jin, M. X.; Liu, C. L.; Ding, D. J.

2018-05-01

Excited-state charge-carrier dynamics governs the performance of organometal trihalide perovskites (OTPs) and is strongly influenced by the crystal structure. Characterizing the excited-state charge-carrier dynamics in OTPs under high pressure is imperative for providing crucial insights into structure-property relations. Here, we conduct in situ high-pressure femtosecond transient absorption spectroscopy experiments to study the excited-state carrier dynamics of CH3NH3PbBr3 (MAPbBr3) under hydrostatic pressure. The results indicate that compression is an effective approach to modulate the carrier dynamics of MAPbBr3. Across each pressure-induced phase, carrier relaxation, phonon scattering, and Auger recombination present different pressure-dependent properties under compression. Responsiveness is attributed to the pressure-induced variation in the lattice structure, which also changes the electronic band structure. Specifically, simultaneous prolongation of carrier relaxation and Auger recombination is achieved in the ambient phase, which is very valuable for excess energy harvesting. Our discussion provides clues for optimizing the photovoltaic performance of OTPs.

Can audio recording of outpatient consultations improve patient outcome?

DEFF Research Database (Denmark)

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

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

A 240W Monolithic Class-D Audio Amplifier Output Stage

OpenAIRE

Nyboe, Flemming; Kaya, Cetin; Risbo, Lars; Andreani, Pietro

2006-01-01

A single-channel class-D audio amplifier output stage outputs 240W undipped into 4Omega 0.1% open-loop THD+N allows using the device in a fully-digital audio signal path with no feedback. The output current capability is plusmn18A and the part is fabricated in a 0.4mum/1.8mum high-voltage BiCMOS process. Over-current sensing protects the output from short circuits.

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

Science.gov (United States)

Oringel, Robert S.

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

Fault Diagnosis using Audio and Vibration Signals in a Circulating Pump

International Nuclear Information System (INIS)

Henríquez, P; Alonso, J B; Ferrer, M A; Travieso, C M; Gómez, G

2012-01-01

This paper presents the use of audio and vibration signals in fault diagnosis of a circulating pump. The novelty of this paper is the use of audio signals acquired by microphones. The objective of this paper is to determine if audio signals are capable to distinguish between normal and different abnormal conditions in a circulating pump. In order to compare results, vibration signals are also acquired and analysed. Wavelet package is used to obtain the energies in different frequency bands from the audio and vibration signals. Neural networks are used to evaluate the discrimination ability of the extracted features between normal and fault conditions. The results show that information from sound signals can distinguish between normal and different faulty conditions with a success rate of 83.33%, 98% and 91.33% for each microphone respectively. These success rates are similar and even higher that those obtained from accelerometers (68%, 90.67% and 71.33% for each accelerometer respectively). Success rates also show that the position of microphones and accelerometers affects on the final results.

Minority Carrier Lifetime Studies of Narrow Bandgap Antimonide Superlattices

Science.gov (United States)

Hoglund, Linda; Ting, David Z.; Khoshakhlagh, Arezou; Soibel, Alexander; Hill, Cory J.; Fisher, Anita; Keo, Sam; Gunapala, Sarath D.

2014-01-01

In this study optical modulation response and photoluminescence spectroscopy were used to study mid-wave Ga-free InAs/InAsSb superlattices. The minority carrier lifetimes in the different samples varied from 480 ns to 4700 ns, partly due to different background doping concentrations. It was shown that the photoluminescence intensity can be used as a fast non-destructive tool to predict the material quality. It was also demonstrated that it is crucial to use a low excitation power in the photoluminescence measurements in order to get a good correlation between the photoluminescence intensity and the minority carrier lifetime.

V123 Beam Synchronous Encoder Module

International Nuclear Information System (INIS)

Kerner, T.; Conkling, C. R.; Oerter, B.

1999-01-01

The V123 Synchronous Encoder Module transmits events to distributed trigger modules and embedded decoders around the RHIC rings where they are used to provide beam instrumentation triggers [1,2,3]. The RHIC beam synchronous event link hardware is mainly comprised of three VMEbus board designs, the central input modules (V201), and encoder modules (V123), and the distributed trigger modules (V124). Two beam synchronous links, one for each ring, are distributed via fiberoptic and fanned out via twisted wire pair cables. The V123 synchronizes with the RF system clock derived from the beam bucket frequency and a revolution fiducial pulse. The RF system clock is used to create the beam synchronous event link carrier and events are synchronized with the rotation fiducial. A low jitter RF clock is later recovered from this carrier by phase lock loops in the trigger modules. Prioritized hardware and software triggers fill up to 15 beam event code transmission slots per revolution while tracking the ramping RF acceleration frequency and storage frequency. The revolution fiducial event is always the first event transmitted which is used to synchronize the firing of the abort kicker and to locate the first bucket for decoders distributed about the ring

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

Directory of Open Access Journals (Sweden)

Koji Iwano

2007-03-01

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

Turkish Music Genre Classification using Audio and Lyrics Features

Directory of Open Access Journals (Sweden)

Önder ÇOBAN

2017-05-01

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

Perceptual Coding of Audio Signals Using Adaptive Time-Frequency Transform

Directory of Open Access Journals (Sweden)

Karthikeyan Umapathy

2007-08-01

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

The relationship between basic audio quality and overall listening experience.

Science.gov (United States)

Schoeffler, Michael; Herre, Jürgen

2016-09-01

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

ANALYSIS OF MULTIMODAL FUSION TECHNIQUES FOR AUDIO-VISUAL SPEECH RECOGNITION

Directory of Open Access Journals (Sweden)

D.V. Ivanko

2016-05-01

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

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

Science.gov (United States)

2010-01-01

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

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

International Nuclear Information System (INIS)

Kubo, H. Dale; Wang Lili

2002-01-01

With breathing synchronized radiotherapy (BSRT), a voltage signal derived from an organ displacement detector is usually displayed on the vertical axis whereas the elapsed time is shown on the horizontal axis. The voltage gate window is set on the breathing voltage signal. Whenever the breathing signal falls between the two gate levels, a gate pulse is produced to enable the treatment machine. In this paper a new gating mechanism, audio (or time-sequence) gating, is introduced and is integrated into the existing voltage gating system. The audio gating takes advantage of the repetitive nature of the breathing signal when repetitive audio instruction is given to the patient. The audio gating is aimed at removing the regions of sharp rises and falls in the breathing signal that cannot be removed by the voltage gating. When the breathing signal falls between voltage gate levels as well as between audio-gate levels, the voltage- and audio-gated radiotherapy (ART) system will generate an AND gate pulse. When this gate pulse is received by a linear accelerator, the linear accelerator becomes 'enabled' for beam delivery and will deliver the beam when all other interlocks are removed. This paper describes a new gating mechanism and a method of recording beam-on signal, both of which are, configured into a laptop computer. The paper also presents evidence of some clinical advantages achieved with the ART system

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

DEFF Research Database (Denmark)

Korhonen, Jari; Reiter, Ulrich; Myakotnykh, Eugene

2010-01-01

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

Temporal modulation transfer functions in cochlear implantees using a method that limits overall loudness cues

Science.gov (United States)

Fraser, Matthew; McKay, Colette M.

2012-01-01

Temporal modulation transfer functions (TMTFs) were measured for six users of cochlear implants, using different carrier rates and levels. Unlike most previous studies investigating modulation detection, the experimental design limited potential effects of overall loudness cues. Psychometric functions (percent correct discrimination of modulated from unmodulated stimuli versus modulation depth) were obtained. For each modulation depth, each modulated stimulus was loudness balanced to the unmodulated reference stimulus, and level jitter was applied in the discrimination task. The loudness-balance data showed that the modulated stimuli were louder than the unmodulated reference stimuli with the same average current, thus confirming the need to limit loudness cues when measuring modulation detection. TMTFs measured in this way had a low-pass characteristic, with a cut-off frequency (at comfortably loud levels) similar to that for normal-hearing listeners. A reduction in level caused degradation in modulation detection efficiency and a lower-cut-off frequency (i.e. poorer temporal resolution). An increase in carrier rate also led to a degradation in modulation detection efficiency, but only at lower levels or higher modulation frequencies. When detection thresholds were expressed as a proportion of dynamic range, there was no effect of carrier rate for the lowest modulation frequency (50 Hz) at either level. PMID:22146425

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

Directory of Open Access Journals (Sweden)

Md. Rabiul Islam

2014-01-01

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

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

Directory of Open Access Journals (Sweden)

Butko Taras

2011-01-01

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

Estimation of inhalation flow profile using audio-based methods to assess inhaler medication adherence

Science.gov (United States)

Lacalle Muls, Helena; Costello, Richard W.; Reilly, Richard B.

2018-01-01

Asthma and chronic obstructive pulmonary disease (COPD) patients are required to inhale forcefully and deeply to receive medication when using a dry powder inhaler (DPI). There is a clinical need to objectively monitor the inhalation flow profile of DPIs in order to remotely monitor patient inhalation technique. Audio-based methods have been previously employed to accurately estimate flow parameters such as the peak inspiratory flow rate of inhalations, however, these methods required multiple calibration inhalation audio recordings. In this study, an audio-based method is presented that accurately estimates inhalation flow profile using only one calibration inhalation audio recording. Twenty healthy participants were asked to perform 15 inhalations through a placebo Ellipta™ DPI at a range of inspiratory flow rates. Inhalation flow signals were recorded using a pneumotachograph spirometer while inhalation audio signals were recorded simultaneously using the Inhaler Compliance Assessment device attached to the inhaler. The acoustic (amplitude) envelope was estimated from each inhalation audio signal. Using only one recording, linear and power law regression models were employed to determine which model best described the relationship between the inhalation acoustic envelope and flow signal. Each model was then employed to estimate the flow signals of the remaining 14 inhalation audio recordings. This process repeated until each of the 15 recordings were employed to calibrate single models while testing on the remaining 14 recordings. It was observed that power law models generated the highest average flow estimation accuracy across all participants (90.89±0.9% for power law models and 76.63±2.38% for linear models). The method also generated sufficient accuracy in estimating inhalation parameters such as peak inspiratory flow rate and inspiratory capacity within the presence of noise. Estimating inhaler inhalation flow profiles using audio based methods may be

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…

Digital signal processing methods and algorithms for audio conferencing systems

OpenAIRE

Lindström, Fredric

2007-01-01

Today, we are interconnected almost all over the planet. Large multinational companies operate worldwide, but also an increasing number of small and medium sized companies do business overseas. As people travel to meet and do businesses, the already exposed earth is subject to even more strain. Audio conferencing is an attractive alternative to travel, which is becoming more and more appreciated. Audio conferences can of course not replace all types of meetings, but can help companies to cut ...

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

Science.gov (United States)

Pimmer, Christoph; Mbvundula, Francis

2018-01-01

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

Impact of Audio-Coaching on the Position of Lung Tumors

International Nuclear Information System (INIS)

Haasbeek, Cornelis J.A.; Spoelstra, Femke; Lagerwaard, Frank J.; Soernsen de Koste, John R. van; Cuijpers, Johan P.; Slotman, Ben J.; Senan, Suresh

2008-01-01

Purpose: Respiration-induced organ motion is a major source of positional, or geometric, uncertainty in thoracic radiotherapy. Interventions to mitigate the impact of motion include audio-coached respiration-gated radiotherapy (RGRT). To assess the impact of coaching on average tumor position during gating, we analyzed four-dimensional computed tomography (4DCT) scans performed both with and without audio-coaching. Methods and Materials: Our RGRT protocol requires that an audio-coached 4DCT scan is performed when the initial free-breathing 4DCT indicates a potential benefit with gating. We retrospectively analyzed 22 such paired scans in patients with well-circumscribed tumors. Changes in lung volume and position of internal target volumes (ITV) generated in three consecutive respiratory phases at both end-inspiration and end-expiration were analyzed. Results: Audio-coaching increased end-inspiration lung volumes by a mean of 10.2% (range, -13% to +43%) when compared with free breathing (p = 0.001). The mean three-dimensional displacement of the center of ITV was 3.6 mm (SD, 2.5; range, 0.3-9.6mm), mainly caused by displacement in the craniocaudal direction. Displacement of ITV caused by coaching was more than 5 mm in 5 patients, all of whom were in the subgroup of 9 patients showing total tumor motion of 10 mm or more during both coached and uncoached breathing. Comparable ITV displacements were observed at end-expiration phases of the 4DCT. Conclusions: Differences in ITV position exceeding 5 mm between coached and uncoached 4DCT scans were detected in up to 56% of mobile tumors. Both end-inspiration and end-expiration RGRT were susceptible to displacements. This indicates that the method of audio-coaching should remain unchanged throughout the course of treatment

audio-ultrasonic waves by argon gas discharge

International Nuclear Information System (INIS)

Ragheb, M.S.

2010-01-01

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

Automatic Detection and Classification of Audio Events for Road Surveillance Applications

Directory of Open Access Journals (Sweden)

Noor Almaadeed

2018-06-01

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

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.

Filtered Carrier Phase Estimator for High-Order QAM Optical Systems

DEFF Research Database (Denmark)

Rozental, Valery; Kong, Deming; Corcoran, Bill

2018-01-01

We investigate, using Monte Carlo simulations, the performance characteristics and limits of a low-complexity filtered carrier phase estimator (F-CPE) in terms of cycle slip occurrences and SNR penalties. In this work, the F-CPE algorithm has been extended to include modulation formats whose oute...

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

Science.gov (United States)

Le, Long N; Jones, Douglas L

2018-03-01

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

Study of audio speakers containing ferrofluid

Energy Technology Data Exchange (ETDEWEB)

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

2008-05-21

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

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

GaN Power Stage for Switch-mode Audio Amplification

DEFF Research Database (Denmark)

Ploug, Rasmus Overgaard; Knott, Arnold; Poulsen, Søren Bang

2015-01-01

Gallium Nitride (GaN) based power transistors are gaining more and more attention since the introduction of the enhancement mode eGaN Field Effect Transistor (FET) which makes an adaptation from Metal-Oxide Semiconductor (MOSFET) to eGaN based technology less complex than by using depletion mode Ga......N FETs. This project seeks to investigate the possibilities of using eGaN FETs as the power switching device in a full bridge power stage intended for switch mode audio amplification. A 50 W 1 MHz power stage was built and provided promising audio performance. Future work includes optimization of dead...

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

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.

Estimation of the center frequency of the highest modulation filter.

Science.gov (United States)

Moore, Brian C J; Füllgrabe, Christian; Sek, Aleksander

2009-02-01

For high-frequency sinusoidal carriers, the threshold for detecting sinusoidal amplitude modulation increases when the signal modulation frequency increases above about 120 Hz. Using the concept of a modulation filter bank, this effect might be explained by (1) a decreasing sensitivity or greater internal noise for modulation filters with center frequencies above 120 Hz; and (2) a limited span of center frequencies of the modulation filters, the top filter being tuned to about 120 Hz. The second possibility was tested by measuring modulation masking in forward masking using an 8 kHz sinusoidal carrier. The signal modulation frequency was 80, 120, or 180 Hz and the masker modulation frequencies covered a range above and below each signal frequency. Four highly trained listeners were tested. For the 80-Hz signal, the signal threshold was usually maximal when the masker frequency equaled the signal frequency. For the 180-Hz signal, the signal threshold was maximal when the masker frequency was below the signal frequency. For the 120-Hz signal, two listeners showed the former pattern, and two showed the latter pattern. The results support the idea that the highest modulation filter has a center frequency in the range 100-120 Hz.

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

Science.gov (United States)

Aleman-Centeno, Josefina R.

1983-01-01

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

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

Directory of Open Access Journals (Sweden)

Kirsten E Smayda

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

FPGA Techniques Based New Hybrid Modulation Strategies for Voltage Source Inverters

Science.gov (United States)

Sudha, L. U.; Baskaran, J.; Elankurisil, S. A.

2015-01-01

This paper corroborates three different hybrid modulation strategies suitable for single-phase voltage source inverter. The proposed method is formulated using fundamental switching and carrier based pulse width modulation methods. The main tale of this proposed method is to optimize a specific performance criterion, such as minimization of the total harmonic distortion (THD), lower order harmonics, switching losses, and heat losses. The proposed method is articulated using fundamental switching and carrier based pulse width modulation methods. Thus, the harmonic pollution in the power system will be reduced and the power quality will be augmented with better harmonic profile for a target fundamental output voltage. The proposed modulation strategies are simulated in MATLAB r2010a and implemented in a Xilinx spartan 3E-500 FG 320 FPGA processor. The feasibility of these modulation strategies is authenticated through simulation and experimental results. PMID:25821852

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

Science.gov (United States)

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

2016-04-01

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

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

Directory of Open Access Journals (Sweden)

Bo Xiao

2016-04-01

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

Large modulation of carrier transport by grain-boundary molecular packing and microstructure in organic thin films

KAUST Repository

Rivnay, Jonathan

2009-11-08

Solution-processable organic semiconductors are central to developing viable printed electronics, and performance comparable to that of amorphous silicon has been reported for films grown from soluble semiconductors. However, the seemingly desirable formation of large crystalline domains introduces grain boundaries, resulting in substantial device-to-device performance variations. Indeed, for films where the grain-boundary structure is random, a few unfavourable grain boundaries may dominate device performance. Here we isolate the effects of molecular-level structure at grain boundaries by engineering the microstructure of the high-performance n-type perylenediimide semiconductor PDI8-CN 2 and analyse their consequences for charge transport. A combination of advanced X-ray scattering, first-principles computation and transistor characterization applied to PDI8-CN 2 films reveals that grain-boundary orientation modulates carrier mobility by approximately two orders of magnitude. For PDI8-CN 2 we show that the molecular packing motif (that is, herringbone versus slip-stacked) plays a decisive part in grain-boundary-induced transport anisotropy. The results of this study provide important guidelines for designing device-optimized molecular semiconductors. © 2009 Macmillan Publishers Limited. All rights reserved.

Tuning and disrupting the brain – modulating the McGurk illusion with electrical stimulation

Directory of Open Access Journals (Sweden)

Lucas M Marques

2014-08-01

Full Text Available In the so-called McGurk illusion, when the synchronized presentation of the visual stimulus /ga/ is paired with the auditory stimulus /ba/, people in general hear it as /da/. Multisensory integration processing underlying this illusion seems to occur within the Superior Temporal Sulcus (STS. Herein, we present evidence demonstrating that bilateral cathodal transcranial direct current stimulation (tDCS of this area can decrease the McGurk illusion-type responses. Additionally, we show that the manipulation of this audio-visual integrated output occurs irrespective of the number of eye-fixations on the mouth of the speaker. Bilateral anodal tDCS of the Parietal Cortex also modulates the illusion, but in the opposite manner, inducing more illusion-type responses. This is the first demonstration of using non-invasive brain stimulation to modulate multisensory speech perception in an illusory context (i.e., both increasing and decreasing illusion-type responses to a verbal audio-visual integration task. These findings provide clear evidence that both the superior temporal and parietal areas contribute to multisensory integration processing related to speech perception. Specifically, STS seems fundamental for the temporal synchronization and integration of auditory and visual inputs. For its part, PPC may adjust the arrival of incoming audio and visual information to STS thereby enhancing their interaction in this latter area.

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...

Cognitive digital receiver for burst mode phase modulated radio over fiber links

DEFF Research Database (Denmark)

Guerrero Gonzalez, Neil; Zibar, Darko; Tafur Monroy, Idelfonso

2010-01-01

A novel cognitive receiver for modulation format recognition with reconfigurable carrier recovery scheme is proposed and experimentally demonstrated for phase modulated radio-over-fibre links. Demodulation of burst-mode mixed modulation formats (PSK and QAM) is demonstrated after 40km...

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

Science.gov (United States)

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

2012-01-01

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

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

Science.gov (United States)

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

2012-01-01

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

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

International Nuclear Information System (INIS)

Mirza, A.H.

1999-01-01

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

Simultaneous determination of effective carrier lifetime and resistivity of Si wafers using the nonlinear nature of photocarrier radiometric signals

Science.gov (United States)

Sun, Qiming; Melnikov, Alexander; Wang, Jing; Mandelis, Andreas

2018-04-01

A rigorous treatment of the nonlinear behavior of photocarrier radiometric (PCR) signals is presented theoretically and experimentally for the quantitative characterization of semiconductor photocarrier recombination and transport properties. A frequency-domain model based on the carrier rate equation and the classical carrier radiative recombination theory was developed. The derived concise expression reveals different functionalities of the PCR amplitude and phase channels: the phase bears direct quantitative correlation with the carrier effective lifetime, while the amplitude versus the estimated photocarrier density dependence can be used to extract the equilibrium majority carrier density and thus, resistivity. An experimental ‘ripple’ optical excitation mode (small modulation depth compared to the dc level) was introduced to bypass the complicated ‘modulated lifetime’ problem so as to simplify theoretical interpretation and guarantee measurement self-consistency and reliability. Two Si wafers with known resistivity values were tested to validate the method.

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

Science.gov (United States)

Anderson, Gerald D.; Olson, David B.

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

Dual-gate operation and carrier transport in SiGe p-n junction nanowires

Science.gov (United States)

Delker, C. J.; Yoo, J. Y.; Bussmann, E.; Swartzentruber, B. S.; Harris, C. T.

2017-11-01

We investigate carrier transport in silicon-germanium nanowires with an axial p-n junction doping profile by fabricating these wires into transistors that feature separate top gates over each doping segment. By independently biasing each gate, carrier concentrations in the n- and p-side of the wire can be modulated. For these devices, which were fabricated with nickel source-drain electrical contacts, holes are the dominant charge carrier, with more favorable hole injection occurring on the p-side contact. Channel current exhibits greater sensitivity to the n-side gate, and in the reverse biased source-drain configuration, current is limited by the nickel/n-side Schottky contact.

The Transfer of Learning Associated with Audio Feedback on Written Work

Directory of Open Access Journals (Sweden)

Tanya Martini

2014-11-01

Full Text Available This study examined whether audio feedback provided to undergraduates (N=51 about one paper would prove beneficial in terms of improving their grades on another, unrelated paper of the same type. We examined this issue both in terms of student beliefs about learning transfer, as well as their actual ability to transfer what had been learned on one assignment to another, subsequent assignment. Results indicated that students believed that they would be able to transfer what they had learned via audio feedback. Moreover, results also suggested that students actually did generalize the overarching comments about content and structure made in the audio files to a subsequent paper, the content of which differed substantially from the initial one. Both students and teaching assistants demonstrated very favourable responses to this type of feedback, suggesting that it was both clear and comprehensive.

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

DEFF Research Database (Denmark)

Laursen, Ditte

This paper is a user evaluation of a mobile phone audio guide developed for visitors to use at the National Gallery of Denmark. The audio guide is offered as a downloadable MP3 file to every incoming visitor who is carrying a mobile phone with an open Bluetooth connection. The guide itself...... according to personal interest, and a conflict between the expectation of a learning experience rather than an aesthetic experience. Results indicate that most visitors are able to make sense of the guide and to use it successfully, in different ways, to enrich their visit. Evaluation also shows...... that visitors are fond of using their own mobile phones - but they have several problems with their phones in downloading the MP3 file. Read more: Guided Expectations: A Case Study of a Sound Collage Audio Guide | conference.archimuse.com...

Cambridge English First 2 audio CDs : authentic examination papers

CERN Document Server

2016-01-01

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

Audio power amplifier techniques with energy efficient power conversion. Vol. 1

Energy Technology Data Exchange (ETDEWEB)

Nielsen, Karsten

1998-04-01

A fundamental study of both analog and digital pulse modulation methods is carried out. A novel class of multi-level pulse modulation methods - Phase Shifted Carrier Pulse Width Modulation (PSCPWM) - is introduced and show to have several advantageous features, primarily caused by the much improved synthesis of the modulating signal. Enhanced digital pulse modulation methods for digital Pulse Modulation Amplifier (PMA) systems are investigated, and a simple methodology for digital PWM modulator synthesis is devised. It is concluded, that the modulator performance is not a limitation in the system, regardless of the domain of modulator implementation. Power conversion in PMA systems is adressed from the perspective of both linearity and efficienty optimization. Based on detailed studies of the distortion mechanisms in the power conversion stage it is concluded, that this is the fundamental limitation on system performance due to several physical limitations. The analysis of general power stage efficiency concludes that dramatic improvements in energy efficiency are possible with PMA systems that are optimized for efficiency. A control system design methodology is devised as a platform for synthesis of robust control systems. Investigations of three fundamental control structures show that even simple control systems offer a remarkable value, although the considered topologies also have their limitations which is verified by practical evaluation in hardware. A novel control method is introduced - Multivariable Enhanced Cascade Control (MECC). MECC provides flexible control over all essential system parameters and is furthermore simple in realization. Practical evaluation of a MECC based PMA shows state-of-the-art performance. The application of non-linear control methods is investigated with the introduction of an enhanced non-linear control/modulator topology. Although the non-linear controller is theoretically interesting, the method proves to suffer from various

Audio-visual materials usage preference among agricultural ...

African Journals Online (AJOL)

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

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

DEFF Research Database (Denmark)

Dimitrov, Smilen; Serafin, Stefania

2011-01-01

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

Minimizing Crosstalk in Self Oscillating Switch Mode Audio Power Amplifiers

DEFF Research Database (Denmark)

Knott, Arnold; Ploug, Rasmus Overgaard

2012-01-01

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

Multilevel tracking power supply for switch-mode audio power amplifiers

DEFF Research Database (Denmark)

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

2018-01-01

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

Convolution-based classification of audio and symbolic representations of music

DEFF Research Database (Denmark)

Velarde, Gissel; Cancino Chacón, Carlos; Meredith, David

2018-01-01

We present a novel convolution-based method for classification of audio and symbolic representations of music, which we apply to classification of music by style. Pieces of music are first sampled to pitch–time representations (piano-rolls or spectrograms) and then convolved with a Gaussian filter......-class composer identification, methods specialised for classifying symbolic representations of music are more effective. We also performed experiments on symbolic representations, synthetic audio and two different recordings of The Well-Tempered Clavier by J. S. Bach to study the method’s capacity to distinguish...

Optically Modulated Multiband Terahertz Perfect Absorber

DEFF Research Database (Denmark)

Seren, Huseyin R.; Keiser, George R.; Cao, Lingyue

2014-01-01

response of resonant metamaterials continues to be a challengingendeavor. Resonant perfect absorbers have flourished as one of the mostpromising metamaterial devices with applications ranging from power har-vesting to terahertz imaging. Here, an optically modulated resonant perfectabsorber is presented....... Utilizing photo-excited free carriers in silicon padsplaced in the capacitive gaps of split ring resonators, a dynamically modu-lated perfect absorber is designed and fabricated to operate in reflection.Large modulation depth (38% and 91%) in two absorption bands (with 97%and 92% peak absorption...

Listening to an audio drama activates two processing networks, one for all sounds, another exclusively for speech.

Directory of Open Access Journals (Sweden)

Robert Boldt

Full Text Available Earlier studies have shown considerable intersubject synchronization of brain activity when subjects watch the same movie or listen to the same story. Here we investigated the across-subjects similarity of brain responses to speech and non-speech sounds in a continuous audio drama designed for blind people. Thirteen healthy adults listened for ∼19 min to the audio drama while their brain activity was measured with 3 T functional magnetic resonance imaging (fMRI. An intersubject-correlation (ISC map, computed across the whole experiment to assess the stimulus-driven extrinsic brain network, indicated statistically significant ISC in temporal, frontal and parietal cortices, cingulate cortex, and amygdala. Group-level independent component (IC analysis was used to parcel out the brain signals into functionally coupled networks, and the dependence of the ICs on external stimuli was tested by comparing them with the ISC map. This procedure revealed four extrinsic ICs of which two-covering non-overlapping areas of the auditory cortex-were modulated by both speech and non-speech sounds. The two other extrinsic ICs, one left-hemisphere-lateralized and the other right-hemisphere-lateralized, were speech-related and comprised the superior and middle temporal gyri, temporal poles, and the left angular and inferior orbital gyri. In areas of low ISC four ICs that were defined intrinsic fluctuated similarly as the time-courses of either the speech-sound-related or all-sounds-related extrinsic ICs. These ICs included the superior temporal gyrus, the anterior insula, and the frontal, parietal and midline occipital cortices. Taken together, substantial intersubject synchronization of cortical activity was observed in subjects listening to an audio drama, with results suggesting that speech is processed in two separate networks, one dedicated to the processing of speech sounds and the other to both speech and non-speech sounds.

The Role of Fast Carrier Dynamics in SOA Based Devices

DEFF Research Database (Denmark)

Mørk, Jesper; Berg, Tommy Winther; Nielsen, Mads Lønstrup

2004-01-01

and the properties of schemes relying on cross-gain as well as cross-phase modulation are discussed. The possible benefits of using SOAs with quantum dot active regions are theoretically analyzed. The bandfilling characteristics and the presence of fast capture processes may allow to reach bitrates in excess of 100......We describe the characteristics of all-optical switching schemes based on semiconductor optical amplifiers (SOAs), with particular emphasis on the role of the fast carrier dynamics. The SOA response to a single short pulse as well as to a data-modulated pulse train is investigated...

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

International Nuclear Information System (INIS)

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

2001-01-01

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

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

Energy Technology Data Exchange (ETDEWEB)

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

2001-11-01

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

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

Directory of Open Access Journals (Sweden)

Niels Böttcher

2013-01-01

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

Kinetics of photo-activated charge carriers in Sn:CdS

Energy Technology Data Exchange (ETDEWEB)

Patidar, Manju Mishra, E-mail: manjumishra.iuc@gmail.com; Gorli, V. R.; Gangrade, Mohan; Nath, R.; Ganesan, V. [UGC-DAE CSR, University Campus, Khandwa Road, Indore (M.P.)-452001 (India); Panda, Richa [S.S. Jain Subodh Girls College, Airport Road Sanganer, Jaipur - 302029 (India)

2016-05-23

Kinetics of the photo-activated charge carriers has been investigated in Tin substituted Cadmium Sulphide, Cd{sub 1-x}Sn{sub x}S (x=0, 0.05, 0.10 and 0.15), thin films prepared by spray pyrolysis. X-Ray Diffraction shows an increase in strain that resulted in the decreased crystallite size upon Sn substitution. At the first sight, the photo current characteristics show a quenching effect on Sn substitution. However, survival of persistent photocurrents is seen even up to 15% of Sn substitution. Transient photo current decay could be explained with a 2τ relaxation model. CdS normally has an n-type character and the Sn doping expected to inject hole carriers. The two fold increase in τ{sub 1}, increase in activation energy and the decrease in photocurrents upon Sn substitution point towards a band gap cleaning scenario that include compensation and associated carrier injection dynamics. In addition Atomic Force Microscopy shows a drastic change in microstructure that modulates the carrier dynamics as a whole.

Agency Video, Audio and Imagery Library

Science.gov (United States)

Grubbs, Rodney

2015-01-01

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

Digital intermediate frequency QAM modulator using parallel processing

Science.gov (United States)

Pao, Hsueh-Yuan [Livermore, CA; Tran, Binh-Nien [San Ramon, CA

2008-05-27

The digital Intermediate Frequency (IF) modulator applies to various modulation types and offers a simple and low cost method to implement a high-speed digital IF modulator using field programmable gate arrays (FPGAs). The architecture eliminates multipliers and sequential processing by storing the pre-computed modulated cosine and sine carriers in ROM look-up-tables (LUTs). The high-speed input data stream is parallel processed using the corresponding LUTs, which reduces the main processing speed, allowing the use of low cost FPGAs.

Audio feature extraction using probability distribution function

Science.gov (United States)

Suhaib, A.; Wan, Khairunizam; Aziz, Azri A.; Hazry, D.; Razlan, Zuradzman M.; Shahriman A., B.

2015-05-01

Voice recognition has been one of the popular applications in robotic field. It is also known to be recently used for biometric and multimedia information retrieval system. This technology is attained from successive research on audio feature extraction analysis. Probability Distribution Function (PDF) is a statistical method which is usually used as one of the processes in complex feature extraction methods such as GMM and PCA. In this paper, a new method for audio feature extraction is proposed which is by using only PDF as a feature extraction method itself for speech analysis purpose. Certain pre-processing techniques are performed in prior to the proposed feature extraction method. Subsequently, the PDF result values for each frame of sampled voice signals obtained from certain numbers of individuals are plotted. From the experimental results obtained, it can be seen visually from the plotted data that each individuals' voice has comparable PDF values and shapes.

Study and analysis of coefficient mismatch in a MASH21 sigma-delta modulator

Energy Technology Data Exchange (ETDEWEB)

Ge Binjie; Wang Xin' an; Zhang Xing; Feng Xiaoxing; Wang Qingqin, E-mail: wangxa@szpku.edu.c [Key Laboratory of Integrated Microsystem Science and Engineering Applications, Shenzhen Graduate School of Peking University, Shenzhen 518055 (China)

2010-01-15

The quantization noise leakage of the first stage in a MASH21 sigma-delta modulator is analyzed. The results show that the finite DC gain of the opamp is the main reason for noise leakage, and finite GBW and SR only generate harmonic distortion. The relationship between DC gain and leakage is modeled and conclusions on design criteria are reached. As an example, a MASH21 modulator for a digital audio application is realized. This modulator, fabricated in an 0.18 {mu}m mixed signal process, achieves an SNDR of 91 dB with 1.8 V supply, which verifies the analysis and design criteria. (semiconductor integrated circuits)

PENGEMBANGAN MULTIMEDIA PEMBELAJARAN FISIKA BERBASIS AUDIO-VIDEO EKSPERIMEN LISTRIK DINAMIS DI SMP

Directory of Open Access Journals (Sweden)

P. Rante

2013-10-01

Full Text Available Penelitian pengembangan ini dilakukan dengan tujuan untuk melihat profil pengembangan multimedia pembelajaran fisika berbasis audio-video eksperimen listrik dinamis yang dapat menjadi solusi ketidakterlaksanaan praktikum di sekolah. Hasil penelitian menunjukkan bahwa propil multimedia berbasis audio-video eksperimen dari segi tampilan menarik, fasilitas runtut, sistematis dan praktis digunakan serta menjadi solusi ketidakterlaksanaan praktikum di sekolah. Produk akhir adalah sebuah paket CD autorun multimedia pembelajaran interaktif sebagai media pembelajaran mandiri dan sebagai media presentase yang dilengkapi perangkat pembelajaran untuk guru. This research aims to see the profile of multimedia learning development on physics based audio-video on the topic dynamic electricity experiment that may become a solution of practicum that not mastered well in the school. The result shows that the profile of develop multimedia based audio-video experiment has interesting display, harmonious facilities, systematic and practical in used as well as become a solution of the practicum that not mastered yet. The final product produced an auto run CD package of interactive learning multimedia as a self learning media and as a representation of media that equipped with teaching and learning media for teacher.

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

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

Directory of Open Access Journals (Sweden)

Jean-Luc Schwartz

2014-07-01

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

Automatic summarization of soccer highlights using audio-visual descriptors.

Science.gov (United States)

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

2015-01-01

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

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

DEFF Research Database (Denmark)

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

2009-01-01

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

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.

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.

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

Science.gov (United States)

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

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

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

DEFF Research Database (Denmark)

Dimitrov, Smilen; Serafin, Stefania

2011-01-01

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

Underdetermined Blind Audio Source Separation Using Modal Decomposition

Directory of Open Access Journals (Sweden)

Abdeldjalil Aïssa-El-Bey

2007-03-01

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

Underdetermined Blind Audio Source Separation Using Modal Decomposition

Directory of Open Access Journals (Sweden)

Aïssa-El-Bey Abdeldjalil

2007-01-01

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

Photoabsorption modulation in GaAs: Ga1-xInx as strained-layer superlattices

International Nuclear Information System (INIS)

Sella, I.; Watkins, D.E.; Laurich, B.K.; Smith, D.L.; Subbanna, S.; Kroemer, H.

1990-01-01

Photoabsorption modulation measurements have been made on Ga 1 -x In x As -- GaAs strained-layer superlattices using two approaches: In the first the modulating beam and the test beam have the same wavelength (near the exciton resonance). In the second, the modulation wavelength is much shorter than the test beam wavelength. A dramatic difference is observed in the modulated transmission spectra near the excitonic level for the two modulating wavelengths. The difference in behavior can be explained by screening of the residual surface electric field, which only occurs for the high photon energy modulating beam. This beam excites carriers that are free to drift in the surface field before they are captured in the quantum wells. Carriers excited by the low photon energy modulation beam are created in the wells and can not effectively screen the surface field. We describe a model which explains the nonlinear intensity saturation profile and qualitatively describes the spectral line shape. 4 refs., 4 figs

Temperature feedback control for long-term carrier-envelope phase locking

Science.gov (United States)

Chang, Zenghu [Manhattan, KS; Yun, Chenxia [Manhattan, KS; Chen, Shouyuan [Manhattan, KS; Wang, He [Manhattan, KS; Chini, Michael [Manhattan, KS

2012-07-24

A feedback control module for stabilizing a carrier-envelope phase of an output of a laser oscillator system comprises a first photodetector, a second photodetector, a phase stabilizer, an optical modulator, and a thermal control element. The first photodetector may generate a first feedback signal corresponding to a first portion of a laser beam from an oscillator. The second photodetector may generate a second feedback signal corresponding to a second portion of the laser beam filtered by a low-pass filter. The phase stabilizer may divide the frequency of the first feedback signal by a factor and generate an error signal corresponding to the difference between the frequency-divided first feedback signal and the second feedback signal. The optical modulator may modulate the laser beam within the oscillator corresponding to the error signal. The thermal control unit may change the temperature of the oscillator corresponding to a signal operable to control the optical modulator.

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

OpenAIRE

Sadlier, David A.

2002-01-01

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

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

Reduction in time-to-sleep through EEG based brain state detection and audio stimulation.

Science.gov (United States)

Zhuo Zhang; Cuntai Guan; Ti Eu Chan; Juanhong Yu; Aung Aung Phyo Wai; Chuanchu Wang; Haihong Zhang

2015-08-01

We developed an EEG- and audio-based sleep sensing and enhancing system, called iSleep (interactive Sleep enhancement apparatus). The system adopts a closed-loop approach which optimizes the audio recording selection based on user's sleep status detected through our online EEG computing algorithm. The iSleep prototype comprises two major parts: 1) a sleeping mask integrated with a single channel EEG electrode and amplifier, a pair of stereo earphones and a microcontroller with wireless circuit for control and data streaming; 2) a mobile app to receive EEG signals for online sleep monitoring and audio playback control. In this study we attempt to validate our hypothesis that appropriate audio stimulation in relation to brain state can induce faster onset of sleep and improve the quality of a nap. We conduct experiments on 28 healthy subjects, each undergoing two nap sessions - one with a quiet background and one with our audio-stimulation. We compare the time-to-sleep in both sessions between two groups of subjects, e.g., fast and slow sleep onset groups. The p-value obtained from Wilcoxon Signed Rank Test is 1.22e-04 for slow onset group, which demonstrates that iSleep can significantly reduce the time-to-sleep for people with difficulty in falling sleep.

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

Science.gov (United States)

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

2009-07-01

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

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

OpenAIRE

Dewey, Christopher; Wakefield, Jonathan P.

2017-01-01

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

Computationally efficient clustering of audio-visual meeting data

NARCIS (Netherlands)

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

2010-01-01

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

Multi-band microwave photonic satellite repeater scheme employing intensity Mach-Zehnder modulators

Institute of Scientific and Technical Information of China (English)

Yin Jie; Dong Tao; Zhang Bin; Hao Yan; Cao Guixing; Cheng Zijing; Xu Kun; Zhou Yue; Dai Jian

2017-01-01

To solve the satellite repeater's flexible and wideband frequency conversion problem,we propose a novel microwave photonic repeater system,which can convert the upload signal's carrier to six different frequencies.The scheme employs one 20 GHz bandwidth dual-drive Mach-Zehnder modulator (MZM) and two 10 GHz bandwidth MZMs.The basic principle of this scheme is filtering out two optical sidebands after the optical carrier suppression (OCS) modulation and combining two sidebands modulated by the input radio frequency (RF) signal.This structure can realize simultaneous multi-band frequency conversion with only one frequency-fixed microwave source and prevent generating harmful interference sidebands by using two corresponding optical filters after optical modulation.In the simulation,one C-band signal of 6 GHz carrier can be successfully converted to 12 GHz (Ku-band),28 GHz,34 GHz,40 GHz,46 GHz (Ka-band) and 52 GHz (V-band),which can be an attractive method to realize multi-band microwave photonic satellite repeater.Alternatively,the scheme can be configured to generate multi-band local oscillators (LOs) for widely satellite onboard clock distribution when the input RF signal is replaced by the internal clock source.

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.

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

Directory of Open Access Journals (Sweden)

Mansoor Hyder

2013-07-01

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

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

International Nuclear Information System (INIS)

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

2013-01-01

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

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...

Enhancing Charge Carrier Lifetime in Metal Oxide Photoelectrodes through Mild Hydrogen Treatment

KAUST Repository

Jang, Ji-Wook

2017-08-25

Widespread application of solar water splitting for energy conversion is largely dependent on the progress in developing not only efficient but also cheap and scalable photoelectrodes. Metal oxides, which can be deposited with scalable techniques and are relatively cheap, are particularly interesting, but high efficiency is still hindered by the poor carrier transport properties (i.e., carrier mobility and lifetime). Here, a mild hydrogen treatment is introduced to bismuth vanadate (BiVO4), which is one of the most promising metal oxide photoelectrodes, as a method to overcome the carrier transport limitations. Time-resolved microwave and terahertz conductivity measurements reveal more than twofold enhancement of the carrier lifetime for the hydrogen-treated BiVO4, without significantly affecting the carrier mobility. This is in contrast to the case of tungsten-doped BiVO4, although hydrogen is also a donor type dopant in BiVO4. The enhancement in carrier lifetime is found to be caused by significant reduction of trap-assisted recombination, either via passivation or reduction of deep trap states related to vanadium antisite on bismuth or vanadium interstitials according to density functional theory calculations. Overall, these findings provide further insights on the interplay between defect modulation and carrier transport in metal oxides, which benefit the development of low-cost, highly-efficient solar energy conversion devices.

Computationally Efficient Clustering of Audio-Visual Meeting Data

Science.gov (United States)

Hung, Hayley; Friedland, Gerald; Yeo, Chuohao

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

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)

Towards CMOS-compatible nanophotonics: Ultra-compact modulators using alternative plasmonic materials

DEFF Research Database (Denmark)

Babicheva, Viktoriia; Kinsey, Nathaniel; Naik, Gururaj V.

2013-01-01

We propose several planar layouts of ultra-compact plasmonic modulators that utilize alternative plasmonic materials such as transparent conducting oxides and titanium nitride. The modulation is achieved by tuning the carrier concentration in a transparent conducting oxide layer into and out of t...

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

Science.gov (United States)

Rasi, Päivi; Vuojärvi, Hanna

2018-01-01

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

Cortical Integration of Audio-Visual Information

Science.gov (United States)

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

2013-01-01

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

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.

Compensation of the ux modulation distortion using an additional coil in a loudspeaker unit

DEFF Research Database (Denmark)

Antonello, Niccoló; Agerkvist, Finn T.

2014-01-01

Flux modulation is one of the main causes of distortion in electrodynamic loudspeaker units. A new com- pensation technique that eliminates this type of non-linearity using an additional compensation coil in the speaker unit is presented. An equivalent circuit model of the device including...... the compensation coil is de- rived. The compensation technique consists on feeding the compensation coil and voice coil with ltered versions of the wanted audio signal. Simulations show that a signicant reduction in ux modulation distor- tion can be achieved with this technique. A simple magnetic circuit has been...

Compensation of the flux modulation distortion using an additional coil in a loudspeaker unit

DEFF Research Database (Denmark)

Antonello, Niccoló; Agerkvist, Finn T.

2014-01-01

the compensation coil is derived. The compensation technique consists on feeding the compensation coil and voice coil with filtered versions of the wanted audio signal. Simulations show that a significant reduction in flux modulation distortion can be achieved with this technique. A simple magnetic circuit has......Flux modulation is one of the main causes of distortion in electrodynamic loudspeaker units. A new compensation technique that eliminates this type of non-linearity using an additional compensation coil in the speaker unit is presented. An equivalent circuit model of the device including...

Audio-Visual Aid in Teaching "Fatty Liver"