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Sample records for carrier modulation audio

  1. Multi Carrier Modulation Audio Power Amplifier with Programmable Logic

    DEFF Research Database (Denmark)

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

    2009-01-01

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

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

    DEFF Research Database (Denmark)

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

    2008-01-01

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

  3. A Dither Modulation Audio Watermarking Algorithm Based on HAS

    Directory of Open Access Journals (Sweden)

    Yi-bo Huang

    2012-11-01

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

  4. Lossless Audio Watermarking Based on the Alpha Statistic Modulation

    Directory of Open Access Journals (Sweden)

    Sunita V. Dhavale

    2012-09-01

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

  5. A novel audio watermarking scheme using multiscale wavelet modulation

    Institute of Scientific and Technical Information of China (English)

    JI Bing; ZHANG De; JI Xiaoyong

    2004-01-01

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

  6. Amplitude Modulated Sinusoidal Signal Decomposition for Audio Coding

    DEFF Research Database (Denmark)

    Christensen, M. G.; Jacobson, A.; Andersen, S. V.;

    2006-01-01

    In this paper, we present a decomposition for sinusoidal coding of audio, based on an amplitude modulation of sinusoids via a linear combination of arbitrary basis vectors. The proposed method, which incorporates a perceptual distortion measure, is based on a relaxation of a nonlinear least-squar......-squares minimization. Rate-distortion curves and listening tests show that, compared to a constant-amplitude sinusoidal coder, the proposed decomposition offers perceptually significant improvements in critical transient signals....

  7. Carrier Distortion in Hysteretic Self-Oscillating Class-D Audio Power

    DEFF Research Database (Denmark)

    Høyerby, Mikkel Christian Kofod; Andersen, Michael A. E.

    2009-01-01

    and output is required, such as in audio power amplifiers or xDSL drivers. From an average-mode point of view, carrier distortion is shown to be caused by nonlinear variation of the hysteretic comparator input average voltage with the output average voltage. This easily causes total harmonic distortion.......03% THD from 100 Hz to 6.7 kHz. Carrier distortion is shown to account for this limitation in THD performance....

  8. Modeling of Carrier Dynamics in Electroabsorption Modulators

    DEFF Research Database (Denmark)

    Højfeldt, Sune

    2002-01-01

    and a phenomenological model for the carrier sweep-out dynamics, we investigate all-optical wavelength conversion, all-optical signal regeneration, and all-optical demultiplexing. A detailed drift-diffusion type model for the sweerp-out of photo-excited carriers in electroabsorption modulators is presented. We use...... the model to calclulate absorption spectra and steady-state carrier distributions in different modulator structures. This allows us to investigate a number of important properties of electroabsorption modulators, such as the electroabsorption effect and th saturation properties. We also investigate...... the influence that carrier recapture has on the device properties, and we discuss the recapture process on a more fundamental level. The model is also used to investigate in detail the carrier sweep-out process in electroabsorption modulators. We investigate how the intrinsic-region width, the separate...

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

    Directory of Open Access Journals (Sweden)

    Sunita V. Dhavale

    2014-02-01

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

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

    Energy Technology Data Exchange (ETDEWEB)

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

    2005-07-01

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

  11. Concept Framework for Audio Information Retrieval: ARF

    Institute of Scientific and Technical Information of China (English)

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

    2003-01-01

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

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

    CERN Document Server

    Khan, Muhammad Fahad

    2012-01-01

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

  13. 即时通讯音频信道载波传输研究%Instant Messaging Audio Channel Carrier Transmission

    Institute of Scientific and Technical Information of China (English)

    李建文; 牛永钢

    2013-01-01

    即时通讯工具在带给公众信息交流便利的同时也成为信息泄露的重要源头。为了提高信息的安全性,采用以即时通讯工具为平台,利用其音频信道进行数字载波信号传送,将相对重要的信息调制到载波上进行发送,接收时再通过解调完成信息的接收。实验中调制和解调采用了二进制差分相移键控方式。将二进制码元信息调制后在腾讯QQ的音频信道中进行传输,接收端成功接收并解调还原出码元信息。通过实验证明了该方法的可行性。%Instant messaging tool is an important source of information leakage, while bringing much con-venience to the public for their information exchange. In order to improve the security of information and with instant communication tool as the platform, the audio channel is used to transmit digital carrier signal, the relatively important information modulated to the carrier for transmission, and the receiving of complete information finished through demodulation. Experiments of modulation and demodulation is done by using the binary differential phase shift keying method. After meta information modulation, the binary code is transmitted in Tencent QQ audio channel, while in the receiving end, the binary code is successfully re-ceived and demodulated. The experiment indicates the feasibility of this method.

  14. Spread Spectrum Modulation by Using Asymmetric-Carrier Random PWM

    DEFF Research Database (Denmark)

    Mathe, Laszlo; Lungeanu, Florin; Sera, Dezso;

    2012-01-01

    This paper presents a new fixed carrier frequency random PWM method, where a new type of carrier wave is proposed for modulation. Based on simulations and experimental measurements, it is shown that the spread effect of the discrete components from the motor current spectra and acoustic spectra i...

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

    DEFF Research Database (Denmark)

    Pracný, Peter; Bruun, Erik

    2012-01-01

    This paper deals with a system-level design of a digital sigma-delta (∑∆) modulator for hearing-aid audio Class D output stage application. The aim of this paper is to provide a thorough discussion on various possibilities and tradeoffs of ∑∆ modulator system-level design parameter combinations...

  16. A High Dynamic Range and Low Power Consumption Audio Delta-Sigma Modulator with Opamp Sharing Technique among Three Integrators

    Science.gov (United States)

    Kanemoto, Daisuke; Ido, Toru; Taniguchi, Kenji

    A low power and high performance with third order delta-sigma modulator for audio applications, fabricated in a 0.18µm CMOS process, is presented. The modulator utilizes a third order noise shaping with only one opamp by using an opamp sharing technique. The opamp sharing among three integrator stages is achieved through the optimal operation timing, which makes use of the load capacitance differences between the three integrator stages. The designed modulator achieves 101.1dB signal-to-noise ratio (A-weighted) and 101.5dB dynamic range (A-weighted) with 7.5mW power consumption from a 3.3V supply. The die area is 1.27mm2. The fabricated delta-sigma modulator achieves the highest figure-of-merit among published high performance low power audio delta-sigma modulators.

  17. Modeling of carrier dynamics in quantum-well electroabsorption modulators

    DEFF Research Database (Denmark)

    Højfeldt, Sune; Mørk, Jesper

    2002-01-01

    We present a comprehensive drift-diffusion-type electroabsorption modulator (EAM) model. The model allows us to investigate both steady-state properties and to follow the sweep-out of carriers after pulsed optical excitation. Furthermore, it allows for the investigation of the influence that vari......We present a comprehensive drift-diffusion-type electroabsorption modulator (EAM) model. The model allows us to investigate both steady-state properties and to follow the sweep-out of carriers after pulsed optical excitation. Furthermore, it allows for the investigation of the influence...... that various design parameters have on the device properties, in particular how they affect the carrier dynamics and the corresponding field dynamics. A number of different types of results are presented. We calculate absorption spectra and steady-state field screening due to carrier pile-up at the separate......-confinement heterobarriers. We then move on to look at carrier sweep-out upon short-pulse optical excitation. For a structure with one well, we analyze how the well position affects the carrier sweep-out and the absorption recovery. We calculate the field dynamics in a multiquantum-well structure and discuss how the changes...

  18. Carrier suppression in quadruple frequency modulation by cascaded optical external modulators for millimeter-wave generation

    Institute of Scientific and Technical Information of China (English)

    Xue Feng; Wei Zhang; Xiaoming Liu

    2009-01-01

    The optical carrier suppression in optical quadruple frequency modulation by cascaded external modulators is investigated theoretically and experimentally. Theoretical analysis demonstrates that the optical carrier suppression ratio is related with not only the initial phase difference of electrical signals applied on the two modulators, but also the optical phase shift between the two modulators. The maximum suppression ratio can be achieved when the total phase difference is equal to nπ+π/2(n=1,2…),which is verified by experiments. By properly controlling the total phase shift, 40-GHz millimeter-wave is generated by using a 10-GHz radio frequency (RF) source and the modulators.

  19. DESIGN OF WAVELET PACKET BASED MODEL FOR MULTI CARRIER MODULATION

    Directory of Open Access Journals (Sweden)

    MIHIR NARAYAN MOHANTY

    2012-04-01

    Full Text Available In current scenario Multi-Carrier modulation (MCM is considered an effective technique for both wire and wireless communications. It divides the entire bandwidth into several parallel sub-channels. This splitting is by dividing the transmit data into several parallel low-bit-rate data streams and then to modulate the carrierscorresponding to those sub-channels. Though MCM technique uses Orthogonal Frequency Division Multiplexing (OFDM model, it is very sensitive to Carrier Frequency Offset (CFO, that leads to a severedistortion in subcarrier orthogonality and causes inter channel interference (ICI. In this paper, Wavelet Packet Transform is designed for the model of MCM as a novel alternative to the most exiting Orthogonal Frequency Division Multiplexing (OFDM technique, because of its time frequency representation and lower side lobes intransmitted signals, that reduces inter carrier interference (ICI, and inter symbol interference (ISI. Performance analysis is investigated for such model. Simulation results show a significant enhancement in terms of spectral efficiency.

  20. Digital Carrier Modulation and Sampling Issues of Matrix Converters

    DEFF Research Database (Denmark)

    Loh, Poh Chiang; Rong, Runjie; Blaabjerg, Frede

    2009-01-01

    digital carrier modulation schemes for controlling conventional (direct) and indirect matrix converters with minimized semiconductor commutation count and smooth sextant transitions with no erroneous states produced. For guaranteeing the latter two features, correct digital sampling instants and state...... sequence reversal must be chosen appropriately, as demonstrated in the paper for the two different topological options,which, to date, have not yet been discussed in the existing literature. To validate the concepts discussed, experimental testing on the implemented conventional and indirect matrix...

  1. Digital carrier modulation and sampling issues of matrix converters

    DEFF Research Database (Denmark)

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

    2008-01-01

    digital carrier modulation schemes for controlling conventional and sparse matrix converters with minimized semiconductor commutation count and smooth sextant transitions with no erroneous states produced. For guaranteeing the latter two features, correct digital sampling instants and state sequence...... reversal must be chosen appropriately, as demonstrated in the paper for the two different topological options, which to date, have not yet been discussed in the existing literature. To validate the concepts discussed, experimental testing on the implemented conventional and sparse matrix laboratory...

  2. Multi-carrier Modulation (MCM) Signaling using OFDM Technique

    Science.gov (United States)

    Umadevi, H.; Gowda, Chandrakanth H.; Gurumurthy, K. S.

    2011-12-01

    OFDM is novel multicarrier modulation (MCM) technique. It has strong advantage of being a generic transmission scheme whose actual characteristics can be widely customized to fulfill several requirements and constraints of an advanced communication system. It adopts wavelet packet function as carriers which have the characteristic of good orthogonality and time-frequency localization. It can be seen from both theoretical analysis and software simulation that multi-carrier modulation and demodulation technique based on wavelet packet transform has unique advantage and great potential in improving the performance of communication system. This paper demonstrates the operation of a Wavelet Packet based multi-carrier modulation (WP-MCM) scheme. The wavelet packets are derived from multistage tree-structured paraunitary filter banks by choosing the right tree structure which would minimize the bit error between the desired and received signal for a particular channel condition. The performance of the system is simulated and analyzed for the AWGN channel. Through simulation results, we demonstrate the efficacy and the flexibility of the proposed wavelet packet based mechanism. The Bit Error rate (BER) performance is shown to be comparable, and even at times better, to conventional Fourier based OFDM. Comparison of different family of wavelets has been carried out and Meyer wavelet seems to be the most suitable wavelet through simulation results.

  3. Chaotic carrier pulse position modulation communication system and method

    Science.gov (United States)

    Abarbanel, Henry D. I.; Larson, Lawrence E.; Rulkov, Nikolai F.; Sushchik, Mikhail M.; Tsimring, Lev S.; Volkovskii, Alexander R.

    2001-01-01

    A chaotic carrier pulse position modulation communication system and method is disclosed. The system includes a transmitter and receiver having matched chaotic pulse regenerators. The chaotic pulse regenerator in the receiver produces a synchronized replica of a chaotic pulse train generated by the regenerator in the transmitter. The pulse train from the transmitter can therefore act as a carrier signal. Data is encoded by the transmitter through selectively altering the interpulse timing between pulses in the chaotic pulse train. The altered pulse train is transmitted as a pulse signal. The receiver can detect whether a particular interpulse interval in the pulse signal has been altered by reference to the synchronized replica it generates, and can therefore detect the data transmitted by the receiver. Preferably, the receiver predicts the earliest moment in time it can expect a next pulse after observation of at least two consecutive pulses. It then decodes the pulse signal beginning at a short time before expected arrival of a pulse.

  4. 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.

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

    Science.gov (United States)

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

    2013-01-01

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

  6. 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...

  7. A 1.1 mW 87 dB dynamic range {Delta} {sigma} modulator for audio applications

    Energy Technology Data Exchange (ETDEWEB)

    Liu Liyuan; Wang Zhihua; Wei Shaojun [Institute of Microelectronics, Tsinghua University, Beijing 100084 (China); Chen Liangdong; Li Dongmei, E-mail: lidmei@tsinghua.edu.c [Department of Electronic Engineering, Tsinghua University, Beijing 100084 (China)

    2010-05-15

    This paper presents a 1.1 mW 87 dB dynamic range third order {Delta} {sigma} modulator implemented in 0.18 {mu}m CMOS technology for audio applications. By adopting a feed-forward multi-bit topology, the signal swing at the output of the first integrator can be suppressed. A simple current mirror single stage OTA with 34 dB DC gain working under 1 V power supply is used in the first integrator. The prototype modulator achieves 87 dB DR and 83.8 dB peak SNDR across the bandwidth from 100 Hz to 24 kHz with 3 kHz input signal. (semiconductor integrated circuits)

  8. Ferroelectrically driven spatial carrier density modulation in graphene.

    Science.gov (United States)

    Baeumer, Christoph; Saldana-Greco, Diomedes; Martirez, John Mark P; Rappe, Andrew M; Shim, Moonsub; Martin, Lane W

    2015-01-22

    The next technological leap forward will be enabled by new materials and inventive means of manipulating them. Among the array of candidate materials, graphene has garnered much attention; however, due to the absence of a semiconducting gap, the realization of graphene-based devices often requires complex processing and design. Spatially controlled local potentials, for example, achieved through lithographically defined split-gate configurations, present a possible route to take advantage of this exciting two-dimensional material. Here we demonstrate carrier density modulation in graphene through coupling to an adjacent ferroelectric polarization to create spatially defined potential steps at 180°-domain walls rather than fabrication of local gate electrodes. Periodic arrays of p-i junctions are demonstrated in air (gate tunable to p-n junctions) and density functional theory reveals that the origin of the potential steps is a complex interplay between polarization, chemistry, and defect structures in the graphene/ferroelectric couple.

  9. A Versatile Carrier Board and Associated Timer Module Applications

    Energy Technology Data Exchange (ETDEWEB)

    Richard Evans; Albert Grippo; Kevin Jordan

    2005-10-10

    Because of a need for fast prototyping due to frequent upgrades and a large variety of users at the Jefferson Lab Free Electron Laser Facility (FEL), a 4-slot carrier board was designed along the lines of the Industry Pack carriers, with a 6U4HP form factor. In one slot is an Altera Field Programmable Gate Array (FPGA) which communicates with an Experimental Physics and Industrial Control Software (EPICS) interface through the main bus, in the immediate case VME, but potentially PCI and others. The FPGA also provides 64 bits of digital i/o and communicates with a local bus that covers the other 3 slots. In these 3 slots may be placed a variety of daughter boards that may be of a size to span 1, 2 or even all 3 spare slots, depending on the job it has to do. Among the uses implemented in the FEL were a simple digital i/o board (only the first slot occupied), and a timer module occupying 2 slots.

  10. Modulation of visual responses in the superior temporal sulcus by audio-visual congruency.

    Science.gov (United States)

    Dahl, Christoph D; Logothetis, Nikos K; Kayser, Christoph

    2010-01-01

    Our ability to identify or recognize visual objects is often enhanced by evidence provided by other sensory modalities. Yet, where and how visual object processing benefits from the information received by the other senses remains unclear. One candidate region is the temporal lobe, which features neural representations of visual objects, and in which previous studies have provided evidence for multisensory influences on neural responses. In the present study we directly tested whether visual representations in the lower bank of the superior temporal sulcus (STS) benefit from acoustic information. To this end, we recorded neural responses in alert monkeys passively watching audio-visual scenes, and quantified the impact of simultaneously presented sounds on responses elicited by the presentation of naturalistic visual scenes. Using methods of stimulus decoding and information theory, we then asked whether the responses of STS neurons become more reliable and informative in multisensory contexts. Our results demonstrate that STS neurons are indeed sensitive to the modality composition of the sensory stimulus. Importantly, information provided by STS neurons' responses about the particular visual stimulus being presented was highest during congruent audio-visual and unimodal visual stimulation, but was reduced during incongruent bimodal stimulation. Together, these findings demonstrate that higher visual representations in the STS not only convey information about the visual input but also depend on the acoustic context of a visual scene.

  11. Modulation of visual responses in the superior temporal sulcus by audio-visual congruency

    Directory of Open Access Journals (Sweden)

    Christoph Dahl

    2010-04-01

    Full Text Available Our ability to identify or recognize visual objects is often enhanced by evidence provided by other sensory modalities. Yet, where and how visual object processing benefits from the information received by the other senses remains unclear. One candidate region is the temporal lobe, which features neural representations of visual objects, and in which previous studies have provided evidence for multisensory influences on neural responses. In the present study we directly tested whether visual representations in the lower bank of the superior temporal sulcus (STS benefit from acoustic information. To this end, we recorded neural responses in alert monkeys passively watching audio-visual scenes, and quantified the impact of simultaneously presented sounds on responses elicited by the presentation of naturalistic visual scenes. Using methods of stimulus decoding and information theory, we then asked whether the responses of STS neurons become more reliable and informative in multisensory contexts. Our results demonstrate that STS neurons are indeed sensitive to the modality composition of the sensory stimulus. Importantly, information provided by STS neurons’ responses about the particular visual stimulus being presented was highest during congruent audio-visual and unimodal visual stimulation, but was reduced during incongruent bimodal stimulation. Together, these findings demonstrate that higher visual representations in the STS not only convey information about the visual input but also depend on the acoustic context of a visual scene.

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

    Directory of Open Access Journals (Sweden)

    Parag Agarwal

    2008-12-01

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

  13. A Audio Watermarking Algorithm Based on Chaotic-Modulation and Dual-Transform-Domain%一种基于混沌调制和双变换域的音频水印算法

    Institute of Scientific and Technical Information of China (English)

    梁浩; 王国明; 岳雨俭

    2014-01-01

    An improved algorithm is proposed directing against the non-blind extraction of the traditional audio watermarking and watermark’s weak security and robustness. Using Logistic chaotic sequence to modulate the watermark signal after reducing the dimensionality,Combining with the multi-resolution capabilities of discrete wavelet transform and the energy-gathered fea-tures of discrete cosine transform, watermark information is embedded in the audio carrier by modifying the intermediate or low frequency coefficient in dual-transform-domain. The simulation results show that improved algorithm owns better safety and ro-bustness performance.%针对传统音频水印算法不能实现盲提取和水印的安全性、鲁棒性不强,提出一种改进算法。采用Logistic混沌序列对降维后的水印信号作进行调制,再结合离散小波变换的多层分辨能力和离散余弦变换的能量汇聚特性,通过修改双变换域的中低频系数,在载体音频中嵌入水印信息。仿真实验表明改进算法具有更好的安全性和鲁棒性。

  14. 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....

  15. Porous carriers for controlled/modulated drug delivery.

    Science.gov (United States)

    Ahuja, G; Pathak, K

    2009-11-01

    Considerable research efforts have been directed in recent years towards the development of porous carriers as controlled drug delivery matrices because of possessing several features such as stable uniform porous structure, high surface area, tunable pore size and well-defined surface properties. Owing to wide range of useful properties porous carriers have been used in pharmaceuticals for many purposes including development of floating drug delivery systems, sustained drug delivery systems. Various types of pores like open, closed, transport and blind pores in the porous solid allow them to adsorb drugs and release them in a more reproducible and predictable manner. Pharmaceutically exploited porous adsorbents includes, silica (mesoporous), ethylene vinyl acetate (macroporous), polypropylene foam powder (microporous), titanium dioxide (nanoporous). When porous polymeric drug delivery system is placed in contact with appropriate dissolution medium, release of drug to medium must be preceded by the drug dissolution in the water filled pores or from surface and by diffusion through the water filled channels. The porous carriers are used to improve the oral bioavailability of poorly water soluble drugs, to increase the dissolution of relatively insoluble powders and conversion of crystalline state to amorphous state.

  16. Porous carriers for controlled/modulated drug delivery

    Directory of Open Access Journals (Sweden)

    Ahuja G

    2009-01-01

    Full Text Available Considerable research efforts have been directed in recent years towards the development of porous carriers as controlled drug delivery matrices because of possessing several features such as stable uniform porous structure, high surface area, tunable pore size and well-defined surface properties. Owing to wide range of useful properties porous carriers have been used in pharmaceuticals for many purposes including development of floating drug delivery systems, sustained drug delivery systems. Various types of pores like open, closed, transport and blind pores in the porous solid allow them to adsorb drugs and release them in a more reproducible and predictable manner. Pharmaceutically exploited porous adsorbents includes, silica (mesoporous, ethylene vinyl acetate (macroporous, polypropylene foam powder (microporous, titanium dioxide (nanoporous. When porous polymeric drug delivery system is placed in contact with appropriate dissolution medium, release of drug to medium must be preceded by the drug dissolution in the water filled pores or from surface and by diffusion through the water filled channels. The porous carriers are used to improve the oral bioavailability of poorly water soluble drugs, to increase the dissolution of relatively insoluble powders and conversion of crystalline state to amorphous state.

  17. Simultaneous optical carrier and radio frequency re-modulation in radio-over-fiber systems employing reflective SOA modulators

    DEFF Research Database (Denmark)

    Kassar, Carvalho; Calabretta, Nicola; Tafur Monroy, Idelfonso

    2007-01-01

    We demonstrate an innovative full-duplex radio-over-fibre transmission system employing a reflective SOA to perform simultaneous reusing of the optical carrier and data re-modulation, thus avoiding the use of local radiofrequency oscillator at the station sites.......We demonstrate an innovative full-duplex radio-over-fibre transmission system employing a reflective SOA to perform simultaneous reusing of the optical carrier and data re-modulation, thus avoiding the use of local radiofrequency oscillator at the station sites....

  18. Cooperative Communications over Flat Fading Channels with Carrier Offsets: A Double-Differential Modulation Approach

    Directory of Open Access Journals (Sweden)

    Lingyang Song

    2008-05-01

    Full Text Available We propose double-differential (DD modulation for the amplify-and-forward protocol over Nakagami-m fading channels with carrier offsets. We propose an emulated maximum ratio combining (EMRC decoder, which could be used by the double-differential receiver in the absence of exact channel knowledge. Approximate bit error rate (BER analysis is performed for the proposed double-differential modulation-based cooperative communication system. The proposed double-differential system is immune to random carrier offsets, whereas the conventional single-differential modulation-based cooperative system breaks down. In addition, the proposed scheme is able to perform better than the same rate training-based cooperative system which utilizes training data for finding estimates of carrier offsets and channel gains.

  19. A novel Modulation Topology for Power Converters utilizing Multiple Carrier Signals

    DEFF Research Database (Denmark)

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

    2008-01-01

    Power converters are known to generate spectral components in the range of interest of electromagnetic compatibility measurements. Common approaches to manipulate some selected components in these frequency ranges are shown here. These approaches add components to the input signal of the modulator...... to derive a slightly varied spectrum. To achieve a rectangular output signal, those modulators use a triangular or saw tooth carrier signal. A novel family of modulators is shown here, using more than one carrier signal to obtain a completely changed spectrum while maintaining the rectangular shaped...... waveform at the output. The multiple carriers are fed into multiple comparators and their outputs are intelligently combined by logic gates to get a single signal to drive one power stage of any type of converter. This commutation distinguishes between the four members of the novel family: the first one...

  20. Alpha parameter in quantum-dot amplifier under optical and electrical carrier modulation

    DEFF Research Database (Denmark)

    Poel, Mike van der; Birkedal, Dan; Hvam, Jørn Märcher;

    2004-01-01

    Alpha parameter of a long-wavelength quantum-dot amplifier near 1.3 ìm is measured to be below one even with saturated gain. A simple model explains difference in apparent alpha parameter under optical and electrical carrier modulation.......Alpha parameter of a long-wavelength quantum-dot amplifier near 1.3 ìm is measured to be below one even with saturated gain. A simple model explains difference in apparent alpha parameter under optical and electrical carrier modulation....

  1. Multi-Carrier Modulation and MIMO Principle Application on Subscriber Lines

    Directory of Open Access Journals (Sweden)

    J. Vodrazka

    2007-12-01

    Full Text Available The multi-carrier modulation is used in many applications, primary for a wireless transmission, for example Wi-Fi and WiMAX networks or DVB-T. But the same physical principle can be used also for metallic lines in access or local networks, for example ADSL and VDSL. The multi-carrier modulation in these cases is called DMT. The dominant source of noise in multi-pair metallic cables is crosstalk when the information capacity is limited dramatically. However, information capacity of metallic lines can be increased, if the system is using MIMO principles, concrete VDMT modulation and line bounding concept. The methods for VDMT modulation and partial crosstalk cancellation are discussed and simulation results are presented.

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

    Science.gov (United States)

    Messer, Dion D.

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

  3. Performance Evaluation of the New Compound-Carrier-Modulated Signal for Future Navigation Signals

    Directory of Open Access Journals (Sweden)

    Ruidan Luo

    2016-01-01

    Full Text Available Navigation Signal based on Compound Carrier (NSCC, is proposed as the potential future global navigation satellite system (GNSS signal modulation scheme. NSCC, a kind of multi-carrier (MC signal, is generated by superposition and multi-parameter adjustment of sub-carriers. Therefore, a judious choice of parameter configation is needed. The main objective of this paper is to investigate the performance of the NSCC which is influenced by these parameters and to demonstrate its structure characteristics and superiority, employing a comprehensive evaluation system. The results show that the proposed NSCC signal processes full spectral efficiency and limited out of band (OOB emissions, satisfying the demands of crowed frequency resources. It also presents better performance in terms of spectral separation coefficients (SSCs, tracking accuracy, multipath mitigation capability and anti-jamming reduction compared with the legacy navigation signals. NSCC modulation represents a serious candidate for navigation satellite augmentation systems, especially for signals applied in challenging environments.

  4. Impact of Free Carriers on Modulational Instability in Silicon-on-insulator Nanowaveguides

    CERN Document Server

    Chaturvedi, Deepa

    2016-01-01

    We have numerically studied the effect of free-carrier-induced loss and dispersion on the modulational instability (MI) gain at low input powers in silicon-on-insulator (SOI) nanowaveguides with normal and anomalous second-order dispersion. We have shown that the free carriers affect the gain spectra even at low input powers. First time we have reported the gain in normal SOI nanowaveguides even in the absence of higher order dispersion parameters, which is due to the interaction of free-carrier-induced dispersion and nonlinearity. The MI gain in an anomalous SOI nanowaveguide vanishes even at a few milliwatt range of input power due to this interaction. We have shown that the gain could be achieved in an anomalous nanowaveguides by reducing the free carrier lifetime.

  5. 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...

  6. Balancing Audio

    DEFF Research Database (Denmark)

    Walther-Hansen, Mads

    2016-01-01

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

  7. 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

  8. A Novel Technique of Measuring SOA Differential Carrier Lifetime and a -Factor Using SOA Optical Modulation Response

    Institute of Scientific and Technical Information of China (English)

    Ki-Hyuk Lee; Woo-Young Choi

    2003-01-01

    We demonstrate a new technique of measuring differential carrier lifetime and linewidth enhancement factor in a semiconductor optical amplifier. In our method, the optical responses and fiber transfer functions of a self-gain modulated SOA are measured and, from these, values of carrier lifetimes and linewidth enhancement factors are determined for various SOA input optical powers.

  9. Real-time Covert Communications Channel for Audio Signals

    Directory of Open Access Journals (Sweden)

    Ashraf Seleym

    2012-09-01

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

  10. 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.

  11. Depletion-mode carrier-plasma optical modulator in zero-change advanced CMOS.

    Science.gov (United States)

    Shainline, Jeffrey M; Orcutt, Jason S; Wade, Mark T; Nammari, Kareem; Moss, Benjamin; Georgas, Michael; Sun, Chen; Ram, Rajeev J; Stojanović, Vladimir; Popović, Miloš A

    2013-08-01

    We demonstrate the first (to the best of our knowledge) depletion-mode carrier-plasma optical modulator fabricated in a standard advanced complementary metal-oxide-semiconductor (CMOS) logic process (45 nm node SOI CMOS) with no process modifications. The zero-change CMOS photonics approach enables this device to be monolithically integrated into state-of-the-art microprocessors and advanced electronics. Because these processes support lateral p-n junctions but not efficient ridge waveguides, we accommodate these constraints with a new type of resonant modulator. It is based on a hybrid microring/disk cavity formed entirely in the sub-90 nm thick monocrystalline silicon transistor body layer. Electrical contact of both polarities is made along the inner radius of the multimode ring cavity via an array of silicon spokes. The spokes connect to p and n regions formed using transistor well implants, which form radially extending lateral junctions that provide index modulation. We show 5 Gbps data modulation at 1265 nm wavelength with 5.2 dB extinction ratio and an estimated 40 fJ/bit energy consumption. Broad thermal tuning is demonstrated across 3.2 THz (18 nm) with an efficiency of 291 GHz/mW. A single postprocessing step to remove the silicon handle wafer was necessary to support low-loss optical confinement in the device layer. This modulator is an important step toward monolithically integrated CMOS photonic interconnects.

  12. Design on Audio Signaling Processing Module Based on DSP%基于DSP的音频信令处理模块实现

    Institute of Scientific and Technical Information of China (English)

    张迎春; 倪永婧

    2012-01-01

    音频信令处理是话音通信不可缺少的处理环节,基于DSP的音频信令处理模块应用广泛。介绍了信令处理平台的硬件设计、软件设计,并提出了A律解码、数字滤波、双音多频/信号音译码、双音多频发生器、信号音发生器及2FSK发生器等各项需要解决的问题,并阐述了MCBSPs的工作条件和接口收发的时序关系。随着电话交换设备信令音的频率偏差,针对不同应用环境提出相应的解决方案。%The audio signaling processing is the essential part of voice communication, and the audio signaling processing module based on DSP has been widely applied. At the beginning of this paper, the designs of software and hardware for signaling processing platform are introduced, and then we present the problems to be solved in A-law decoding, digital filtering, dual tone multi frequency/ signal tone decoding (DTMF/STD) , dual tone multi frequency generator, signal tone generator, 2FSK generator, etc. The operation condition of McBSPs and the temporal relation of interface sending and receiving are also illustrated. With the variation of frequency offset of signal tone of telephone switch, this paper proposes the methods to improve the performance in different application environment.

  13. 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...... 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...

  14. Nanoscale Tunable Strong Carrier Density Modulation of 2D Materials for Metamaterials and Other Tunable Optoelectronics

    Science.gov (United States)

    Peng, Cheng; Efetov, Dmitri; Shiue, Ren-Jye; Nanot, Sebastien; Hempel, Marek; Kong, Jing; Koppens, Frank; Englund, Dirk

    Strong spatial tunability of the charge carrier density at nanoscale is essential to many 2D-material-based electronic and optoelectronic applications. As an example, plasmonic metamaterials with nanoscale dimensions would make graphene plasmonics at visible and near-infrared wavelengths possible. However, existing gating techniques based on conventional dielectric gating geometries limit the spatial resolution and achievable carrier concentration, strongly restricting the available wavelength, geometry, and quality of the devices. Here, we present a novel spatially selective electrolyte gating approach that allows for in-plane spatial Fermi energy modulation of 2D materials of more than 1 eV (carrier density of n = 1014 cm-2) across a length of 2 nm. We present electrostatic simulations as well as electronic transport, photocurrent, cyclic voltammetry and optical spectroscopy measurements to characterize the performance of the gating technique applied to graphene devices. The high spatial resolution, high doping capacity, full tunability and self-aligned device geometry of the presented technique opens a new venue for nanoscale metamaterial engineering of 2D materials for complete optical absorption, nonlinear optics and sensing, among other applications.

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

    Energy Technology Data Exchange (ETDEWEB)

    Kierkegaard, P.

    1998-10-01

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

  16. Intelligent audio analysis

    CERN Document Server

    Schuller, Björn W

    2013-01-01

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

  17. Harmonics Reduction of Multilevel Inverter Drive Using Sine Carrier Pulse Width Modulation Techniques

    Directory of Open Access Journals (Sweden)

    S. Ebanezar Pravin

    2016-11-01

    Full Text Available The main objective of this paper is to control the speed of an induction motor by using seven level diode clamped multilevel inverter and improve the high quality sinusoidal output voltage with reduced harmonics. The presented scheme for diode clamped multilevel inverter is sine carrier Pulse Width Modulation control. An open loop speed control can be achieved by using V/ƒ method. This method can be implemented by changing the supply voltage and frequency applied to the three phase induction motor at constant ratio. The presented system is an effective replacement for the conventional method which has high switching losses, its result ends in a poor drive performance. The simulation result portrays the effective control in the motor speed and an enhanced drive performance through reduction in total harmonic distortion (THD. The effectiveness of the system is verified through simulation using PSIM6.1 Simulink package.

  18. Multiperiodicity, modulations and flip-flops in variable star light curves I. Carrier fit method

    CERN Document Server

    Pelt, J; Mantere, M J; Tuominen, I

    2011-01-01

    The light curves of variable stars are commonly described using simple trigonometric models, that make use of the assumption that the model parameters are constant in time. This assumption, however, is often violated, and consequently, time series models with components that vary slowly in time are of great interest. In this paper we introduce a class of data analysis and visualization methods which can be applied in many different contexts of variable star research, for example spotted stars, variables showing the Blazhko effect, and the spin-down of rapid rotators. The methods proposed are of explorative type, and can be of significant aid when performing a more thorough data analysis and interpretation with a more conventional method.Our methods are based on a straightforward decomposition of the input time series into a fast "clocking" periodicity and smooth modulating curves. The fast frequency, referred to as the carrier frequency, can be obtained from earlier observations (for instance in the case of p...

  19. Audio Indexing for Efficiency

    Science.gov (United States)

    Rahnlom, Harold F.; Pedrick, Lillian

    1978-01-01

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

  20. 基于模型的数字音频广播信号调制系统设计%Design of Digital Audio Broadcasting Signal Modulation System Based on Model

    Institute of Scientific and Technical Information of China (English)

    陆探; 张徐亮; 朱伟杰; 朱万经

    2012-01-01

    本文设计并在FPGA芯片中实现了数字音频广播系统的信号调制系统。信号调制系统位于整个数字音频广潘系统基带信号处理链的末端,是基带数字信号处理的核心系统。根据Eureka147标准,信号调制系统需要对输入的基带码流进行数字调制、频域交织、差分调制以及正交频分复用等一系列处理。所设计的信号调制系统能够对输入的基带码流进行实时处理,完成上述信号处理算法,并输出数字音频广播的基带信号。%A signal modulation system of digital audio broadcasting system is designed and implemented in FPGA. Signal modulation system which is at the end of digital audio broadcasting baseband signal processing chain, is the core system of baseband digital signal processing. According to the Eureka 147 standard, signal modulation system completes digital modulation, frequency interleave, differential modulation and orthogonal frequency division multiplexing of the input baseband stream. The signal modulation system can process input baseband stream in real time, completes above signal processing algorithms and outputs digital audio broadcasting baseband signals.

  1. Audio Steganography Techniques-A Survey

    Directory of Open Access Journals (Sweden)

    Navneet Kaur

    2014-06-01

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

  2. 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

  3. 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.

  4. 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.

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

    Institute of Scientific and Technical Information of China (English)

    黄俊杰; 黄时俊; 黄丹阳; 陈仲本

    2012-01-01

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

  6. Ferroelectric-induced carrier modulation for ambipolar transition metal dichalcogenide transistors

    Science.gov (United States)

    Yin, Lei; Wang, Zhenxing; Wang, Feng; Xu, Kai; Cheng, Ruiqing; Wen, Yao; Li, Jie; He, Jun

    2017-03-01

    For multifarious electronic and optoelectronic applications, it is indispensable exploration of stable and simple method to modulate electrical behavior of transition metal dichalcogenides (TMDs). In this study, an effective method to adjust the electrical properties of ambipolar TMDs is developed by introducing the dipole electric field from poly(vinylidene fluoride-trifluoroethylene) (P(VDF-TrFE)) ferroelectric polymer. The transition from ambipolar to p-type conductive characteristics is realized, and the transistor performances are also significantly enhanced. Hole density of MoTe2- and WSe2-based back-gate field effect transistors increases by 4.4 and 2.5 times. Moreover, the corresponding hole mobilities are strikingly improved from 0.27 to 10.7 cm2 V-1 s-1 and from 1.6 to 59.8 cm2 V-1 s-1, respectively. After optimizing, p-channel MoTe2 phototransistors present ultrahigh responsivity of 3521 A/W, which is superior to most layered phototransistors. The remarkable control of conductive type, carrier concentration, and field-effect mobility of ambipolar TMDs via P(VDF-TrFE) treatment paves a way for realization of high-performance and versatile electronic and optoelectronic devices.

  7. Principles of Audio Watermarking

    Directory of Open Access Journals (Sweden)

    Martin Hrncar

    2008-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Luca Cossettini

    2009-11-01

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

  9. Modulation of carrier dynamics and threshold characteristics in 1.3-μm quantum dot photonic crystal nanocavity lasers

    Science.gov (United States)

    Xing, Enbo; Tong, Cunzhu; Rong, Jiamin; Shu, Shili; Wu, Hao; Wang, Lijie; Tian, Sicong; Wang, Lijun

    2016-08-01

    A self-consistent all-pathway quantum dot (QD) rate equation model, in which all possible relaxation pathways are considered, is used to investigate the influence of quality (Q) factor on the carrier dynamics of 1.3-μm InAs/GaAs QD photonic crystal (PhC) nanolasers. It is found that Q factor not only affects the photon lifetime, but also modulates the carrier occupation in QDs. About three times increases of carrier injection efficiency in QD ground state can be realized in nanocavity with high Q factor. However, it also reveals that over 90% improvement of threshold current happens when Q factor increases from 2000 to 7000, which means it might be not necessary to pursuit for ultrahigh Q factor for the purpose of low threshold current.

  10. Audio Papers - A Manifesto

    DEFF Research Database (Denmark)

    Krogh Groth, Sanne; Samson, Kristine

    2016-01-01

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

  11. Digital Audio Legal Recorder

    Data.gov (United States)

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

  12. 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.

  13. Robust audio hashing for audio authentication watermarking

    Science.gov (United States)

    Zmudzinski, Sascha; Steinebach, Martin

    2008-02-01

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

  14. The Study of Audio Watermarking

    Institute of Scientific and Technical Information of China (English)

    王景; 唐晟

    2011-01-01

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

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

    Science.gov (United States)

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

    2006-08-01

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

  16. 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...

  17. 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.

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

    DEFF Research Database (Denmark)

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

    2007-01-01

    -frequency domain. Pure-tone carrier amplitude-modulation (AM) depth-discrimination thresholds were found to be similar using both traditional gated stimuli and using a temporally modulated fringe for a fixed standard depth (ms=0.25) and a range of AM frequencies (4-64 Hz). In a second experiment, masked sinusoidal......In general, the temporal structure of stimuli must be considered to account for certain observations made in detection and masking experiments in the audio-frequency domain. Two such phenomena are (1) a heightened sensitivity to amplitude increments with a temporal fringe compared to gated level...

  19. 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.

  20. Enhancement of minority carrier lifetime of GaInP with lateral composition modulation structure grown by molecular beam epitaxy

    Energy Technology Data Exchange (ETDEWEB)

    Park, K. W.; Ravindran, Sooraj; Kang, S. J.; Hwang, H. Y.; Jho, Y. D. [School of Information and Communications, Gwangju Institute of Science and Technology, Gwangju 500-712 (Korea, Republic of); Park, C. Y. [School of Information and Communications, Gwangju Institute of Science and Technology, Gwangju 500-712 (Korea, Republic of); Jo, Y. R.; Kim, B. J. [School of Material Science and Engineering, Gwangju Institute of Science and Technology, Gwangju 500-712 (Korea, Republic of); Lee, Y. T., E-mail: ytlee@gist.ac.kr [School of Information and Communications, Gwangju Institute of Science and Technology, Gwangju 500-712 (Korea, Republic of); Advanced Photonics Research Institute, Gwangju Institute of Science and Technology, Gwangju 500-712 (Korea, Republic of)

    2014-07-28

    We report the enhancement of the minority carrier lifetime of GaInP with a lateral composition modulated (LCM) structure grown using molecular beam epitaxy (MBE). The structural and optical properties of the grown samples are studied by transmission electron microscopy and photoluminescence, which reveal the formation of vertically aligned bright and dark slabs corresponding to Ga-rich and In-rich GaInP regions, respectively, with good crystal quality. With the decrease of V/III ratio during LCM GaInP growth, it is seen that the band gap of LCM GaInP is reduced, while the PL intensity remains high and is comparable to that of bulk GaInP. We also investigate the minority carrier lifetime of LCM structures made with different flux ratios. It is found that the minority carrier lifetime of LCM GaInP is ∼37 times larger than that of bulk GaInP material, due to the spatial separation of electrons and holes by In-rich and Ga-rich regions of the LCM GaInP, respectively. We further demonstrate that the minority carrier lifetime of the grown LCM GaInP structures can easily be tuned by simply adjusting the V/III flux ratio during MBE growth, providing a simple yet powerful technique to tailor the electrical and optical properties at will. The exceptionally high carrier lifetime and the reduced band gap of LCM GaInP make them a highly attractive candidate for forming the top cell of multi-junction solar cells and can enhance their efficiency, and also make them suitable for other optoelectronics devices, such as photodetectors, where longer carrier lifetime is beneficial.

  1. Performance tradeoff between lateral and interdigitated doping patterns for high speed carrier-depletion based silicon modulators.

    Science.gov (United States)

    Yu, Hui; Pantouvaki, Marianna; Van Campenhout, Joris; Korn, Dietmar; Komorowska, Katarzyna; Dumon, Pieter; Li, Yanlu; Verheyen, Peter; Absil, Philippe; Alloatti, Luca; Hillerkuss, David; Leuthold, Juerg; Baets, Roel; Bogaerts, Wim

    2012-06-04

    Carrier-depletion based silicon modulators with lateral and interdigitated PN junctions are compared systematically on the same fabrication platform. The interdigitated diode is shown to outperform the lateral diode in achieving a low VπLπ of 0.62 V∙cm with comparable propagation loss at the expense of a higher depletion capacitance. The low VπLπ of the interdigitated PN junction is employed to demonstrate 10 Gbit/s modulation with 7.5 dB extinction ration from a 500 µm long device whose static insertion loss is 2.8 dB. In addition, up to 40 Gbit/s modulation is demonstrated for a 3 mm long device comprising a lateral diode and a co-designed traveling wave electrode.

  2. Research on Feature Extraction of Composite Pseudocode Phase Modulation-Carrier Frequency Modulation Signal Based on PWD Transform

    Institute of Scientific and Technical Information of China (English)

    LI Ming-zi; ZHAO Hui-chang

    2008-01-01

    The identification features of composite pseudocode phase modulation and carry frequency modulation signal in-clude pseudocode and modulation frequency. In this paper, PWD is used to extract these features. First, the feature of pseudocode is extracted using the amplitude output of PWD and the correlation filter technology. Then the feature of fre-quency modulation is extracted by way of PWD analysis on the signal processed by anti-phase operation according to the extracted feature of pseudo code, i.e. position information of changed abruptly point of phase. The simulation result shows that both the features of frequency modulation and phase change position caused by the pseudocode phase modula-tion can be extracted effectively for SNR = 3 dB.

  3. Experimental demonstration of a digital maximum likelihood based feedforward carrier recovery scheme for phase-modulated radio-over-fibre links

    DEFF Research Database (Denmark)

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

    2008-01-01

    Maximum likelihood based feedforward RF carrier synchronization scheme is proposed for a coherently detected phase-modulated radio-over-fiber link. Error-free demodulation of 100 Mbit/s QPSK modulated signal is experimentally demonstrated after 25 km of fiber transmission.......Maximum likelihood based feedforward RF carrier synchronization scheme is proposed for a coherently detected phase-modulated radio-over-fiber link. Error-free demodulation of 100 Mbit/s QPSK modulated signal is experimentally demonstrated after 25 km of fiber transmission....

  4. 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.

  5. QPSK调制信号的同步载波提取%Synchronous carrier extraction of QPSK modulated signal

    Institute of Scientific and Technical Information of China (English)

    张小莉; 樊延虎

    2015-01-01

    为了在QPSK数字调制系统中恢复QPSK的载波同步信号,以确保接收机能够接收到无失真的数据,在对QPSK调制理论阐述后提出基于DDS的平方环直接提取载波的方法,分析了电路中各个部分的作用和功能,用Matlab软件进行仿真实验,对锁相环提取的载波信号进行分析,在锁相环性能良好的前提下,实现了载波信号的提取。利用Verilog HDL语言对硬件电路进行行为级描述,综合出RTL级电路。%In order to restore the QPSK Carrier synchronization signals in the QPSK digital modulation system and ensure that the receiver can receive data without distortion,QPSK modulation theory is described and the DDS⁃based method that the carrier is directly extracted by the square ring is proposed. The role and function of each part of the circuit are analyzed. Matlab software was used in the simulation experiment to analysis the carrier signal extracted by phase⁃locked loop. In a premise to maintain good performance of the phase⁃locked loop,the extraction of the carrier signal was achieved. The behavioral level de⁃scription of hardware circuits is performed with Verilog HDL. The RTL⁃level circuit was obtained by synthetical method.

  6. Forensic audio watermark detection

    Science.gov (United States)

    Steinebach, Martin; Zmudzinski, Sascha; Petrautzki, Dirk

    2012-03-01

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

  7. Introduction to AVS Audio

    Institute of Scientific and Technical Information of China (English)

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

    2006-01-01

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

  8. 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.

  9. Influence of carrier dynamics on the modulation bandwidth of quantum-dot based nanocavity devices

    DEFF Research Database (Denmark)

    Lorke, Michael; Nielsen, Torben Roland; Mørk, Jesper

    2010-01-01

    We theoretically investigate the modulation response of quantum-dot based nanocavity light emitting devices. For high Purcell enhancement factors, our theory predicts the possibility of decreasing the modulation bandwidth with increasing scattering rate into the lasing quantum-dot state. This cou......We theoretically investigate the modulation response of quantum-dot based nanocavity light emitting devices. For high Purcell enhancement factors, our theory predicts the possibility of decreasing the modulation bandwidth with increasing scattering rate into the lasing quantum-dot state...

  10. Paralleled Phase Shifted Carrier Pulse Width Modulation (PSCPWM) Schemes - A Fundamental Analysis

    DEFF Research Database (Denmark)

    Christensen, Frank Schwartz; Frederiksen, Thomas Mansachs; Nielsen, Karsten

    1999-01-01

    The paper presents a fundamental analysis of modulation schemes and their spectral aspects for a range of powerstage topologies, from a simple 2-level switching leg to more complex multi-level switching topologies. A family of modulation schemes are introduced and the double Fourier series based...

  11. Demonstration of 48-Gb/s 16-QAM signal transmission using half cycle sub-carrier modulation in intensity modulation/direct detection system

    Science.gov (United States)

    Tang, Jin; He, Jing; Chen, Ming; Li, Danyu; Chen, Lin

    2015-01-01

    A simple spectral-efficiency intensity modulation/direct detection (IM/DD) system based on half cycle sub-carrier modulation (SCM) signal is proposed for short reach fiber communications in this paper. The signal impairment of frequency selective fading due to fiber chromatics dispersion (CD) is mathematically analyzed. To reduce the performance deterioration caused by the non-flat transfer function, digital pre- and post-equalization is applied in the system. The peak to average power ratio (PAPR) of the signal is also discussed in comparison with that of orthogonal frequency division multiplexing (OFDM). The transmission of 16-QAM half cycle SCM signal with a sub-carrier frequency of half the symbol rate and Nyquist pulse shaping is experimentally demonstrated. The bit-error rate (BER) of 48 Gb/s polarization multiplexing division (PDM) 16 QAM half cycle SCM signal is less than 7% forward-error-correction (FEC) threshold of 3.8 ×10-3 after transmission over 83 km standard single-mode fiber (SSMF).

  12. 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.

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

    DEFF Research Database (Denmark)

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

    In the audio-frequency domain, the envelope apparently plays an important role in detection of intensity increments and in comodulation masking release (CMR). The current study addressed the question whether the second-order envelope ("venelope") contributes similarly for comparable experiments...... were found to be the same in conditions with a continuous (modulated) carrier and with traditional gated stimuli for AM frequencies ranging from 4 –64 Hz. The second set of experiments compared the amount of CMR in a tone-in-noise detection task when slow, regular fluctuations were imposed...

  14. High-speed carrier-depletion silicon Mach-Zehnder optical modulators with lateral PN junctions

    Directory of Open Access Journals (Sweden)

    Graham Trevor Reed

    2014-12-01

    Full Text Available This paper presents new experimental data from a lateral PN junction silicon Mach-Zehnder optical modulator. Efficiencies in the 1.4V.cm to 1.9V.cm range are demonstrated for drive voltages between 0V and 6V. High speed operation up to 52Gbit/s is also presented. The performance of the device which has its PN junction positioned in the centre of the waveguide is then compared to previously reported data from a lateral PN junction device with the junction self-aligned to the edge of the waveguide rib. An improvement in modulation efficiency is demonstrated when the junction is positioned in the centre of the waveguide. Finally we propose schemes for achieving high modulation efficiency whilst retaining self-aligned formation of the PN junction.

  15. 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)

    Xu Kui

    2009-01-01

    Full Text Available Abstract 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.

  16. 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.

  17. 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

    Matrix converters with three-level phase switching and five-level line switching characteristics have recently been proposed as improved ldquoall semiconductorrdquo ac-ac power processors. For their control, different modulation schemes have also been developed with different researchers claiming...... be implemented on a single common hardware platform....

  18. Embedded Audio Without Beeps

    DEFF Research Database (Denmark)

    Overholt, Daniel; Møbius, Nikolaj Friis

    2014-01-01

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

  19. Editing Audio with Audacity

    Directory of Open Access Journals (Sweden)

    Brandon Walsh

    2016-08-01

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

  20. Multipurpose audio watermarking algorithm

    Institute of Scientific and Technical Information of China (English)

    Ning CHEN; Jie ZHU

    2008-01-01

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

  1. Audio Feedback -- Better Feedback?

    Science.gov (United States)

    Voelkel, Susanne; Mello, Luciane V.

    2014-01-01

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

  2. Categorizing Video Game Audio

    DEFF Research Database (Denmark)

    Westerberg, Andreas Rytter; Schoenau-Fog, Henrik

    2015-01-01

    This paper dives into the subject of video game audio and how it can be categorized in order to deliver a message to a player in the most precise way. A new categorization, with a new take on the diegetic spaces, can be used a tool of inspiration for sound- and game-designers to rethink how they ...

  3. Low-cost bidirectional hybrid fiber-visible laser light communication system based on carrier-less amplitude phase modulation

    Science.gov (United States)

    He, Jing; Dong, Huan; Deng, Rui; Chen, Lin

    2016-08-01

    We propose a bidirectional hybrid fiber-visible laser light communication (fiber-VLC) system. To reduce the cost of the system, the cheap and easy integration red vertical cavity surface emitting lasers, low-complexity carrier-less amplitude phase modulation format, and wavelength reuse technique are utilized. Meanwhile, the automatic gain control amplifier voltage and bias voltage for downlink and uplink are optimized. The simulation results show that, by using the proposed system, the bit error rate of 3.8×10-3 can be achieved for 16-Gbps CAP signal after 30-km standard single mode fiber and 8-m VLC bidirectional transmission. Therefore, it indicates the feasibility and potential of proposed system for indoor access network.

  4. Free-carrier electro-absorption and electro-refraction modulation in group IV materials at mid-infrared wavelengths

    Science.gov (United States)

    Nedeljkovic, Milos; Soref, Richard A.; Mashanovich, Goran Z.

    2012-01-01

    Mid-infrared group IV photonics is emerging as a field with possible applications ranging from gas sensing to free-space communications. Free-carrier induced electro-absorption and electro-refraction have become the most widely used modulation mechanisms in active near-infrared silicon photonic devices. This work examines the magnitude of this effect in group IV materials at mid-infrared wavelengths. In silicon electro-absorption effects are calculated from experimental absorption coefficient data, and electro-refraction is calculated through numerical Kramers-Kronig analysis of absorption spectra. In germanium the Drude-Lorentz equations are used to estimate both change in absorption and change in refractive index.

  5. Voltage-Balancing Method for Modular Multilevel Converters Under Phase-Shifted Carrier-Based Pulsewidth Modulation

    DEFF Research Database (Denmark)

    Deng, Fujin; Chen, Zhe

    2015-01-01

    The modular multilevel converter (MMC) becomes attractive for medium- or high-power applications because of the advantages of high modularity, availability, and power quality. One of the technical challenges associated with an MMC is the balancing of the capacitors' voltages. In this paper......, a voltage-balancing control method is proposed for the MMC under phase-shifted carrier-based pulsewidth modulation. The proposed voltage-balancing method uses the linearization method for pulse sorting without arm current measurement, which can control the capacitor charge transfer to balance the capacitor....../EMTDC are conducted, and a downscale MMC prototype is also tested with the proposed method. The study results show the effectiveness of the proposed voltage-balancing method....

  6. 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

  7. Capacitively-Induced Free-Carrier Effects in Nanoscale Silicon Waveguides for Electro-Optic Modulation

    CERN Document Server

    Sharma, Rajat; Lin, Hung-Hsi; Isichenko, Andrei; Vallini, Felipe; Fainman, Yeshaiahu

    2015-01-01

    We fabricate silicon waveguides in silicon-on-insulator (SOI) wafers clad with either silicon dioxide, silicon nitride, or aluminum oxide, and by measuring the electro-optic behavior of ring resonators, we characterize the cladding-dependent and capacitively-induced free-carrier effects in each of these waveguides. By comparing our measured data with simulation results, we confirm that the observed voltage dependencies of the transmission spectra are due to changes in the concentrations of holes and electrons within the semiconductor waveguide, and we show for the first time how strongly these effects depend on the cladding material which comes into contact with the silicon waveguide. Additionally, the waveguide loss is found to have a particularly high sensitivity to the applied voltage, and may therefore find use in a wide range of applications which require low- or high-loss propagation. Collectively, these phenomena may be incorporated into more complex waveguide designs in the future to create high-effic...

  8. Perancangan Sistem Audio Mobil berbasiskan Sistem Pakar dan Web

    Directory of Open Access Journals (Sweden)

    Djunaidy Santoso

    2011-11-01

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

  9. AC-3 audio coder

    Science.gov (United States)

    Todd, Craig

    1995-12-01

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

  10. Carrier frequency offset estimation for an acoustic-electric channel using 16 QAM modulation

    Science.gov (United States)

    Cunningham, Michael T.; Anderson, Leonard A.; Wilt, Kyle R.; Chakraborty, Soumya; Saulnier, Gary J.; Scarton, Henry A.

    2016-05-01

    Acoustic-electric channels can be used to send data through metallic barriers, enabling communications where electromagnetic signals are ineffective. This paper considers an acoustic-electric channel that is formed by mounting piezoelectric transducers on metallic barriers that are separated by a thin water layer. The transducers are coupled to the barriers using epoxy and the barriers are positioned to axially-align the PZTs, maximizing energy transfer efficiency. The electrical signals are converted by the transmitting transducers into acoustic waves, which propagate through the elastic walls and water medium to the receiving transducers. The reverberation of the acoustic signals in these channels can produce multipath distortion with a significant delay spread that introduces inter-symbol interference (ISI) into the received signal. While the multipath effects can be severe, the channel does not change rapidly which makes equalization easier. Here we implement a 16-QAM system on this channel, including a method for obtaining accurate carrier frequency offset (CFO) estimates in the presence of the quasi-static multipath propagation. A raised-power approach is considered but found to suffer from excessive data noise resulting from the ISI. An alternative approach that utilizes a pilot tone burst at the start of a data packet is used for CFO estimation and found to be effective. The autocorrelation method is used to estimate the frequency of the received burst. A real-time prototype of the 16 QAM system that uses a Texas Instruments MSP430 microcontroller-based transmitter and a personal computer-based receiver is presented along with performance results.

  11. Young adult male carriers of the fragile X premutation exhibit genetically modulated impairments in visuospatial tasks controlled for psychomotor speed

    Directory of Open Access Journals (Sweden)

    Wong Ling M

    2012-11-01

    Full Text Available Abstract Background A previous study reported enhanced psychomotor speed, and subtle but significant cognitive impairments, modulated by age and by mutations in the fragile X mental retardation 1 (FMR1 gene in adult female fragile X premutation carriers (fXPCs. Because male carriers, unlike females, do not have a second, unaffected FMR1 allele, male fXPCs should exhibit similar, if not worse, impairments. Understanding male fXPCs is of particular significance because of their increased risk of developing fragile X-associated tremor/ataxia syndrome (FXTAS. Methods Male fXPCs (n = 18 and healthy control (HC adults (n = 26 aged less than 45 years performed two psychomotor speed tasks (manual and oral and two visuospatial tasks (magnitude comparison and enumeration. In the magnitude comparison task, participants were asked to compare and judge which of two bars was larger. In the enumeration task, participants were shown between one and eight green bars in the center of the screen, and asked to state the total number displayed. Enumeration typically proceeds in one of two modes: subitizing, a fast and accurate process that works only with a small set of items, and counting, which requires accurate serial-object detection and individuation during visual search. We examined the associations between the performance on all tasks and the age, full-scale intelligent quotient, and CGG repeat length of participants. Results We found that in the magnitude comparison and enumeration tasks, male fXPCs exhibited slower reaction times relative to HCs, even after controlling for simple reaction time. Conclusions Our results indicate that male fXPCs as a group show impairments (slower reaction times in numerical visuospatial tasks, which are consistent with previous findings. This adds to a growing body of literature characterizing the phenotype in fXPCs who are asymptomatic for FXTAS. Future longitudinal studies are needed to determine how these impairments

  12. Parametric Coding of Stereo Audio

    Directory of Open Access Journals (Sweden)

    Erik Schuijers

    2005-06-01

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

  13. Digital audio and video broadcasting by satellite

    Science.gov (United States)

    Yoshino, Takehiko

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

  14. Generation of Carrier and Odd Sidebands Suppressed Optical MM-Wave with Signal Only on One Sideband Using an External Integrated Mach-Zehnder Modulator

    Institute of Scientific and Technical Information of China (English)

    XU Wei; XIN Xiang-Jun; ZHAO Tong-Gang; LING Jing; YU Chong-Xiu

    2009-01-01

    A novel scheme to generate millimeter(mm)-wave is proposed where the quadrupling of local radio frequency is formed by using an external integrated Mach-Zehnder modulator through intensity modulation or phase modulation.Generated optical mm-wave signal suffers from neither power periodical fading nor time shift of the sidebands as it is transmitted along the fiber.Receiver sensitivity of our 10Gbit/s radio-over-fiber system based on the proposed scheme is-28.3dBm under intensity modulation while-24dBm under phase modulation after 65 km transmission,and bit error rate is at 10~(-4) level after 100km transmission.Optical carrier in uplink is provided by the central station to simplify the base station,which also reduces the cost of the base station.

  15. Assessing Associations between the AURKA-HMMR-TPX2-TUBG1 Functional Module and Breast Cancer Risk in BRCA1/2 Mutation Carriers

    OpenAIRE

    2015-01-01

    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 BRC...

  16. 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.

  17. Portable Audio Design

    DEFF Research Database (Denmark)

    Groth, Sanne Krogh

    2014-01-01

    within online and physical institutional contexts. The approach focuses especially on the relationship to specific sites, and how an awareness of the relationship between the site and the production can be part of the design process. Such awareness entails several approaches: the necessity of paying...... attention to the specific genre; a grasping of the complex relationship between site and time, the actual and the virtual; and getting aquatint with the specific site’s soundscape by approaching it both intuitively and systematically. These steps will finally lead to an audio production that not only...

  18. 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

  19. Digital Audio Radio Broadcast Systems Laboratory Testing Nearly Complete

    Science.gov (United States)

    2005-01-01

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

  20. 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)

  1. Audio Watermarking with Error Correction

    Directory of Open Access Journals (Sweden)

    Aman Chadha

    2011-09-01

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

  2. Audio Watermarking with Error Correction

    CERN Document Server

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

    2011-01-01

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

  3. 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.

  4. Assessing Associations between the AURKA-HMMR-TPX2-TUBG1 Functional Module and Breast Cancer Risk in BRCA1/2 Mutation Carriers

    Science.gov (United States)

    Blanco, Ignacio; Kuchenbaecker, Karoline; Cuadras, Daniel; Wang, Xianshu; Barrowdale, Daniel; de Garibay, Gorka Ruiz; Librado, Pablo; Sánchez-Gracia, Alejandro; Rozas, Julio; Bonifaci, Núria; McGuffog, Lesley; Pankratz, Vernon S.; Islam, Abul; Mateo, Francesca; Berenguer, Antoni; Petit, Anna; Català, Isabel; Brunet, Joan; Feliubadaló, Lidia; Tornero, Eva; Benítez, Javier; Osorio, Ana; Cajal, Teresa Ramón y; Nevanlinna, Heli; Aittomäki, Kristiina; Arun, Banu K.; Toland, Amanda E.; Karlan, Beth Y.; Walsh, Christine; Lester, Jenny; Greene, Mark H.; Mai, Phuong L.; Nussbaum, Robert L.; Andrulis, Irene L.; Domchek, Susan M.; Nathanson, Katherine L.; Rebbeck, Timothy R.; Barkardottir, Rosa B.; Jakubowska, Anna; Lubinski, Jan; Durda, Katarzyna; Jaworska-Bieniek, Katarzyna; Claes, Kathleen; Van Maerken, Tom; Díez, Orland; Hansen, Thomas V.; Jønson, Lars; Gerdes, Anne-Marie; Ejlertsen, Bent; de la Hoya, Miguel; Caldés, Trinidad; Dunning, Alison M.; Oliver, Clare; Fineberg, Elena; Cook, Margaret; Peock, Susan; McCann, Emma; Murray, Alex; Jacobs, Chris; Pichert, Gabriella; Lalloo, Fiona; Chu, Carol; Dorkins, Huw; Paterson, Joan; Ong, Kai-Ren; Teixeira, Manuel R.; Hogervorst, Frans B. L.; van der Hout, Annemarie H.; Seynaeve, Caroline; van der Luijt, Rob B.; Ligtenberg, Marjolijn J. L.; Devilee, Peter; Wijnen, Juul T.; Rookus, Matti A.; Meijers-Heijboer, Hanne E. J.; Blok, Marinus J.; van den Ouweland, Ans M. W.; Aalfs, Cora M.; Rodriguez, Gustavo C.; Phillips, Kelly-Anne A.; Piedmonte, Marion; Nerenstone, Stacy R.; Bae-Jump, Victoria L.; O'Malley, David M.; Ratner, Elena S.; Schmutzler, Rita K.; Wappenschmidt, Barbara; Rhiem, Kerstin; Engel, Christoph; Meindl, Alfons; Ditsch, Nina; Arnold, Norbert; Plendl, Hansjoerg J.; Niederacher, Dieter; Sutter, Christian; Wang-Gohrke, Shan; Steinemann, Doris; Preisler-Adams, Sabine; Kast, Karin; Varon-Mateeva, Raymonda; Gehrig, Andrea; Bojesen, Anders; Pedersen, Inge Sokilde; Sunde, Lone; Jensen, Uffe Birk; Thomassen, Mads; Kruse, Torben A.; Foretova, Lenka; Peterlongo, Paolo; Bernard, Loris; Peissel, Bernard; Scuvera, Giulietta; Manoukian, Siranoush; Radice, Paolo; Ottini, Laura; Montagna, Marco; Agata, Simona; Maugard, Christine; Simard, Jacques; Soucy, Penny; Berger, Andreas; Fink-Retter, Anneliese; Singer, Christian F.; Rappaport, Christine; Geschwantler-Kaulich, Daphne; Tea, Muy-Kheng; Pfeiler, Georg; John, Esther M.; Miron, Alex; Neuhausen, Susan L.; Terry, Mary Beth; Chung, Wendy K.; Daly, Mary B.; Goldgar, David E.; Janavicius, Ramunas; Dorfling, Cecilia M.; van Rensburg, Elisabeth J.; Fostira, Florentia; Konstantopoulou, Irene; Garber, Judy; Godwin, Andrew K.; Olah, Edith; Narod, Steven A.; Rennert, Gad; Paluch, Shani Shimon; Laitman, Yael; Friedman, Eitan; Liljegren, Annelie; Rantala, Johanna; Stenmark-Askmalm, Marie; Loman, Niklas; Imyanitov, Evgeny N.; Hamann, Ute; Spurdle, Amanda B.; Healey, Sue; Weitzel, Jeffrey N.; Herzog, Josef; Margileth, David; Gorrini, Chiara; Esteller, Manel; Gómez, Antonio; Sayols, Sergi; Vidal, Enrique; Heyn, Holger; Stoppa-Lyonnet, Dominique; Léoné, Melanie; Barjhoux, Laure; Fassy-Colcombet, Marion; de Pauw, Antoine; Lasset, Christine; Ferrer, Sandra Fert; Castera, Laurent; Berthet, Pascaline; Cornelis, François; Bignon, Yves-Jean; Damiola, Francesca; Mazoyer, Sylvie; Maxwell, Christopher A.; Vijai, Joseph; Robson, Mark; Kauff, Noah; Corines, Marina J.; Villano, Danylko; Cunningham, Julie; Lee, Adam; Lindor, Noralane; Lázaro, Conxi; Easton, Douglas F.; Offit, Kenneth; Chenevix-Trench, Georgia; Couch, Fergus J.; Antoniou, Antonis C.; Pujana, Miguel Angel

    2015-01-01

    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. PMID:25830658

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

    Science.gov (United States)

    Blanco, Ignacio; Kuchenbaecker, Karoline; Cuadras, Daniel; Wang, Xianshu; Barrowdale, Daniel; de Garibay, Gorka Ruiz; Librado, Pablo; Sánchez-Gracia, Alejandro; Rozas, Julio; Bonifaci, Núria; McGuffog, Lesley; Pankratz, Vernon S; Islam, Abul; Mateo, Francesca; Berenguer, Antoni; Petit, Anna; Català, Isabel; Brunet, Joan; Feliubadaló, Lidia; Tornero, Eva; Benítez, Javier; Osorio, Ana; Ramón y Cajal, Teresa; Nevanlinna, Heli; Aittomäki, Kristiina; Arun, Banu K; Toland, Amanda E; Karlan, Beth Y; Walsh, Christine; Lester, Jenny; Greene, Mark H; Mai, Phuong L; Nussbaum, Robert L; Andrulis, Irene L; Domchek, Susan M; Nathanson, Katherine L; Rebbeck, Timothy R; Barkardottir, Rosa B; Jakubowska, Anna; Lubinski, Jan; Durda, Katarzyna; Jaworska-Bieniek, Katarzyna; Claes, Kathleen; Van Maerken, Tom; Díez, Orland; Hansen, Thomas V; Jønson, Lars; Gerdes, Anne-Marie; Ejlertsen, Bent; de la Hoya, Miguel; Caldés, Trinidad; Dunning, Alison M; Oliver, Clare; Fineberg, Elena; Cook, Margaret; Peock, Susan; McCann, Emma; Murray, Alex; Jacobs, Chris; Pichert, Gabriella; Lalloo, Fiona; Chu, Carol; Dorkins, Huw; Paterson, Joan; Ong, Kai-Ren; Teixeira, Manuel R; Hogervorst, Frans B L; van der Hout, Annemarie H; Seynaeve, Caroline; van der Luijt, Rob B; Ligtenberg, Marjolijn J L; Devilee, Peter; Wijnen, Juul T; Rookus, Matti A; Meijers-Heijboer, Hanne E J; Blok, Marinus J; van den Ouweland, Ans M W; Aalfs, Cora M; Rodriguez, Gustavo C; Phillips, Kelly-Anne A; Piedmonte, Marion; Nerenstone, Stacy R; Bae-Jump, Victoria L; O'Malley, David M; Ratner, Elena S; Schmutzler, Rita K; Wappenschmidt, Barbara; Rhiem, Kerstin; Engel, Christoph; Meindl, Alfons; Ditsch, Nina; Arnold, Norbert; Plendl, Hansjoerg J; Niederacher, Dieter; Sutter, Christian; Wang-Gohrke, Shan; Steinemann, Doris; Preisler-Adams, Sabine; Kast, Karin; Varon-Mateeva, Raymonda; Gehrig, Andrea; Bojesen, Anders; Pedersen, Inge Sokilde; Sunde, Lone; Jensen, Uffe Birk; Thomassen, Mads; Kruse, Torben A; Foretova, Lenka; Peterlongo, Paolo; Bernard, Loris; Peissel, Bernard; Scuvera, Giulietta; Manoukian, Siranoush; Radice, Paolo; Ottini, Laura; Montagna, Marco; Agata, Simona; Maugard, Christine; Simard, Jacques; Soucy, Penny; Berger, Andreas; Fink-Retter, Anneliese; Singer, Christian F; Rappaport, Christine; Geschwantler-Kaulich, Daphne; Tea, Muy-Kheng; Pfeiler, Georg; John, Esther M; Miron, Alex; Neuhausen, Susan L; Terry, Mary Beth; Chung, Wendy K; Daly, Mary B; Goldgar, David E; Janavicius, Ramunas; Dorfling, Cecilia M; van Rensburg, Elisabeth J; Fostira, Florentia; Konstantopoulou, Irene; Garber, Judy; Godwin, Andrew K; Olah, Edith; Narod, Steven A; Rennert, Gad; Paluch, Shani Shimon; Laitman, Yael; Friedman, Eitan; Liljegren, Annelie; Rantala, Johanna; Stenmark-Askmalm, Marie; Loman, Niklas; Imyanitov, Evgeny N; Hamann, Ute; Spurdle, Amanda B; Healey, Sue; Weitzel, Jeffrey N; Herzog, Josef; Margileth, David; Gorrini, Chiara; Esteller, Manel; Gómez, Antonio; Sayols, Sergi; Vidal, Enrique; Heyn, Holger; Stoppa-Lyonnet, Dominique; Léoné, Melanie; Barjhoux, Laure; Fassy-Colcombet, Marion; de Pauw, Antoine; Lasset, Christine; Ferrer, Sandra Fert; Castera, Laurent; Berthet, Pascaline; Cornelis, François; Bignon, Yves-Jean; Damiola, Francesca; Mazoyer, Sylvie; Sinilnikova, Olga M; Maxwell, Christopher A; Vijai, Joseph; Robson, Mark; Kauff, Noah; Corines, Marina J; Villano, Danylko; Cunningham, Julie; Lee, Adam; Lindor, Noralane; Lázaro, Conxi; Easton, Douglas F; Offit, Kenneth; Chenevix-Trench, Georgia; Couch, Fergus J; Antoniou, Antonis C; Pujana, Miguel Angel

    2015-01-01

    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.

  6. Audio steganography by quantization index modulation in the DCT domain%基于DCT域QIM的音频信息伪装算法

    Institute of Scientific and Technical Information of China (English)

    陈铭; 张茹; 刘凡凡; 钮心忻; 杨义先

    2009-01-01

    音频与图像相比具有信息冗余大、随机性强的特点,在音频中实现无误码的信息提取的难度更大.提出一种基于DCT域QIM(quantization index modulation)的音频信息伪装算法,算法特点如下:应用QIM原理,以量化的方式嵌入信息,根据量化区间与信息比特的映射关系提取信息,可实现盲提取;采用改进的QIM方案,针对信息提取的误码,在嵌入端与提取端进行容错处理,保证了隐藏信息的强顽健性;隐藏容量大,可达357.6biffs.实验表明,算法与传统QIM方法相比具有更好的不可感知性,100%嵌入的载密音频的信噪比在30dB以上,并且对于MP3压缩、重量化、重采样、低通滤波等攻击具有很强的顽健性,同时算法运算量小,易于实现,实用性强.

  7. Digital Audio Collections

    Directory of Open Access Journals (Sweden)

    Jason Tenter

    2010-11-01

    Full Text Available

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

  8. Digital Audio Watermarking: An Overview

    Directory of Open Access Journals (Sweden)

    Bhuvnesh Kumar Singh

    2013-10-01

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

  9. 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,

  10. Tag Based Audio Search Engine

    Directory of Open Access Journals (Sweden)

    Parameswaran Vellachu

    2012-03-01

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

  11. 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.

  12. 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

  13. 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....

  14. Asymmetric Carrier Random PWM

    DEFF Research Database (Denmark)

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

    2010-01-01

    This paper presents a new fixed carrier frequency random PWM method, where a new type of carrier wave is proposed for modulation. Based on the measurements, it is shown that the spread effect of the discrete components from the motor current spectra is very effective independent of the modulation...... 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...

  15. Modeling Audio Fingerprints: Structure, Distortion, Capacity

    OpenAIRE

    Doets, P.J.O.

    2010-01-01

    An audio fingerprint is a compact low-level representation of a multimedia signal. An audio fingerprint can be used to identify audio files or fragments in a reliable way. The use of audio fingerprints for identification consists of two phases. In the enrollment phase known content is fingerprinted, and ingested into a database, together with all relevant metadata. In the identification phase, unknown audio content is fingerprinted, and the fingerprints form the query to the database. The que...

  16. Mie Resonance-Modulated Spatial Distributions of Photogenerated Carriers in Poly(3-hexylthiophene-2,5-diyl)/Silicon Nanopillars

    Science.gov (United States)

    Kim, Eunah; Cho, Yunae; Sohn, Ahrum; Hwang, Heewon; Lee, Y. U.; Kim, Kyungkon; Park, Hyeong-Ho; Kim, Joondong; Wu, J. W.; Kim, Dong-Wook

    2016-07-01

    Organic/silicon hybrid solar cells have great potential as low-cost, high-efficiency photovoltaic devices. The superior light trapping capability, mediated by the optical resonances, of the organic/silicon hybrid nanostructure-based cells enhances their optical performance. In this work, we fabricated Si nanopillar (NP) arrays coated with organic semiconductor, poly(3-hexylthiophene-2,5-diyl), layers. Experimental and calculated optical properties of the samples showed that Mie-resonance strongly concentrated incoming light in the NPs. Spatial mapping of surface photovoltage, i.e., changes in the surface potential under illumination, using Kelvin probe force microscopy enabled us to visualize the local behavior of the photogenerated carriers in our samples. Under red light, surface photovoltage was much larger (63 meV) on the top surface of a NP than on a planar sample (13 meV), which demonstrated that the confined light in the NPs produced numerous carriers within the NPs. Since the silicon NPs provide pathways for efficient carrier transportation, high collection probability of the photogenerated carriers near the NPs can be expected. This suggests that the optical resonance in organic/silicon hybrid nanostructures benefits not only broad-band light trapping but also efficient carrier collection.

  17. 12.5 Gbps optical modulation of silicon racetrack resonator based on carrier-depletion in asymmetric p-n diode.

    Science.gov (United States)

    You, Jong-Bum; Park, Miran; Park, Jeong-Woo; Kim, Gyungock

    2008-10-27

    We present a high speed optical modulation using carrier depletion effect in an asymmetric silicon p-n diode resonator. To optimize coupling efficiency and reduce bending loss, two-step-etched waveguide is used in the racetrack resonator with a directional coupler. The quality factor of the resonator with a circumference of 260 um is 9,482, and the DC on/off ratio is 8 dB at -12V. The device shows the 3dB bandwidth of approximately8 GHz and the data transmission up to 12.5Gbit/s.

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

    Science.gov (United States)

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

    2007-09-01

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

  19. Location audio simplified capturing your audio and your audience

    CERN Document Server

    Miles, Dean

    2014-01-01

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

  20. Generation of coherent and frequency-lock multi-carriers using cascaded phase modulators and recirculating frequency shifter for Tb/s optical communication.

    Science.gov (United States)

    Zhang, Junwen; Chi, Nan; Yu, Jianjun; Shao, Yufeng; Zhu, Jiangbo; Huang, Bo; Tao, Li

    2011-07-04

    We investigate to generate coherent and frequency-lock optical multi-carriers by using cascaded phase modulators and recirculating frequency shifter (RFS) based on an EDFA loop. The phase and amplitude relation of RF signals on two cascaded phase modulators and the impact of EDFA gain are investigated. Experimental results are in good agreement with the theoretical analysis. The performance of 113 coherent and frequency-lock subcarriers with tone-to-noise ratio larger than 26dB and amplitude difference of 5dB obtained after a tilt filter covering totally 22.6nm shows that this scheme is a promising technique for the coming Tb/s optical communication.

  1. Evaluation of Audio Compression Artifacts

    Directory of Open Access Journals (Sweden)

    M. Herrera Martinez

    2007-01-01

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

  2. 数字调制器载波产生电路的FPGA设计%Design of Digital Modulator Carrier Producing Circuit Based on FPGA

    Institute of Scientific and Technical Information of China (English)

    雷能芳

    2011-01-01

    The common approach to implement Digital Modulator Carrier producing circuit on FPGA is based on a lookup table, which requires a huge volume of ROM to achieve high resolution. This paper porposes a pipelined architecture for implementation of digital modulator carrier on FPGA, which, based on CORDIC algorithm, can save considerable hardware resources and improve the speed performance as well. The system was implemented in EP1C12Q240C8, and the hardware practical test was done by embedded logic analyzer SignalTap Ⅱ of Quartus Ⅱ. The correctness and feasibility of this design is verified by practical test result.%数字调制器载波产生电路的FPGA实现通常都是基于查找表的方法,为了达到高精度要求,需要耗费大量的ROM资源去建立庞大的查找表.文中提出了一种基于流水线CORDIC算法的实现方案,可有效地节省FPGA的硬件资源,提高运算速度.电路在FPGA芯片EPIC12Q240C8上实现,并通过QuartusⅡ嵌入式逻辑分析仪SignalTapⅡ对硬件进行了实时测试,测试结果验证了设计的正确性及可行性.

  3. Multiperiodicity, modulations and flip-flops in variable star light curves III. Carrier fit analysis of LQ Hya photometry for 1982-2014

    CERN Document Server

    Olspert, Nigul; Pelt, Jaan; Cole, Elizabeth M; Hackman, Thomas; Lehtinen, Jyri; Henry, Gregory W

    2014-01-01

    We study LQ Hya photometry for 1982-2014 with the carrier fit (CF) -method and compare our results to earlier photometric analysis and recent Doppler imaging maps. We first utilize different types of statistical methods to estimate various candidates for the carrier period for the CF method. Secondly, a global fit to the whole data set and local fits to shorter segments are computed with the period that is found to be the optimal one. The harmonic least-squares analysis of all the available data reveals a short period close to 1.6 days as a limiting value for a set of significant frequencies. We interpret this as the rotation period of the spots near the equatorial region. In addition, the distribution of the significant periods is found to be bimodal, hinting of a longer-term modulating period, which we set out to study with a two-harmonic CF model. Weak modulation signal is, indeed retrieved, with a period of roughly 6.9 years. The phase dispersion analysis gives a clear symmetric minimum for coherence time...

  4. Joint Acoustic and Modulation Frequency

    Directory of Open Access Journals (Sweden)

    Les Atlas

    2003-06-01

    Full Text Available There is a considerable evidence that our perception of sound uses important features which is related to underlying signal modulations. This topic has been studied extensively via perceptual experiments, yet there are few, if any, well-developed signal processing methods which capitalize on or model these effects. We begin by summarizing evidence of the importance of modulation representations from psychophysical, physiological, and other sources. The concept of a two-dimensional joint acoustic and modulation frequency representation is proposed. A simple single sinusoidal amplitude modulator of a sinusoidal carrier is then used to illustrate properties of an unconstrained and ideal joint representation. Added constraints are required to remove or reduce undesired interference terms and to provide invertibility. It is then noted that the constraints would also apply to more general and complex cases of broader modulation and carriers. Applications in single-channel speaker separation and in audio coding are used to illustrate the applicability of this joint representation. Other applications in signal analysis and filtering are suggested.

  5. Amplificador de audio Clase D

    OpenAIRE

    2012-01-01

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

  6. 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

  7. Simple Solutions for Space Station Audio Problems

    Science.gov (United States)

    Wood, Eric

    2016-01-01

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

  8. 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...

  9. BASIC AMINO ACID CARRIER 2 gene expression modulates arginine and urea content and stress recovery in Arabidopsis leaves.

    Directory of Open Access Journals (Sweden)

    Séverine ePlanchais

    2014-07-01

    Full Text Available In plants, basic amino acids are important for the synthesis of proteins and signaling molecules and for nitrogen recycling. The Arabidopsis nuclear gene BASIC AMINO ACID CARRIER 2 (BAC2 encodes a mitochondria-located carrier that transports basic amino acids in vitro. We present here an analysis of the physiological and genetic function of BAC2 in planta. When BAC2 is overexpressed in vivo, it triggers catabolism of arginine, a basic amino acid, leading to arginine depletion and urea accumulation in leaves. BAC2 expression was known to be strongly induced by stress. We found that compared to wild type plants, bac2 null mutants (bac2-1 recover poorly from hyperosmotic stress when restarting leaf expansion. The bac2-1 transcriptome differs from the wild-type transcriptome in control conditions and under hyperosmotic stress. The expression of genes encoding stress-related transcription factors, arginine metabolism enzymes, and transporters is particularly disturbed in bac2-1, and in control conditions, the bac2-1 transcriptome has some hallmarks of a wild-type stress transcriptome. The BAC2 carrier is therefore involved in controlling the balance of arginine and arginine-derived metabolites and its associated amino acid metabolism is physiologically important in equipping plants to respond to and recover from stress.

  10. Audio watermark a comprehensive foundation using Matlab

    CERN Document Server

    Lin, Yiqing

    2015-01-01

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

  11. Audio Watermarking Using Lsb With Adjustment Method

    Directory of Open Access Journals (Sweden)

    Ansith.S, Priyanka Udayabhanu

    2013-05-01

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

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

    NARCIS (Netherlands)

    Westerveld, Hugo; Schinkel, Daniël; Tuijl, van Ed

    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 ba

  13. 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 -…

  14. 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…

  15. Digital Augmented Reality Audio Headset

    Directory of Open Access Journals (Sweden)

    Jussi Rämö

    2012-01-01

    Full Text Available Augmented reality audio (ARA combines virtual sound sources with the real sonic environment of the user. An ARA system can be realized with a headset containing binaural microphones. Ideally, the ARA headset should be acoustically transparent, that is, it should not cause audible modification to the surrounding sound. A practical implementation of an ARA mixer requires a low-latency headphone reproduction system with additional equalization to compensate for the attenuation and the modified ear canal resonances caused by the headphones. This paper proposes digital IIR filters to realize the required equalization and evaluates a real-time prototype ARA system. Measurements show that the throughput latency of the digital prototype ARA system can be less than 1.4 ms, which is sufficiently small in practice. When the direct and processed sounds are combined in the ear, a comb filtering effect is brought about and appears as notches in the frequency response. The comb filter effect in speech and music signals was studied in a listening test and it was found to be inaudible when the attenuation is 20 dB. Insert ARA headphones have a sufficient attenuation at frequencies above about 1 kHz. The proposed digital ARA system enables several immersive audio applications, such as a virtual audio tourist guide and audio teleconferencing.

  16. Haptic and Audio Interaction Design

    DEFF Research Database (Denmark)

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

  17. The Audio-Visual Man.

    Science.gov (United States)

    Babin, Pierre, Ed.

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

  18. Audio-Visual Aids: Historians in Blunderland.

    Science.gov (United States)

    Decarie, Graeme

    1988-01-01

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

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

    Science.gov (United States)

    Morand, J J

    1989-01-01

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

  20. Spatial audio quality perception (part 1)

    DEFF Research Database (Denmark)

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

    2015-01-01

    Spatial audio processes (SAPs) commonly encountered in consumer audio reproduction systems are known to produce a range of impairments to spatial quality. By way of two listening tests, this paper investigated the degree of degradation of the spatial quality of six 5-channel audio recordings resu...

  1. Design of an Audio Interface for Patmos

    OpenAIRE

    Ausin, Daniel Sanz; Goerge, Fabian

    2017-01-01

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

  2. Hiding secret data into a carrier image

    OpenAIRE

    Ovidiu COSMA

    2012-01-01

    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 wh...

  3. Self-phase modulation of a single-cycle terahertz pulse by nonlinear free-carrier response in a semiconductor

    DEFF Research Database (Denmark)

    Turchinovich, Dmitry; Hvam, Jørn Märcher; Hoffmann, Matthias C.

    2012-01-01

    We investigate the self-phase modulation (SPM) of a single-cycle terahertz pulse in a semiconductor, using bulk n-GaAs as a model system. The SPM arises from the heating of free electrons in the electric field of the terahertz pulse, leading to an ultrafast reduction of the plasma frequency, and ...

  4. Two-dimensional audio watermark for MPEG AAC audio

    Science.gov (United States)

    Tachibana, Ryuki

    2004-06-01

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

  5. 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...

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

    Directory of Open Access Journals (Sweden)

    Nitin Arora

    2010-06-01

    Full Text Available AudioRegent is a command-line Python script currently being used by the University of Alabama Libraries’ Digital Services to create web-deliverable MP3s from regions within archival audio files. In conjunction with a small-footprint XML file called SimpleADL and SoX, an open-source command-line audio editor, AudioRegent batch processes archival audio files, allowing for one or many user-defined regions, particular to each audio file, to be extracted with additional audio processing in a transparent manner that leaves the archival audio file unaltered. Doing so has alleviated many of the tensions of cumbersome workflows, complicated documentation, preservation concerns, and reliance on expensive closed-source GUI audio applications.

  7. Haptic and Audio Interaction Design

    DEFF Research Database (Denmark)

    This book constitutes the refereed proceedings of the 5th International Workshop on Haptic and Audio Interaction Design, HAID 2010 held in Copenhagen, Denmark, in September 2010. The 21 revised full papers presented were carefully reviewed and selected for inclusion in the book. The papers are or...... are organized in topical sections on multimodal integration, tactile and sonic explorations, walking and navigation interfaces, prototype design and evaluation, and gestures and emotions.......This book constitutes the refereed proceedings of the 5th International Workshop on Haptic and Audio Interaction Design, HAID 2010 held in Copenhagen, Denmark, in September 2010. The 21 revised full papers presented were carefully reviewed and selected for inclusion in the book. The papers...

  8. Sterol carrier protein 2 regulates proximal tubule size in the Xenopus pronephric kidney by modulating lipid rafts.

    Science.gov (United States)

    Cerqueira, Débora M; Tran, Uyen; Romaker, Daniel; Abreu, José G; Wessely, Oliver

    2014-10-01

    The kidney is a homeostatic organ required for waste excretion and reabsorption of water, salts and other macromolecules. To this end, a complex series of developmental steps ensures the formation of a correctly patterned and properly proportioned organ. While previous studies have mainly focused on the individual signaling pathways, the formation of higher order receptor complexes in lipid rafts is an equally important aspect. These membrane platforms are characterized by differences in local lipid and protein compositions. Indeed, the cells in the Xenopus pronephric kidney were positive for the lipid raft markers ganglioside GM1 and Caveolin-1. To specifically interfere with lipid raft function in vivo, we focused on the Sterol Carrier Protein 2 (scp2), a multifunctional protein that is an important player in remodeling lipid raft composition. In Xenopus, scp2 mRNA was strongly expressed in differentiated epithelial structures of the pronephric kidney. Knockdown of scp2 did not interfere with the patterning of the kidney along its proximo-distal axis, but dramatically decreased the size of the kidney, in particular the proximal tubules. This phenotype was accompanied by a reduction of lipid rafts, but was independent of the peroxisomal or transcriptional activities of scp2. Finally, disrupting lipid micro-domains by inhibiting cholesterol synthesis using Mevinolin phenocopied the defects seen in scp2 morphants. Together these data underscore the importance for localized signaling platforms in the proper formation of the Xenopus kidney.

  9. Spacelab carrier complement thermal design and performance

    Science.gov (United States)

    Bancroft, S.; Key, R.; Kittredge, S.

    1992-01-01

    The present discussion of the Spacelab carrier complement, which encompasses a Module Carrier, a Module-Pallet Carrier, and a Multiplexer/Demultiplexer Pallet, gives attention to both active and passive thermal performance capabilities, and presents ground testing and analytical results obtained to date. An account is given of the prospective use of a Spacelab Multipurpose Experiment Support Structure.

  10. 对线性调频脉冲压缩雷达的多载波调制转发干扰%Multi-carrier Modulation Repeater Jamming against Linear Frequency Modulated Pulse-compression Radar

    Institute of Scientific and Technical Information of China (English)

    王杰贵; 张鹏程

    2015-01-01

    目前对线性调频(LFM)脉冲压缩雷达转发欺骗干扰主要通过移频调制转发和采样直接转发实现,常规转发干扰样式简单,干扰信号规律性强、复杂度低。该文提出一种基于间歇采样的多载波调制转发新型干扰样式。首先引用码片的概念对间歇采样过程重新建模,在此基础上,通过对当前采样码片附加不同移频量,结合多载波并行调制体制对其进行串并转换,利用不同次转发信号各子载波间的干扰累积,实现对LFM脉冲压缩雷达的数量、幅度、空间分布可控的逼真假目标干扰。仿真表明该干扰样式比移频干扰和直接转发干扰具有更好的干扰效果。%Repeater deception jamming against Linear Frequency Modulated (LFM) pulse-compression radar is realized by frequency-shift repeater and direct repeater jamming so far. Conventional repeater jamming type is simple. Regularity of jamming signal is strong and complexity is low. A new repeater jamming type with multi-carrier modulation based on intermittent sampling is proposed. Firstly, the model of intermittent sampling is rebuilt with the code chip concept. Based on this, lifelike false targets with the quantity, amplitude and space distribution which can be controlled are produced by attaching different frequency-shift component to the present sampling code chip, deserializing signal used multi-carrier parallel modulation system and utilizing the accumulation of different times repeater signal jamming effect among sub-carriers. The simulation results show that the new jamming type has better performance than frequency-shift jamming and direct repeater jamming.

  11. 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.

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

    Directory of Open Access Journals (Sweden)

    Theodoros Giannakopoulos

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

  13. 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.

  14. Review of AVS Audio Coding Standard

    Institute of Scientific and Technical Information of China (English)

    ZHANG Tao; ZHANG Caixia; ZHAO Xin

    2016-01-01

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

  15. 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...

  16. 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.

  17. A novel fiber audio transmission system for secure communication

    Institute of Scientific and Technical Information of China (English)

    SU Ke; JIA Bo

    2005-01-01

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

  18. Distortion Estimation in Compressed Music Using Only Audio Fingerprints

    NARCIS (Netherlands)

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

    2008-01-01

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

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

    NARCIS (Netherlands)

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

    2006-01-01

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

  20. Drug release-modulating mechanism of hydrophilic hydroxypropylmethylcellulose matrix tablets: distribution of atoms and carrier and texture analysis.

    Science.gov (United States)

    Park, Jun-Bom; Lim, Jisung; Kang, Chin-Yang; Lee, Beom-Jin

    2013-12-01

    Although release profiles of drug from hydrophilic matrices have been well recognized, the visual distribution of hydroxypropylmethylcellulose (HPMC) and atoms inside of internal structures of hydrophilic HPMC matrices has not been characterized. In this paper, drug release mechanism from HPMC matrix tablet was investigated based on the release behaviors of HPMC, physical properties of gelled HPMC tablet and atomic distributions of formulation components using diverse instruments. A matrix tablet consisting of hydroxypropyl methylcellulose (HPMC 6, 4,000 and 100,000 mPa·s), chlorpheniramine maleate (CPM) as a model and fumed silicon dioxide (Aerosil(®) 200) was prepared via direct compression. The distribution of atoms and HPMC imaging were characterized using scanning electron microscope (SEM)/ energy-dispersive X-ray spectroscopy (EDX), and near-infrared (NIR) analysis, respectively as a function of time. A texture analyzer was also used to characterize the thickness and maintenance of gel layer of HPMC matrix tablet. The HPMC matrix tablets showed Higuchi release kinetics with no lag time against the square root of time. High viscosity grades of HPMC gave retarded release rate because of the greater swelling and gel thickness as characterized by texture analyzer. According to the NIR imaging, low-viscosity-grade HPMC (6 mPa·s) quickly leached out onto the surface of the tablet, while the high-viscosity-grade HPMC (4000 mPa·s) formed much thicker gel layer around the tablet and maintained longer via slow erosion, resulting in retarded drug release. The atomic distribution of the drug (chlorine, carbon, oxygen), HPMC (carbon, oxygen) and silicon dioxide (silica, oxygen) and NIR imaging of HPMC corresponded with the dissolution behaviors of drug as a function of time. The use of imaging and texture analyses could be applicable to explain the release- modulating mechanism of hydrophilic HPMC matrix tablets.

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

    DEFF Research Database (Denmark)

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

    2012-01-01

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

  2. 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.

  3. Audio Classification from Time-Frequency Texture

    CERN Document Server

    Yu, Guoshen

    2008-01-01

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

  4. Presence and the utility of audio spatialization

    DEFF Research Database (Denmark)

    Bormann, Karsten

    2005-01-01

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

  5. 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...

  6. Audio-Visual Aids in Universities

    Science.gov (United States)

    Douglas, Jackie

    1970-01-01

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

  7. Digital Advances in Contemporary Audio Production.

    Science.gov (United States)

    Shields, Steven O.

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

  8. Stego-audio Using Genetic Algorithm Approach

    Directory of Open Access Journals (Sweden)

    V. Santhi

    2014-06-01

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

  9. Video salient event classification using audio features

    Science.gov (United States)

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

    2014-03-01

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

  10. 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.

  11. The HDTV digital audio matrix

    Science.gov (United States)

    Mason, A. J.

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

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

    CERN Document Server

    Dangarwala, Kruti

    2010-01-01

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

  13. 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

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

    OpenAIRE

    Westerveld, Hugo; Schinkel, Daniël; Tuijl, van, B.A.J.

    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 band. These mechanisms lead to a higher noise level and more distortion in the audio band. OBN also leads to interference in the LF and MF band, compromising e.g. AM radio reception. To avoid these p...

  15. Implementing Audio-CASI on Windows' Platforms.

    Science.gov (United States)

    Cooley, Philip C; Turner, Charles F

    1998-01-01

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

  16. 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

  17. Audio-Visual Classification of Sports Types

    DEFF Research Database (Denmark)

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

    2015-01-01

    In this work we propose a method for classification of sports types from combined audio and visual features ex- tracted from thermal video. From audio Mel Frequency Cepstral Coefficients (MFCC) are extracted, and PCA are applied to reduce the feature space to 10 dimensions. From the visual modality...... short trajectories are constructed to rep- resent the motion of players. From these, four motion fea- tures are extracted and combined directly with audio fea- tures for classification. A k-nearest neighbour classifier is applied for classification of 180 1-minute video sequences from three sports types...

  18. Quantization of wavelet packet audio coding

    Institute of Scientific and Technical Information of China (English)

    Tan Jianguo; Zhang Wenjun; Liu Peilin

    2006-01-01

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

  19. Study of carrier energetics in ITO/P(VDF-TrFE)/pentacene/Au diode by using electric-field-induced optical second harmonic generation measurement and charge modulation spectroscopy

    Science.gov (United States)

    Otsuka, Takako; Taguchi, Dai; Manaka, Takaaki; Iwamoto, Mitsumasa

    2017-02-01

    By using electric-field-induced optical second harmonic generation (EFISHG) measurement and charge modulation spectroscopy (CMS), we studied carrier behavior and polarization reversal in ITO/ poly(vinylidene fluoride trifluoroethylene) (P(VDF-TrFE))/pentacene/Au diodes with a ferroelectric P(VDF-TrFE) layer in terms of carrier energetics. The current-voltage (I-V) characteristics of the diodes showed three-step polarization reversal in the dark. However, the I-V was totally different under illumination and exhibited two-step behavior. EFISHG probed the internal electric field in the pentacene layer and accounted for the polarization reversal change due to charge accumulation at the pentacene/P(VDF-TrFE) interface. CMS probed the related carrier energetics and indicated that exciton dissociation in pentacene molecular states governed carrier accumulation at the pentacene/ferroelectric interface, leading to different polarization reversal processes in the dark and under light illumination. Combining EFISHG measurement and CMS provides us a way to study carrier energetics that govern polarization reversal in ferroelectric P(VDF-TrFE)/pentacene diodes.

  20. Physical Problems, Sonic Implications. A discussion of the ethics of preservation treatments and audio recordings

    Directory of Open Access Journals (Sweden)

    Kevin Bradley

    2009-09-01

    Full Text Available Conservators have traditionally operated under a particular set of ethical constraints. The AIC code of ethics, for example, states “The conservation professional should only recommend or undertake treatment that is judged suitable to the preservation of the aesthetic, conceptual, and physical characteristics of the cultural property”. However, when treating sound recordings the situation may well arise where a physical treatment will alter the physical characteristics of the audio carrier, though simultaneously restore or improve the ability of the carrier to reproduce the sound it carries. Where does the responsibility of the sound archivist lie? This paper considers some of the ethical issues surrounding treatments of audio recordings and considers just what it is that we are trying to preserve.

  1. The Introduction and Refinement of the Assessment of Digitally Recorded Audio Presentations

    Science.gov (United States)

    Sinclair, Stefanie

    2016-01-01

    This case study critically evaluates benefits and challenges of a form of assessment included in a final year undergraduate Religious Studies Open University module, which combines a written essay task with a digital audio recording of a short oral presentation. Based on the analysis of student and tutor feedback and sample assignments, this study…

  2. Implementation of Audio signal by using wavelet transform

    Directory of Open Access Journals (Sweden)

    Chakresh kumar,

    2010-10-01

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

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

    OpenAIRE

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

    2003-01-01

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

  4. 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.

  5. 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.

  6. Audio-visual affective expression recognition

    Science.gov (United States)

    Huang, Thomas S.; Zeng, Zhihong

    2007-11-01

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

  7. 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.

  8. Definición de audio

    OpenAIRE

    Montañez, Luis A.; Cabrera, Juan G.

    2015-01-01

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

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

    Science.gov (United States)

    Beasley, Augie

    1995-01-01

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

  10. 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...

  11. 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.

  12. Audio watermarking for live performance

    Science.gov (United States)

    Tachibana, Ryuki

    2003-06-01

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

  13. Audio description as an accessibility enhancer

    OpenAIRE

    Martins, Cláudia Susana Nunes

    2012-01-01

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

  14. Watermarking-Based Digital Audio Data Authentication

    Directory of Open Access Journals (Sweden)

    Jana Dittmann

    2003-09-01

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

  15. Audio Video Compression Stream Synthesis and Implementation

    Institute of Scientific and Technical Information of China (English)

    徐燕凌; 方向忠; 周源华

    2004-01-01

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

  16. PENGGUNAAN MEDIA AUDIO DALAM PEMBELAJARAN STENOGRAFI

    Directory of Open Access Journals (Sweden)

    S Martono

    2011-06-01

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

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

    OpenAIRE

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

    2015-01-01

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

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

    OpenAIRE

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

    2013-01-01

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

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

    Science.gov (United States)

    Vigil, A. J.

    1995-12-01

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

  20. 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.

  1. Design of mine-used DC carrier telephone based on STM32

    Science.gov (United States)

    Chen, Goufan; Zhou, Hui; Zhan, Minhua; Wang, Jian

    2016-01-01

    Abide by the design principles of mine intrinsically safe circuit, according to the need of underground communication in coal mine, the paper proposed a design scheme of DC carrier telephone which can dial. The design circuit of the telephone is introduced in detail. The telephone's voice signals are generated by the microphone. After enlarged then the voice signals are modulated to frequency signals by LM567 chip. The frequency signals are coupled by transformer and then transmitted by 12V DC power supply line to the other voice terminals. In the voice terminal the signals are demodulated by LM567 demodulation circuit and enlarged by LM386, then, the amplified audio signals are output from a speaker. The dialing circuit is designed based on the STM32 MCU. The dial information is transmitted to the other telephone terminals by CAN bus. The measured distance calls is greater than 2000m, volume is larger than 85dB, good results.

  2. Audio Journal in an ELT Context

    Directory of Open Access Journals (Sweden)

    Neşe Aysin Siyli

    2012-09-01

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

  3. Aircraft Carriers

    DEFF Research Database (Denmark)

    Nødskov, Kim; Kværnø, Ole

    in Asia and will balance the carrier acquisitions of the United States, the United Kingdom, Russia and India. China’s current military strategy is predominantly defensive, its offensive elements being mainly focused on Taiwan. If China decides to acquire a large carrier with offensive capabilities......, then the country will also acquire the capability to project military power into the region beyond Taiwan, which it does not possess today. In this way, China will have the military capability to permit a change of strategy from the mainly defensive, mainland, Taiwan-based strategy to a more assertive strategy...... catapult with which to launch the fi ghter aircraft, not to mention the possible development of a nuclear power plant for the ship. The Russian press has indicated that China is negotiating to buy SU-33 fi ghters, which Russia uses on the Kuznetsov carrier. The SU-33 is, in its modernized version...

  4. 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...

  5. 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.

  6. Modulation of the Microenvironment Surrounding the Active Site of Penicillin G Acylase Immobilized on Acrylic Carriers Improves the Enzymatic Synthesis of Cephalosporins

    Directory of Open Access Journals (Sweden)

    Paolo Bonomi

    2013-11-01

    Full Text Available The catalytic properties of penicillin G acylase (PGA from Escherichia coli in kinetically controlled synthesis of β-lactam antibiotics are negatively affected upon immobilization on hydrophobic acrylic carriers. Two strategies have been here pursued to improve the synthetic performance of PGA immobilized on epoxy-activated acrylic carriers. First, an aldehyde-based spacer was inserted on the carrier surface by glutaraldehyde activation (immobilization yield = 50%. The resulting 3-fold higher synthesis/hydrolysis ratio (vs/vh1 = 9.7 ± 0.7 and 10.9 ± 0.7 for Eupergit® C and Sepabeads® EC-EP, respectively with respect to the unmodified support (vs/vh1 = 3.3 ± 0.4 was ascribed to a facilitated diffusion of substrates and products as a result of the increased distance between the enzyme and the carrier surface. A second series of catalysts was prepared by direct immobilization of PGA on epoxy-activated acrylic carriers (Eupergit® C, followed by quenching of oxiranes not involved in the binding with the protein with different nucleophiles (amino acids, amines, amino alcohols, thiols and amino thiols. In most cases, this derivatization increased the synthesis/hydrolysis ratio with respect to the non derivatized carrier. Particularly, post-immobilization treatment with cysteine resulted in about 2.5-fold higher vs/vh1 compared to the untreated biocatalyst, although the immobilization yield decreased from 70% (untreated Eupergit® C to 20%. Glutaraldehyde- and cysteine-treated Eupergit® C catalyzed the synthesis of cefazolin in 88% (±0.9 and 87% (±1.6 conversion, respectively, whereas untreated Eupergit® C afforded this antibiotic in 79% (±1.2 conversion.

  7. Nonlinear dynamic macromodeling techniques for audio systems

    Science.gov (United States)

    Ogrodzki, Jan; Bieńkowski, Piotr

    2015-09-01

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

  8. Information Security using Audio Steganography -A Survey

    Directory of Open Access Journals (Sweden)

    B. Santhi

    2012-07-01

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

  9. 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.

  10. Synchronization and comparison of Lifelog audio recordings

    DEFF Research Database (Denmark)

    Nielsen, Andreas Brinch; Hansen, Lars Kai

    2008-01-01

    We investigate concurrent ‘Lifelog’ audio recordings to locate segments from the same environment. We compare two techniques earlier proposed for pattern recognition in extended audio recordings, namely cross-correlation and a fingerprinting technique. If successful, such alignment can be used...... as a preprocessing step to select and synchronize recordings before further processing. The two methods perform similarly in classification, but fingerprinting scales better with the number of recordings, while cross-correlation can offer sample resolution synchronization. We propose and investigate the benefits...

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

    Directory of Open Access Journals (Sweden)

    Anja Kuchenbuch

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

  12. 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.

  13. Assessing Associations between the AURKA-HMMR-TPX2-TUBG1 Functional Module and Breast Cancer Risk in BRCA1/2 Mutation Carriers

    NARCIS (Netherlands)

    Blanco, Ignacio; Kuchenbaecker, Karoline; Cuadras, Daniel; Wang, Xianshu; Barrowdale, Daniel; Ruiz de Garibay, Gorka; Librado, Pablo; Sanchez-Gracia, Alejandro; Rozas, Julio; Bonifaci, Nuria; McGuffog, Lesley; Pankratz, Vernon S.; Islam, Abul; Mateo, Francesca; Berenguer, Antoni; Petit, Anna; Catala, Isabel; Brunet, Joan; Feliubadalo, Lidia; Tornero, Eva; Benitez, Javier; Osorio, Ana; Cajal, Teresa Ramon Y.; Nevanlinna, Heli; Aittomaki, Kristiina; Arun, Banu K.; Toland, Amanda E.; Karlan, Beth Y.; Walsh, Christine; Lester, Jenny; Greene, Mark H.; Mai, Phuong L.; Nussbaum, Robert L.; Andrulis, Irene L.; Domchek, Susan M.; Nathanson, Katherine L.; Rebbeck, Timothy R.; Barkardottir, Rosa B.; Jakubowska, Anna; Lubinski, Jan; Durda, Katarzyna; Jaworska-Bieniek, Katarzyna; Claes, Kathleen; Van Maerken, Tom; Diez, Orland; Hansen, Thomas V.; Jonson, Lars; Gerdes, Anne-Marie; Ejlertsen, Bent; de la Hoya, Miguel; Caldes, Trinidad; Dunning, Alison M.; Oliver, Clare; Fineberg, Elena; Cook, Margaret; Peock, Susan; McCann, Emma; Murray, Alex; Jacobs, Chris; Pichert, Gabriella; Lalloo, Fiona; Chu, Carol; Dorkins, Huw; Paterson, Joan; Ong, Kai-Ren; Teixeira, Manuel R.; Teixeira, J.; Hogervorst, Frans B. L.; van der Hout, Annemarie H.; Seynaeve, Caroline; van der Luijt, Rob B.; Ligtenberg, Marjolijn J. L.; Devilee, Peter; Wijnen, Juul T.; Rookus, Matti A.; Meijers-Heijboer, Hanne E. J.; Blok, Marinus J.; van den Ouweland, Ans M. W.; Aalfs, Cora M.; Rodriguez, Gustavo C.; Phillips, Kelly-Anne A.; Piedmonte, Marion; Nerenstone, Stacy R.; Bae-Jump, Victoria L.; O'Malley, David M.; Ratner, Elena S.; Schmutzler, Rita K.; Wappenschmidt, Barbara; Rhiem, Kerstin; Engel, Christoph; Meindl, Alfons; Ditsch, Nina; Arnold, Norbert; Plendl, Hansjoerg J.; Niederacher, Dieter; Sutter, Christian; Wang-Gohrke, Shan; Steinemann, Doris; Preisler-Adams, Sabine; Kast, Karin; Varon-Mateeva, Raymonda; Gehrig, Andrea; Bojesen, Anders; Pedersen, Inge Sokilde; Sunde, Lone; Jensen, Uffe Birk; Thomassen, Mads; Kruse, Torben A.; Foretova, Lenka; Peterlongo, Paolo; Bernard, Loris; Peissel, Bernard; Scuvera, Giulietta; Manoukian, Siranoush; Radice, Paolo; Ottini, Laura; Montagna, Marco; Agata, Simona; Maugard, Christine; Simard, Jacques; Soucy, Penny; Berger, Andreas; Fink-Retter, Anneliese; Singer, Christian F.; Rappaport, Christine; Geschwantler-Kaulich, Daphne; Tea, Muy-Kheng; Pfeiler, Georg; John, Esther M.; Miron, Alex; Neuhausen, Susan L.; Terry, Mary Beth; Chung, Wendy K.; Daly, Mary B.; Goldgar, David E.; Janavicius, Ramunas; Dorfling, Cecilia M.; van Rensburg, Elisabeth J.; Fostira, Florentia; Konstantopoulou, Irene; Garber, Judy; Godwin, Andrew K.; Olah, Edith; Narod, Steven A.; Rennert, Gad; Paluch, Shani Shimon; Laitman, Yael; Friedman, Eitan; Liljegren, Annelie; Rantala, Johanna; Stenmark-Askmalm, Marie; Loman, Niklas; Imyanitov, Evgeny N.; Hamann, Ute; Spurdle, Amanda B.; Healey, Sue; Weitzel, Jeffrey N.; Herzog, Josef; Margileth, David; Gorrini, Chiara; Esteller, Manel; Gomez, Antonio; Sayols, Sergi; Vidal, Enrique; Heyn, Holger; Stoppa-Lyonnet, Dominique; Leone, Melanie; Barjhoux, Laure; Fassy-Colcombet, Marion; de Pauw, Antoine; Lasset, Christine; Ferrer, Sandra Fert; Castera, Laurent; Berthet, Pascaline; Cornelis, Francois; Bignon, Yves-Jean; Damiola, Francesca; Mazoyer, Sylvie; Sinilnikova, Olga M.; Maxwell, Christopher A.; Vijai, Joseph; Robson, Mark; Kauff, Noah; Corines, Marina J.; Villano, Danylko; Cunningham, Julie; Lee, Adam; Lindor, Noralane; Lazaro, Conxi; Easton, Douglas F.; Offit, Kenneth; Chenevix-Trench, Georgia; Couch, Fergus J.; Antoniou, Antonis C.; Angel Pujana, Miguel

    2015-01-01

    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

  14. Assessing Associations between the AURKA-HMMR-TPX2-TUBG1 Functional Module and Breast Cancer Risk in BRCA1/2 Mutation Carriers

    DEFF Research Database (Denmark)

    Blanco, Ignacio; Kuchenbaecker, Karoline; Cuadras, Daniel;

    2015-01-01

    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 ...

  15. Audio Mining with emphasis on Music Genre Classification

    DEFF Research Database (Denmark)

    Meng, Anders

    2004-01-01

    Audio is an important part of our daily life, basically it increases our impression of the world around us whether this is communication, music, danger detection etc. Currently the field of Audio Mining, which here includes areas of music genre, music recognition / retrieval, playlist generation...... in searching / retrieving audio effectively is needed. Currently, search engines such as e.g. Google, AltaVista etc. do not search into audio files, but uses either the textual information attached to the audio file or the textual information around the audio. Also in the hearing aid industries around...... the world the problem of detecting environments from the input audio is researched as to increase the life quality of hearing-impaired. Basically there is a lot of work within the field of audio mining. The presentation will mainly focus on music genre classification where we have a fixed amount of genres...

  16. 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...

  17. Cross-modal retrieval of scripted speech audio

    Science.gov (United States)

    Owen, Charles B.; Makedon, Fillia

    1997-12-01

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

  18. A Double Modulation Wave Carrier-Based PWM Strategy for Three-Level Neutral Point Clamped Converter%基于双调制波的三电平NPC变流器载波调制策略

    Institute of Scientific and Technical Information of China (English)

    李宁; 王跃; 王兆安

    2014-01-01

    In modulation strategies for three-level neutral point clamped (NPC) converters the double modulation wave carrier-based PWM (DMWPWM) strategy is such a modulation strategy, by which the neutral point voltage control under the entire modulation index and full power factor can be achieved. Firstly, the basic principle of existing DMWPWM strategies are analyzed and the relation between modulation wave in DMWPWM and that in traditional sinusoidal pulse width modulation (SPWM) is analyzed, and on this basis a DMWPWM strategy for three-level NPC converter is proposed;secondly, in the aspects of total harmonic distortion (THD) characteristic of output phase voltage, DC voltage utilization and switching loss of the device the proposed DMWPWM strategy is compared with traditional SPWM strategy to further expound the features of the proposed method. The correctness of theoretical analysis is validated by results from simulation and experiments.%三电平NPC变流器的调制策略中,双调制波载波调制策略(double modulation wave carrier-based PWM , DMWPWM)是一种可以实现全调制度和全功率因数中点电压无波动的调制策略。首先分析了已有DMWPWM策略的基本原理,推导了DMWPWM策略调制波与传统SPWM策略调制波的关系,在此基础上提出了应用于三电平NPC变流器的DMWPWM策略。然后,从输出相电压总谐波畸变率(total harmonic distortion,THD)特性、直流电压利用率以及器件开关损耗三方面对新型DMWPWM策略与传统正弦脉冲宽度调制(sinusoidal pulse width modulation,SPWM)策略进行对比,进一步阐明了所提方法特点。仿真和实验结果验证了理论分析的正确性。

  19. 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.

  20. Relevant Research on Audio-Tutorial Methods

    Science.gov (United States)

    Novak, Joseph D.

    1970-01-01

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

  1. 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.

  2. Transparency benchmarking on audio watermarks and steganography

    Science.gov (United States)

    Kraetzer, Christian; Dittmann, Jana; Lang, Andreas

    2006-02-01

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

  3. 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...

  4. An ESL Audio-Script Writing Workshop

    Science.gov (United States)

    Miller, Carla

    2012-01-01

    The roles of dialogue, collaborative writing, and authentic communication have been explored as effective strategies in second language writing classrooms. In this article, the stages of an innovative, multi-skill writing method, which embeds students' personal voices into the writing process, are explored. A 10-step ESL Audio Script Writing Model…

  5. Audible Aliasing Distortion in Digital Audio Synthesis

    Directory of Open Access Journals (Sweden)

    J. Schimmel

    2012-04-01

    Full Text Available This paper deals with aliasing distortion in digital audio signal synthesis of classic periodic waveforms with infinite Fourier series, for electronic musical instruments. When these waveforms are generated in the digital domain then the aliasing appears due to its unlimited bandwidth. There are several techniques for the synthesis of these signals that have been designed to avoid or reduce the aliasing distortion. However, these techniques have high computing demands. One can say that today's computers have enough computing power to use these methods. However, we have to realize that today’s computer-aided music production requires tens of multi-timbre voices generated simultaneously by software synthesizers and the most of the computing power must be reserved for hard-disc recording subsystem and real-time audio processing of many audio channels with a lot of audio effects. Trivially generated classic analog synthesizer waveforms are therefore still effective for sound synthesis. We cannot avoid the aliasing distortion but spectral components produced by the aliasing can be masked with harmonic components and thus made inaudible if sufficient oversampling ratio is used. This paper deals with the assessment of audible aliasing distortion with the help of a psychoacoustic model of simultaneous masking and compares the computing demands of trivial generation using oversampling with those of other methods.

  6. Structuring Broadcast Audio for Information Access

    Directory of Open Access Journals (Sweden)

    Gauvain Jean-Luc

    2003-01-01

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

  7. 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.

  8. Audio-visual integration in schizophrenia

    NARCIS (Netherlands)

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

    2003-01-01

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

  9. Audio-visual integration in schizophrenia.

    NARCIS (Netherlands)

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

    2003-01-01

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

  10. Building Digital Audio Preservation Infrastructure and Workflows

    Science.gov (United States)

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

    2010-01-01

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

  11. Calibration of an audio frequency noise generator

    DEFF Research Database (Denmark)

    Diamond, Joseph M.

    1966-01-01

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

  12. Debugging of Class-D Audio Power Amplifiers

    DEFF Research Database (Denmark)

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

    2012-01-01

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

  13. Progressive Syntax-Rich Coding of Multichannel Audio Sources

    Directory of Open Access Journals (Sweden)

    Dai Yang

    2003-09-01

    Full Text Available Being able to transmit the audio bitstream progressively is a highly desirable property for network transmission. MPEG-4 version 2 audio supports fine grain bit rate scalability in the generic audio coder (GAC. It has a bit-sliced arithmetic coding (BSAC tool, which provides scalability in the step of 1 Kbps per audio channel. There are also several other scalable audio coding methods, which have been proposed in recent years. However, these scalable audio tools are only available for mono and stereo audio material. Little work has been done on progressive coding of multichannel audio sources. MPEG advanced audio coding (AAC is one of the most distinguished multichannel digital audio compression systems. Based on AAC, we develop in this work a progressive syntax-rich multichannel audio codec (PSMAC. It not only supports fine grain bit rate scalability for the multichannel audio bitstream but also provides several other desirable functionalities. A formal subjective listening test shows that the proposed algorithm achieves an excellent performance at several different bit rates when compared with MPEG AAC.

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

    OpenAIRE

    Pradeep Kumar Singh; R.K.Aggrawal

    2010-01-01

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

  15. 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...

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

    DEFF Research Database (Denmark)

    Ljusev, Petar; Andersen, Michael Andreas E.

    2005-01-01

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

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

    Science.gov (United States)

    2010-10-01

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

  18. Efficient audio signal processing for embedded systems

    Science.gov (United States)

    Chiu, Leung Kin

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

  19. 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.

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

    DEFF Research Database (Denmark)

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

    2014-01-01

    , annoyance, balance and blend, and confusion. Ratings using these attributes were collected in the fourth stage, and a principal component analysis performed. This suggested two dimensions underlying the perception of an audio-on-audio interference situation: The first dimension was labeled “distraction......” and accounted for 89% of the variance; the second dimension, accounting for 10% of the variance, was labeled “balance and blend.” © 2014 Acoustical Society of America...

  1. Embedded Design of Power Line Carrier Communication Module and Its Intelligent Application%嵌入式电力线载波通信模块设计及其智能应用

    Institute of Scientific and Technical Information of China (English)

    曾素琼

    2012-01-01

    In this paper, the module system of low- voltage power line carrier communication is designed based on LPC2132 and ST7538. The implementation of system is discussed mainly. The hardware and software are designed for the modular system. Finally, the application of the module on the intelligent home are designed and analysed. Module test:+ 12V power supply, communication speed: 4800bps, each frame length: 128Byte, power line carrier frequency: 82+0. 3kHz, the communication distance: about 500m, through experiments, the module apply successful on the intelligent home. The modular system is added only a small number of components, is added the control chip, which can be conveniently applied to narrow-band signals in the low -voltage power line carrier communication occasions. The design has the advantades of simple constructure, flexible operation mode, reliable communication, anti-interference ability etc.%设计了基于LPC2132与 ST7538低压电力线载波通信模块系统,重点介绍系统的实现过程;对模块系统作硬件和软件设计,对模块在智能家居上的应用作设计及分析;模块应用试验:±12V供电,通信速率:4800bps,每帧长度:128Byte,电力线载波频率:82±0.3kHz,通信距离:约500m,通过实验,模块成功地应用在智能家居上;模块应用时只需加少量元器件、控制芯片,可方便地应用于窄带低压电力线载波通信各场合,设计具有结构简单、工作方式灵活、可靠、抗干扰能力强等特点.

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

    DEFF Research Database (Denmark)

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

    2014-01-01

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

  3. Research and Development Platform for Multimedia Streaming of MP3 Audio Content

    Directory of Open Access Journals (Sweden)

    Andrei Novak

    2006-07-01

    Full Text Available In the recent years, the MPEG Layer III (MP3 music compression format hasbecome an extremely popular choice for digital audio compression. Its high compressionratio, and near CD quality sound make it a natural choice for storing and distributingmusic - especially over the internet, where space and bandwidth are importantconsiderations. For example, using MPEG Layer-3 compression, 40 MBytes audio fileshave been compressed to approximately 3.5 MBytes. As a result of the MP3 popularity, avariety of portable MP3 players entered the market. We decided to design and implement aHard Disk based MP3 player similar to products currently available (e.g. Creative LabsNomad, Archos Jukebox 6000, Apple Ipod, etc.. Our goal was to design the player withminimal cost and to implement a FM Stereo Radio Transmitter module for ease ofconnectivity. This module resolves the compatibility problems with the current availablecar audio systems. In the same time system flexibility and scalability as well as systemevolution to more advanced architectures were the main principles that drove thedevelopment of this platform. The primary enhancement of the platform will be to switchthe communication module from a analogical FM radio transmitter to digital wired orwireless communication solutions.

  4. Radio Science Measurements with Suppressed Carrier

    Science.gov (United States)

    Asmar, Sami; Divsalar, Dariush; Oudrhiri, Kamal

    2013-01-01

    Radio Science started when it became apparent with early Solar missions that occultations by planetary atmospheres would affect the quality of radio communications. Since then the atmospheric properties and other aspects of planetary science, solar science, and fundamental physics were studied by scientists. Radio Science data was always extracted from a received pure residual carrier (without data modulation). For some missions, it is very desirable to obtain Radio Science data from a suppressed carrier modulation. In this paper we propose a method to extract Radio Science data when a coded suppressed carrier modulation is used in deep space communications. Type of modulation can be BPSK, QPSK, OQPSK, MPSK or even GMSK. However we concentrate mostly on BPSK modulation. The proposed method for suppressed carrier simply tries to wipe out data that acts as an interference for Radio Science measurements. In order to measure the estimation errors in amplitude and phase of the Radio Science data we use Cramer-Rao bound (CRB). The CRB for the suppressed carrier modulation with non-ideal data wiping is then compared with residual carrier modulation under the same noise condition. The method of derivation of CRB for non-ideal data wiping is an innovative method that presented here. Some numerical results are provided for coded system.

  5. 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.

  6. A Physiologically Inspired Method for Audio Classification

    Directory of Open Access Journals (Sweden)

    David V. Anderson

    2005-06-01

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

  7. Integrated 10 Gb/s multilevel multiband passive optical network and 500 Mb/s indoor visible light communication system based on Nyquist single carrier frequency domain equalization modulation.

    Science.gov (United States)

    Wang, Yuanquan; Shi, Jianyang; Yang, Chao; Wang, Yiguang; Chi, Nan

    2014-05-01

    We propose and experimentally demonstrate a novel integrated passive optical network (PON) and indoor visible light communication (VLC) system based on Nyquist single carrier frequency domain equalization (N-SC-FDE) modulation with direct detection. In this system, a directly modulated laser and a commercially available red light emitting diode are served as the transmitters of the PON and VLC, respectively. To enable high spectral efficiency, high-speed transmission, and flexible multiple access with simplified optical network unit-side digital signal processing, multilevel, multiband quadrature amplitude modulations 128/64/16 are implemented here. VLC N-SC-FDE signals are successfully delivered a further 30 cm indoor distance after transmitting over a span of 40 km single mode fiber (SMF) together with 3 sub-band PON signals. As a proof of concept, a 10 Gb/s PON and 500 Mb/s VLC integrated system for three wired users and one wireless user is successfully achieved, which shows the promising potential and feasibility of this proposal to extend multiple services from metropolitan to suburban areas.

  8. Museum audio guides as an accessibility enhancer

    OpenAIRE

    Martins, Cláudia Susana Nunes

    2012-01-01

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

  9. PDE-SVD Based Audio Denoising

    OpenAIRE

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

    2012-01-01

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

  10. Indexing spoken audio by LSA and SOMs

    OpenAIRE

    2000-01-01

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

  11. 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.

  12. Capacity-optimized mp2 audio watermarking

    Science.gov (United States)

    Steinebach, Martin; Dittmann, Jana

    2003-06-01

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

  13. Hydrogen carriers

    Science.gov (United States)

    He, Teng; Pachfule, Pradip; Wu, Hui; Xu, Qiang; Chen, Ping

    2016-12-01

    Hydrogen has the potential to be a major energy vector in a renewable and sustainable future energy mix. The efficient production, storage and delivery of hydrogen are key technical issues that require improvement before its potential can be realized. In this Review, we focus on recent advances in materials development for on-board hydrogen storage. We highlight the strategic design and optimization of hydrides of light-weight elements (for example, boron, nitrogen and carbon) and physisorbents (for example, metal-organic and covalent organic frameworks). Furthermore, hydrogen carriers (for example, NH3, CH3OH-H2O and cycloalkanes) for large-scale distribution and for on-site hydrogen generation are discussed with an emphasis on dehydrogenation catalysts.

  14. Audio Steganography Techniques-A Survey

    OpenAIRE

    Navneet Kaur; Sunny Behal

    2014-01-01

    we can communicate with each other by passing messages which is not secure, but we make a communication be kept secret by embedding the message into carrier or by special tools such as invisible ink, microdots etc. Steganography is the science that involves communicating secret data in an appropriate carrier which is used from hundreds of years. In digital age new techniques of hiding the data inside the carrier are invented which are known as digital steganography. Nowadays, t...

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

    DEFF Research Database (Denmark)

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

    2013-01-01

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

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

    OpenAIRE

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

    2016-01-01

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

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

    Science.gov (United States)

    Martin, R R; Haroldson, S K

    1992-06-01

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

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

    Institute of Scientific and Technical Information of China (English)

    2011-01-01

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

  19. A content-based digital audio watermarking algorithm

    Science.gov (United States)

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

    2015-12-01

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

  20. 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.

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

    Institute of Scientific and Technical Information of China (English)

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

    2004-01-01

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

  2. The suppressor of AAC2 Lethality SAL1 modulates sensitivity of heterologously expressed artemia ADP/ATP carrier to bongkrekate in yeast.

    Science.gov (United States)

    Wysocka-Kapcinska, Monika; Torocsik, Beata; Turiak, Lilla; Tsaprailis, George; David, Cynthia L; Hunt, Andrea M; Vekey, Karoly; Adam-Vizi, Vera; Kucharczyk, Roza; Chinopoulos, Christos

    2013-01-01

    The ADP/ATP carrier protein (AAC) expressed in Artemia franciscana is refractory to bongkrekate. We generated two strains of Saccharomyces cerevisiae where AAC1 and AAC3 were inactivated and the AAC2 isoform was replaced with Artemia AAC containing a hemagglutinin tag (ArAAC-HA). In one of the strains the suppressor of ΔAAC2 lethality, SAL1, was also inactivated but a plasmid coding for yeast AAC2 was included, because the ArAACΔsal1Δ strain was lethal. In both strains ArAAC-HA was expressed and correctly localized to the mitochondria. Peptide sequencing of ArAAC expressed in Artemia and that expressed in the modified yeasts revealed identical amino acid sequences. The isolated mitochondria from both modified strains developed 85% of the membrane potential attained by mitochondria of control strains, and addition of ADP yielded bongkrekate-sensitive depolarizations implying acquired sensitivity of ArAAC-mediated adenine nucleotide exchange to this poison, independent from SAL1. However, growth of ArAAC-expressing yeasts in glycerol-containing media was arrested by bongkrekate only in the presence of SAL1. We conclude that the mitochondrial environment of yeasts relying on respiratory growth conferred sensitivity of ArAAC to bongkrekate in a SAL1-dependent manner.

  3. On Steganography in Lost Audio Packets

    CERN Document Server

    Mazurczyk, Wojciech; Szczypiorski, Krzysztof

    2011-01-01

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

  4. Non Audio-Video gesture recognition system

    DEFF Research Database (Denmark)

    Craciunescu, Razvan; Mihovska, Albena Dimitrova; Kyriazakos, Sofoklis

    2016-01-01

    Gesture recognition is a topic in computer science and language technology with the goal of interpreting human gestures via mathematical algorithms. Gestures can originate from any bodily motion or state but commonly originate from the face or hand. Current research focus includes on the emotion...... recognition from the face and hand gesture recognition. Gesture recognition enables humans to communicate with the machine and interact naturally without any mechanical devices. This paper investigates the possibility to use non-audio/video sensors in order to design a low-cost gesture recognition device...

  5. Audio frequency in vivo optical coherence elastography

    Energy Technology Data Exchange (ETDEWEB)

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

    2009-05-21

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

  6. Mixing audio concepts, practices and tools

    CERN Document Server

    Izhaki, Roey

    2013-01-01

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

  7. Audio marketing v ČR

    OpenAIRE

    Timanov, Vladimir

    2015-01-01

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

  8. 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

  9. A mixture of oleic, erucic and conjugated linoleic acids modulates cerebrospinal fluid inflammatory markers and improve somatosensorial evoked potential in X-linked adrenoleukodystrophy female carriers.

    Science.gov (United States)

    Cappa, Marco; Bizzarri, Carla; Petroni, Anna; Carta, Gianfranca; Cordeddu, Lina; Valeriani, Massimiliano; Vollono, Catello; De Pasquale, Loredana; Blasevich, Milena; Banni, Sebastiano

    2012-09-01

    X-linked adrenoleukodystrophy is a rare inherited demyelinating disorder characterized by an abnormal accumulation of very long chain fatty acids, mainly hexacosanoic acid (26:0), due to a mutation of the gene encoding for a peroxisomal membrane protein. The only available, and partially effective, therapeutic treatment consists of dietary intake of a 4:1 mixture of triolein and trierucin, called Lorenzo's oil (LO), targeted to inhibit the elongation of docosanoic acid (22:0) to 26:0. In this study we tested whether, besides inhibiting elongation, an enhancement of peroxisomal beta oxidation induced by conjugated linoleic acid (CLA), will improve somatosensory evoked potentials and modify inflammatory markers in adrenoleukodystrophy females carriers. We enrolled five heterozygous women. They received a mixture of LO (40 g/day) with CLA (5 g/day) for 2 months. The therapeutic efficacy was evaluated by the means of plasma levels of 26:0, 26:0/22:0 ratio, modification of cerebrospinal fluid (CSF) inflammatory markers and somatosensory evoked potentials. Changes of fatty acid profile, and in particular CLA incorporation, were also evaluated in CSF and plasma. The results showed that CLA promptly passes the blood brain barrier and the mixture was able to lower both 26:0 and 26:0/22:0 ratio in plasma. The mixture improved somatosensory evoked potentials, which were previously found unchanged or worsened with dietary LO alone, and reduced IL-6 levels in CSF in three out of five patients. Our data suggest that the synergic activity of CLA and LO, by enhancing peroxisomal beta-oxidation and preventing 26:0 formation, improves the somatosensory evoked potentials and reduces neuroinflammation.

  10. 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.

  11. A photovoltaic module

    OpenAIRE

    Krebs, Frederik C.; Sommer-Larsen, Peter

    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 photovolt...

  12. 基于光载波抑制调制的星间微波光子下变频研究%Research on inter-satellite microwave photonic frequency down conversion based on optical carrier suppression modulation

    Institute of Scientific and Technical Information of China (English)

    李轩; 赵尚弘; 张薇; 朱子行; 韩磊; 赵静

    2013-01-01

    To solve the problem of optical handling of microwave signal in satellite communication,the inter-satellite microwave photonic frequency down conversion system is modeled,two parallel dual-electrode Mach-Zehnder modulators based on optical carrier suppression modulation are utilized to modulate the uplink microwave signal received by satellite and the local oscillator signal produced in satellite,respectively,and the microwave signal is optically amplified,transmitted and frequency-down converted in the inter-satellite optical link.The output signal and noise of system are analyzed with Bessel expansion,the local oscillator signal power is optimized,and the effects of modulator bias phase drift,phase shifter error and emission optical power on the system performance are simulated.The results show that the deterioration of output carrier to noise ratio (CNR) is under 0.05 dB while the modulator bias phase drift is less than 5 ℃,the output CNR deterioration is under 0.02 dB while the phase shifter error is less than 5 ℃,and the frequency down conversion system has high stability.When the emission optical power is 10.48 dB,the system output CNR is 31.33 dB,which can meet the practical requirement.The inter-satellite microwave photonic frequency down conversion system can be applied to the optical handling of microwave signal in the future satellite optical communications.%针对卫星通信中微波信号光学处理问题,建立了星间微波光子下变频系统模型,采用两个双电极马赫-曾德尔调制器(DE-MZM)并联形式,以光载波抑制(DCS)方式实现了星间微波信号的光域放大、传输和下变频.利用贝塞尔函数展开分析了下变频系统中信号和各噪声分量,对射频本振信号功率进行了优化,仿真研究了调制器直流偏置漂移、移相器相移误差和发射光功率对系统性能的影响.结果表明,调制器直流偏置相位漂移小于5℃时输出载噪比(CNR)恶化小于0.05 dB

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

    Science.gov (United States)

    Baggaley, Jon; Klaas, Jim

    2006-01-01

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

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

    DEFF Research Database (Denmark)

    Knott, Arnold; Ploug, Rasmus Overgaard

    2012-01-01

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

  15. 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...

  16. 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...

  17. DOA Estimation of Audio Sources in Reverberant Environments

    DEFF Research Database (Denmark)

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

    2016-01-01

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

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

    NARCIS (Netherlands)

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

    2012-01-01

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

  19. 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

  20. Voice activity detection using audio-visual information

    DEFF Research Database (Denmark)

    Petsatodis, Theodore; Pnevmatikakis, Aristodemos; Boukis, Christos

    2009-01-01

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

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

    Science.gov (United States)

    McCrocklin, Shannon

    2012-01-01

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

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

    Institute of Scientific and Technical Information of China (English)

    XU Shuzheng; ZHANG Peng; WANG Pengjun; YANG Huazhong

    2009-01-01

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

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

    NARCIS (Netherlands)

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

    2008-01-01

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

  4. An Audio Stream Redirector for the Ethernet Speaker

    Science.gov (United States)

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

    2004-01-01

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

  5. Circular microphone array for multi channel audio recording

    NARCIS (Netherlands)

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

    2004-01-01

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

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

    NARCIS (Netherlands)

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

    2008-01-01

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

  7. The Role of pH in Modulating the Electronic State Properties of Minocycline Drug and Its Inclusion within Micellar Carriers.

    Science.gov (United States)

    Clementi, Catia; Cesaretti, Alessio; Carlotti, Benedetta; Elisei, Fausto

    2016-07-14

    A detailed investigation of the spectral and photophysical properties of minocycline (MC) in water at different pHs, solvents of different polarity, and micellar surfactant solutions was carried out in this study. An unusual behavior was highlighted with respect to other tetracyclines due to the presence of an additional dimethylamino group in the MC molecular structure. In particular, four equilibrium constants associated with mono-deprotonation reactions were characterized by steady-state spectroscopy. Femtosecond time-resolved pump-probe and fluorescence up-conversion measurements allowed the dynamics of the lowest excited singlet state of the five different acid-base species of MC to be characterized in terms of lifetimes and transient spectra. Two emissive species associated with keto-enol tautomerism resulting from excited-state intramolecular proton transfer (ESIPT) were revealed with time constants of a few and tens of picoseconds. TD-DFT quantum mechanical calculations were also performed to define the state order and nature of the differently protonated species, together with their absorption spectra. The role of pH proved to be fundamental in modulating the drug charge and therefore the interaction with cationic micelles where the neutral form of MC, that is the biologically active one, resulted efficiently included.

  8. Dynamic Bayesian Networks for Audio-Visual Speech Recognition

    Directory of Open Access Journals (Sweden)

    Liang Luhong

    2002-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Haider Ismael Shahadi

    2011-09-01

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

  10. 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…

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

    NARCIS (Netherlands)

    Pocta, P.; Beerends, J.G.

    2015-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Frank Kurth

    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.

  13. A power supply error correction method for single-ended digital audio class D amplifiers

    Science.gov (United States)

    Yu, Zeqi; Wang, Fengqin; Fan, Yangyu

    2016-12-01

    In single-ended digital audio class D amplifiers (CDAs), the errors caused by power supply noise in the power stages degrade the output performance seriously. In this article, a novel power supply error correction method is proposed. This method introduces the power supply noise of the power stage into the digital signal processing block and builds a power supply error corrector between the interpolation filter and the uniform-sampling pulse width modulation (UPWM) lineariser to pre-correct the power supply error in the single-ended digital audio CDA. The theoretical analysis and implementation of the method are also presented. To verify the effectiveness of the method, a two-channel single-ended digital audio CDA with different power supply error correction methods is designed, simulated, implemented and tested. The simulation and test results obtained show that the method can greatly reduce the error caused by the power supply noise with low hardware cost, and that the CDA with the proposed method can achieve a total harmonic distortion + noise (THD + N) of 0.058% for a -3 dBFS, 1 kHz input when a 55 V linear unregulated direct current (DC) power supply (with the -51 dBFS, 100 Hz power supply noise) is used in the power stages.

  14. 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.

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

    Directory of Open Access Journals (Sweden)

    Beracoechea JA

    2006-01-01

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

  16. Enhanced Audio LSB Steganography for Secure Communication

    Directory of Open Access Journals (Sweden)

    Muhammad Junaid Hussain

    2016-01-01

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

  17. Particle Filtering on the Audio Localization Manifold

    CERN Document Server

    Ettinger, Evan

    2010-01-01

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

  18. A direct broadcast satellite-audio experiment

    Science.gov (United States)

    Vaisnys, Arvydas; Abbe, Brian; Motamedi, Masoud

    1992-03-01

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

  19. Time-Scale Invariant Audio Data Embedding

    Directory of Open Access Journals (Sweden)

    Mansour Mohamed F

    2003-01-01

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

  20. Optimization of audio - ultrasonic plasma system parameters

    Science.gov (United States)

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

    2016-10-01

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

  1. 三电平空间矢量与载波调制策略统一理论的研究%Research of the Unity Theory Between Three-level Space Vector and Carrier-based PWM Modulation Strategy

    Institute of Scientific and Technical Information of China (English)

    陈娟; 何英杰; 王新宇; 刘进军

    2013-01-01

    There is an inherent correlation between the carrier-based PWM and SVPWM in two level inverters.Some researchers have proved that this inherent relationship also exists between those two modulation strategies in three level inverters.In the former researches,the unity correlation was focused on the traditional 8 segments SVPWM switching sequences,but it couldn't be applied to some switching sequences more than 8 segments obtained by using redundancy vectors in the practical SVPWM modulation strategy.So far,seldom researchers have the unity test of SVPWM and SPWM under these conditions,so the existing unity theory between SVPWM and SPWM contains some shortage.In this paper,a technique of decomposing the original modulation wave into two sub-modulation waves has been proposed which is utilized to discuss the unity relations between SPWM and SVPWM with more than 8 segments.The strict theoretical derivation and simulation result effectively evolves the unity theory of SPWM and SVPWM successfully.%两电平空间矢量脉宽调制(space vector pulse width modulation,SVPWM)与正弦脉宽调制(sinusoidal pulse width modulation,SPWM)存有一定的本质关系,对于三电平,在常见的8段式SVPWM序列中可以证明,SVPWM与SPWM本质上也存在某种统一.实际的SVPWM调制中,由于要考虑中点电压平衡、减少谐波等其它性能问题,一般会利用开关冗余状态得到8段以上的开关调制序列.目前的文献中还未曾有对这些情况进行统一性地验证,因此三电平SVPWM与SPWM的统一理论存在一定的不足.研究发现,可通过严格地理论推导将传统意义上的调制波分解为2个子调制波,再进行SPWM调制,可以得到与SVPWM完全吻合的序列.该文在此方法基础上深入研究了8段以上(10、12、14段)的SVPWM调制序列与SPWM的统一.经过严格地理论推导,成功实现了含任意段数的SVPWM序列与SPWM的统一.仿真结果证明了该理论的正确性.

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

    Science.gov (United States)

    Koenig, Bruce E; Lacey, Douglas S

    2012-01-01

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

  3. Quality Enhancement of Compressed Audio Based on Statistical Conversion

    Directory of Open Access Journals (Sweden)

    Mouchtaris Athanasios

    2008-01-01

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

  4. Quality Enhancement of Compressed Audio Based on Statistical Conversion

    Directory of Open Access Journals (Sweden)

    Chris Kyriakakis

    2008-07-01

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

  5. Sampling Function of Degree 2 for DVD-Audio

    Science.gov (United States)

    Toraichi, Kazuo; Nakamura, Koji

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

  6. 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.

  7. Beyond podcasting: creative approaches to designing educational audio

    Directory of Open Access Journals (Sweden)

    Andrew Middleton

    2009-12-01

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

  8. 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

  9. Lattice Vector Quantization Applied to Speech and Audio Coding

    Institute of Scientific and Technical Information of China (English)

    Minjie Xie

    2012-01-01

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

  10. A Review on Audio-visual Translation Studies

    Institute of Scientific and Technical Information of China (English)

    李瑶

    2008-01-01

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

  11. Aurally Aided Visual Search Performance Comparing Virtual Audio Systems

    DEFF Research Database (Denmark)

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

    2014-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Mohammed S. Altaei

    2012-04-01

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

  13. Robust message authentication code algorithm for digital audio recordings

    Science.gov (United States)

    Zmudzinski, Sascha; Steinebach, Martin

    2007-02-01

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

  14. 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

  15. 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....

  16. Aurally Aided Visual Search Performance Comparing Virtual Audio Systems

    DEFF Research Database (Denmark)

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

    2014-01-01

    Due to increased computational power, reproducing binaural hearing in real-time applications, through usage of head-related transfer functions (HRTFs), is now possible. This paper addresses the differences in aurally-aided visual search performance between a HRTF enhanced audio system (3D......) and an amplitude panning audio system (panning) in a virtual environment. We present a performance study involving 33 participants locating aurally-aided visual targets placed at fixed positions, under different audio conditions. A varying amount of visual distractors were present, represented as black circles...

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

    NARCIS (Netherlands)

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

    2009-01-01

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

  18. 37 CFR 201.27 - Initial notice of distribution of digital audio recording devices or media.

    Science.gov (United States)

    2010-07-01

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

  19. Isothermal Containment Module

    Science.gov (United States)

    1999-01-01

    Isothermal Containment Modules are the temperature-controlling carrier that BioServe built to carry Commercial Generic Bioprocessing Apparatus (CGBA) and in the future, Space Automated Bioproduct Lab (SABL) to the International Space Station.

  20. 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...

  1. Reception of infrasound and audio current in derma nerves

    Institute of Scientific and Technical Information of China (English)

    Jianwen Li; Ziyu Li; Xuezong Ma

    2010-01-01

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

  2. Audio CAPTCHA for SIP-Based VoIP

    Science.gov (United States)

    Soupionis, Yannis; Tountas, George; Gritzalis, Dimitris

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

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

    NARCIS (Netherlands)

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

    2004-01-01

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

  4. 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)

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

    DEFF Research Database (Denmark)

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

    Introduction In order to give patients possibility to listen to their consultation again, we have designed a system which gives the patients access to digital audio recordings of their consultations. An Interactive Voice Response platform enables the audio recording and gives the patients access...... to replay their consultation. The intervention is evaluated in a randomised controlled trial with 5.460 patients in order to determine whether providing patients with digital audio recording of the consultation affects the patients overall perception of their consultation. In addition to this primary...... objective we want to investigate if replay of the consultations improves the patients’ recall of the information given. Methods Interviews are carried out with 40 patients whose consultations have been audio recorded. Patients are divided into two groups, those who have listened to their consultation...

  6. Simple PWM modulator topology with excellent dynamic behavior

    DEFF Research Database (Denmark)

    Poulsen, Søren; Andersen, Michael Andreas E.

    2004-01-01

    This paper proposes a new PWM modulator topology. The modulator is used in switch mode audio power amplifiers, but the topology can be used in a wide range of applications. Due to excellent transient behavior, the modulator is very suited for VRMs or other types of DC-DC or DC-AC applications....

  7. A New Audio Steganography System Based on Auto-Key Generator

    Directory of Open Access Journals (Sweden)

    Inas Jawad Kadhim

    2012-01-01

    Full Text Available Stenography is the art of hiding the very presence of communication by embedding secret message into innocuous looking cover document, such as digital image, videos, sound files, and other computer files that contain perceptually irrelevant or redundant information as covers or carriers to hide secret messages.In this paper, a new Least Significant Bit (LSB nonsequential embedding technique in wave audio files is introduced. To support the immunity of proposed hiding system, and in order to recover some weak aspect inherent with the pure implementation of stego-systems, some auxiliary processes were suggested and investigated including the use of hidden text jumping process and stream ciphering algorithm. Besides, the suggested system used self crypto-hiding pseudo random key generator. The auto-key generator has purposes to investigate the encryption and embedding processes .The hiding results shows no noise in the stego-wave file after embedding process, also no difference in size is found between the original wave audio file and stego-wave file.

  8. Audio-Visual Integration of Emotional Information

    Directory of Open Access Journals (Sweden)

    Penny Bergman

    2011-10-01

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

  9. Audio-visual voice activity detection

    Institute of Scientific and Technical Information of China (English)

    LIU Peng; WANG Zuo-ying

    2006-01-01

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

  10. Multi-Level Audio Classification Architecture

    Directory of Open Access Journals (Sweden)

    Jozef Vavrek

    2015-01-01

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

  11. 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.

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

    Science.gov (United States)

    Sánchez, J; Lumbreras, M

    1999-01-01

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

  13. 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...

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

    DEFF Research Database (Denmark)

    Dimitrov, Smilen; Serafin, Stefania

    2011-01-01

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

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

    DEFF Research Database (Denmark)

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

    2015-01-01

    listener can be viewed as having a personal sound zone system. In order to evaluate and optimize such situations in a perceptually relevant manner, the authors created a predictive model using the features that contribute to the distraction from unwanted sounds. Feature extraction was motivated......There are many situations in which multiple audio programs are replayed over loudspeakers in the same acoustic environment, allowing listeners to focus on their desired target program. Where this situation is deliberately created and the different program items are centrally controlled, each...... separation, and frequency content of the interferer. The model was found to predict accurately for the training and validation datasets....

  16. 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.

  17. Collection of Digital Audio-visual Material Preservation and Backup Data Transfer%典藏音像资料保存与数字化备份转移

    Institute of Scientific and Technical Information of China (English)

    李浚

    2011-01-01

    According to the audio and video material carrier form, storage media technical features type classification accord- ing to different types of collection, audio-visual materials of corresponding preserving method proposed. In audio and video material carrier storage life could not infinite long cases, and many early video data broadcast devices will be eliminated, causing many valuable audio-visual material will collapse of reality, audio-visual materials need to put forward the urgency views. Finally talk about how video data provide detailed digital transfer methods.%根据音像资料载体形式、存储媒介技术特点进行类型划分,针对不同类型的典藏音像资料提出各种相应的保存方法。在音像资料载体保存期不可能无限长的情况下,以及很多早期音像资料播放设备即将被淘汰,致使许多珍贵声像资料面·临无法使用的现实,为此提出音像资料迫切需要数字化的观点。最后为音像资料怎样数字化转移提供了详细方法

  18. 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.

  19. Susceptibility study of audio recording devices to electromagnetic stimulations

    Energy Technology Data Exchange (ETDEWEB)

    Halligan, Matthew S.; Grant, Steven.

    2014-02-01

    Little research has been performed to study how intentional electromagnetic signals may couple into recording devices. An electromagnetic susceptibility study was performed on an analog tape recorder, a digital video camera, a wired computer microphone, and a wireless microphone system to electromagnetic interference. Devices were subjected to electromagnetic stimulations in the frequency range of 1-990 MHz and field strengths up to 4.9 V/m. Carrier and message frequencies of the stimulation signals were swept, and the impacts of device orientation and antenna polarization were explored. Message signals coupled into all devices only when amplitude modulated signals were used as stimulation signals. Test conditions that produced maximum sensitivity were highly specific to each device. Only narrow carrier frequency ranges could be used for most devices to couple messages into recordings. A basic detection technique using cross-correlation demonstrated the need for messages to be as long as possible to maximize message detection and minimize detection error. Analysis suggests that detectable signals could be coupled to these recording devices under realistic ambient conditions.

  20. Carrier-dependent temporal processing in an auditory interneuron.

    Science.gov (United States)

    Sabourin, Patrick; Gottlieb, Heather; Pollack, Gerald S

    2008-05-01

    Signal processing in the auditory interneuron Omega Neuron 1 (ON1) of the cricket Teleogryllus oceanicus was compared at high- and low-carrier frequencies in three different experimental paradigms. First, integration time, which corresponds to the time it takes for a neuron to reach threshold when stimulated at the minimum effective intensity, was found to be significantly shorter at high-carrier frequency than at low-carrier frequency. Second, phase locking to sinusoidally amplitude modulated signals was more efficient at high frequency, especially at high modulation rates and low modulation depths. Finally, we examined the efficiency with which ON1 detects gaps in a constant tone. As reflected by the decrease in firing rate in the vicinity of the gap, ON1 is better at detecting gaps at low-carrier frequency. Following a gap, firing rate increases beyond the pre-gap level. This "rebound" phenomenon is similar for low- and high-carrier frequencies.

  1. Modulation masking produced by second-order modulators

    DEFF Research Database (Denmark)

    Füllgrabe, Christian; Moore, Brian C.J.; Demany, Laurent;

    2005-01-01

    Recent studies suggest that an auditory nonlinearity converts second-order sinusoidal amplitude modulation (SAM) (i.e., modulation of SAM depth) into a first-order SAM component, which contributes to the perception of second-order SAM. However, conversion may also occur in other ways......-carrier modulation frequency, phase relationship between the probe and masker modulator, and probe modulation depth. In experiment 1, the carrier was a 5-kHz sinusoid presented either alone or within a notched-noise masker in order to restrict off-frequency listening. In experiment 2, the carrier was a white noise....... The data obtained in both carrier conditions are consistent with the existence of a modulation distortion component. However, the phase yielding poorest detection performance varied across experimental conditions between 0° and 180°, confirming that, in addition to nonlinear mechanisms, cochlear filtering...

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

    Science.gov (United States)

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

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

  3. 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. 

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

    Directory of Open Access Journals (Sweden)

    Cheng Ji-Shiung

    2003-01-01

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

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

    OpenAIRE

    Garrido López, David

    2010-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Muhammad Shoaib

    2016-10-01

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

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

    Directory of Open Access Journals (Sweden)

    Lhoucine Bahatti

    2016-05-01

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

  8. Audio system using binaural synthesis for multimodal telepresence applications

    DEFF Research Database (Denmark)

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

    2013-01-01

    of microphones, headphones and loudspeakers as well as measurements of network latency and bandwidth requirements of the system. Furthermore, measurements were made to determine whether the level of echo and cross talk cause any issues. The overall system employs multiple modalities to virtually transport......An audio system was developed as part of a multimodal system aiming to go beyond current state of the art in telepresence.This paper provides an overview of how the audio was implemented and documents measurements that were performed on the audio system. The measurements include equalization...... a person (the visitor) to a different physical location (the destination). The goal is that both the visitor and the people physically at the destination (the locals) should be provided with a sensation that the visitor is really there. Both the general multimodal system and the auditory part...

  9. 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.

  10. 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...

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

    DEFF Research Database (Denmark)

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

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

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

    Directory of Open Access Journals (Sweden)

    Amir Reza Fazli

    2012-06-01

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

  13. 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....

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

    Science.gov (United States)

    Steinebach, Martin; Zmudzinski, Sascha

    2008-02-01

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

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

    Directory of Open Access Journals (Sweden)

    Urel Sawyers

    2005-11-01

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

  16. Audio Steganography Coding Using the Discrete Wavelet Transforms

    Directory of Open Access Journals (Sweden)

    Siwar Rekik

    2012-02-01

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

  17. 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

  18. DOA Estimation of Audio Sources in Reverberant Environments

    DEFF Research Database (Denmark)

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

    2016-01-01

    Reverberation is well-known to have a detrimental impact on many localization methods for audio sources. We address this problem by imposing a model for the early reflections as well as a model for the audio source itself. Using these models, we propose two iterative localization methods that est...... bias. Our simulation results show that we can estimate the DOA of the desired signal more accurately with this procedure compared to state-of-theart estimator in both synthetic and real data experiments with reverberation....

  19. 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.

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

    Directory of Open Access Journals (Sweden)

    Jon Baggaley

    2006-06-01

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

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

    Institute of Scientific and Technical Information of China (English)

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

    2011-01-01

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

  2. Peptide-Carrier Conjugation

    DEFF Research Database (Denmark)

    Hansen, Paul Robert

    2015-01-01

    To produce antibodies against synthetic peptides it is necessary to couple them to a protein carrier. This chapter provides a nonspecialist overview of peptide-carrier conjugation. Furthermore, a protocol for coupling cysteine-containing peptides to bovine serum albumin is outlined....

  3. FTGS Audio Track Circuit Tester Design and Implementation%FTGS音频轨道电路测试仪设计与实现

    Institute of Scientific and Technical Information of China (English)

    唐匀生; 黄斌

    2013-01-01

    In order to enhance learning interest and operation practice ability, and add Urban rail traffic control professional training and experiment contend.This paper design a set of FTGS audio track circuit testing system using STC1T single-chip microcomputer,it can measure FTGS audio track circuit voltage, carrier and bit pattern parameters and so on. And measure combination of two carrier and three kinds of bit patterns.The results show that the system can best measure FTGS audio track circuit information parameters, fully meet the students' learning and measuring use.%  为了增强学习兴趣和操作动手能力,增加城轨专业的实训、实验内容。利用STC1T单片机设计出一套FTGS音频轨道电路测试系统,可以测量FTGS音频轨道电路的电压、载频及位模式等参数,并对两种载频及3种位模式的组合进行了测量实验。结果表明,该系统能很好的测量FTGS音频轨道电路的信息参数,完全满足学生学习、测量使用。

  4. Synchronized audio-visual transients drive efficient visual search for motion-in-depth.

    Directory of Open Access Journals (Sweden)

    Marina Zannoli

    Full Text Available In natural audio-visual environments, a change in depth is usually correlated with a change in loudness. In the present study, we investigated whether correlating changes in disparity and loudness would provide a functional advantage in binding disparity and sound amplitude in a visual search paradigm. To test this hypothesis, we used a method similar to that used by van der Burg et al. to show that non-spatial transient (square-wave modulations of loudness can drastically improve spatial visual search for a correlated luminance modulation. We used dynamic random-dot stereogram displays to produce pure disparity modulations. Target and distractors were small disparity-defined squares (either 6 or 10 in total. Each square moved back and forth in depth in front of the background plane at different phases. The target's depth modulation was synchronized with an amplitude-modulated auditory tone. Visual and auditory modulations were always congruent (both sine-wave or square-wave. In a speeded search task, five observers were asked to identify the target as quickly as possible. Results show a significant improvement in visual search times in the square-wave condition compared to the sine condition, suggesting that transient auditory information can efficiently drive visual search in the disparity domain. In a second experiment, participants performed the same task in the absence of sound and showed a clear set-size effect in both modulation conditions. In a third experiment, we correlated the sound with a distractor instead of the target. This produced longer search times, indicating that the correlation is not easily ignored.

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

    OpenAIRE

    Saleh Al-Zhrani; Mubarak AlQahtani

    2010-01-01

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

  6. The Design of Car Digital Audio System%车载数字音频系统设计

    Institute of Scientific and Technical Information of China (English)

    徐登峰

    2013-01-01

      With the development of electronic technology ,automotive electronics is a hot issue .To design a multi-vehicle digital audio sys-tem has great market value .Information broadcast system uses AU6860B microcontroller as the master chip ,to achieve a variety of external devices ,automatic detection function play USB and SD amount of storage devices ,MP3 and WMA audio files ,and interpolation methods and EEPROM to save various states .By internal structure of the state machine in oftware function modules ,the car audio system perfor-mance is greatly improved .%  随着电子技术发展,汽车电子成为了一个热点问题,设计一款多功能车载数字音频系统具有很大的市场价值。系统采用AU6860B单片机作主控芯片,能实现各种外部设备的自动检测功能播放USB、SD卡海量存储设备上的MP3、WMA音频文件,并采用后插先播方式和EEPROM保存各种状态的信息。软件功能模块内部采用状态机结构,使车载音频系统性能有较大的提高。

  7. 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 ...

  8. Project BIOTECH: Use of Modules in Technician Training

    Science.gov (United States)

    Glazer, Richard B.

    1974-01-01

    Describes Project Biotech, a program that utilizes the audio-tutorial principle to develop skill-oriented modules. The modules are self-pacing, independent units of instruction which concentrate on a few well-defined objectives and allow the student to learn at his own rate with minimal supervision. (PB)

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

    Science.gov (United States)

    Lee, Ki-Seung

    2008-08-01

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

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

    Science.gov (United States)

    Wormnaes, Siri; Sellaeg, Nina

    2013-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Raquel Viciana-Abad

    2014-05-01

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

  12. Audio and Video Reflections to Promote Social Justice

    Science.gov (United States)

    Boske, Christa

    2011-01-01

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

  13. Efficiency Optimization in Class-D Audio Amplifiers

    DEFF Research Database (Denmark)

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

    2015-01-01

    This paper presents a new power efficiency optimization routine for designing Class-D audio amplifiers. The proposed optimization procedure finds design parameters for the power stage and the output filter, and the optimum switching frequency such that the weighted power losses are minimized unde...

  14. Current-Driven Switch-Mode Audio Power Amplifiers

    DEFF Research Database (Denmark)

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

    2012-01-01

    The conversion of electrical energy into sound waves by electromechanical transducers is proportional to the current through the coil of the transducer. However virtually all audio power amplifiers provide a controlled voltage through the interface to the transducer. This paper is presenting a sw...

  15. Deep learning, audio adversaries, and music content analysis

    DEFF Research Database (Denmark)

    Kereliuk, Corey Mose; Sturm, Bob L.; Larsen, Jan

    2015-01-01

    We present the concept of adversarial audio in the context of deep neural networks (DNNs) for music content analysis. An adversary is an algorithm that makes minor perturbations to an input that cause major repercussions to the system response. In particular, we design an adversary for a DNN...

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

    NARCIS (Netherlands)

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

    2006-01-01

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

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

    Science.gov (United States)

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

    2014-01-01

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

  18. Output impedance and stability of audio power amplifiers

    NARCIS (Netherlands)

    Schaink, T.

    2006-01-01

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

  19. Comparative study of Audio-lingual method and CLT

    Institute of Scientific and Technical Information of China (English)

    2013-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Chen Shuixian

    2010-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Shuixian Chen

    2010-01-01

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

  2. An Audio-Visual Lecture Course in Russian Culture

    Science.gov (United States)

    Leighton, Lauren G.

    1977-01-01

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

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

    DEFF Research Database (Denmark)

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

    2013-01-01

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

  4. 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...

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

    Science.gov (United States)

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

    2016-01-01

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

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

    Science.gov (United States)

    Szarkowska, Agnieszka

    2013-01-01

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

  7. Real-time Loudspeaker Distance Estimation with Stereo Audio

    DEFF Research Database (Denmark)

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

    2015-01-01

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

  8. 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.

  9. Market potential for interactive audio-visual media

    NARCIS (Netherlands)

    Leurdijk, A.; Limonard, S.

    2005-01-01

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

  10. Digital audio recordings improve the outcomes of patient consultations

    DEFF Research Database (Denmark)

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

    2016-01-01

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

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

    Science.gov (United States)

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

    2012-04-01

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

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

    Energy Technology Data Exchange (ETDEWEB)

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

    1999-07-01

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

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

    Directory of Open Access Journals (Sweden)

    Petr Motlicek

    2013-01-01

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

  14. Composite cam carrier

    Energy Technology Data Exchange (ETDEWEB)

    Wicks, Christopher Donald; Madin, Mark Michael

    2017-03-14

    A cam carrier assembly includes a cylinder head having valves and a camshaft having lobes. A cam carrier has a first side coupled with the cylinder head engaging around the valves and a second side with bearing surfaces supporting the camshaft. A series of apertures extend between the first and second sides for the lobes to interface with the valves. The cam carrier is made of carbon fiber composite insulating the camshaft from the cylinder head and providing substantial weight reduction to an upper section of an associated engine.

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

    DEFF Research Database (Denmark)

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

    2016-01-01

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

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

    Science.gov (United States)

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

    2010-07-01

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

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

    Energy Technology Data Exchange (ETDEWEB)

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

    2010-07-15

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

  18. PAPR Reduction in OFDM Systems with Large Number of Sub-Carriers by Carrier Interferometry Approaches

    Institute of Scientific and Technical Information of China (English)

    HE Jian-hui; QUAN Zi-yi; MEN Ai-dong

    2004-01-01

    High Peak-to-Average Power Ratio (PAPR) is one of the major drawbacks of Orthogonal Frequency Division Multiplexing ( OFDM) systems. This paper presents the structures of the particular bit sequences leading to the maximum PAPR (PAPRmax) in Carrier-Interferometry OFDM (CI/OFDM) and Pseudo Orthogonal Carrier-Interferometry OFDM (PO-CI/OFDM) systems for Binary Phase Shift Keying (BPSK) modulation. Furthermore, the simulation and analysis of PAPRmax and PAPR cumulative distribution in CI/OFDM and PO-CI/OFDM systems with 2048 sub-carriers are presented in this paper. The results show that the PAPR of OFDM system with large number of sub-carriers reduced evidently via CI approaches.

  19. The carrier-generating analysis of MEMS gyroscope interface circuit

    Directory of Open Access Journals (Sweden)

    GuangMin Yuan

    2014-03-01

    Full Text Available In this paper, the main factors which influence the noise ratio of gyroscope output signal were analysed, according to the MEMS gyro interface circuit technology. A working principle of a carrier in the gyroscope circuit was discussed, the process formula of the carrier amplitude and frequency in the interface circuit of modulation and demodulation was deduced, and the error components lead-in from carrier to gyroscope circuit was distinguished. Several commonly used carrier-generating circuit schemes were analysed and compared, and a carrier-generating program in the interface circuits of the micro-gyroscope was designed, which was applied in a MEMS gyro developed by our laboratory. The measurement results show that the amplitude stability and frequency stability is 1.3 ppm and 12 ppm, respectively, meeting the performance requirements of carrier generating in the MEMS gyro circuit.

  20. The carrier-generating analysis of MEMS gyroscope interface circuit

    Science.gov (United States)

    Yuan, GuangMin; Yuan, Weizheng; Zhu, Xiaobo; Chang, HongLong

    2014-03-01

    In this paper, the main factors which influence the noise ratio of gyroscope output signal were analysed, according to the MEMS gyro interface circuit technology. A working principle of a carrier in the gyroscope circuit was discussed, the process formula of the carrier amplitude and frequency in the interface circuit of modulation and demodulation was deduced, and the error components lead-in from carrier to gyroscope circuit was distinguished. Several commonly used carrier-generating circuit schemes were analysed and compared, and a carrier-generating program in the interface circuits of the micro-gyroscope was designed, which was applied in a MEMS gyro developed by our laboratory. The measurement results show that the amplitude stability and frequency stability is 1.3 ppm and 12 ppm, respectively, meeting the performance requirements of carrier generating in the MEMS gyro circuit.

  1. 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)…

  2. 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"--

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

    Directory of Open Access Journals (Sweden)

    M.B.I. Reaz

    2012-04-01

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

  4. Photoinduced Transformation between Charge Carrier and Spin Carrier in Polymers

    Institute of Scientific and Technical Information of China (English)

    MEI Yuan; ZHAO Chang; SUN Xin

    2006-01-01

    By dynamical simulations, we show a transforming process between neutral soliton (spin carrier) and charged soliton (charge carrier) in polymers via photo-excitation, taking a polaron as the transitional bridge. It is photoinduced transformation between spin carrier and charge carrier. In this way, we demonstrate an access for polymers to be applied to spintronics.

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

    Energy Technology Data Exchange (ETDEWEB)

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

    2007-07-01

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

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

    Science.gov (United States)

    Rui, Rui; Bao, Chang-chun

    2013-03-01

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

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

    Science.gov (United States)

    Barton, Robert J., III; Chodura, Hartmut

    2000-07-01

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

  8. Practical Implementation and Error Analysis of PSCPWM-Based Switching Audio Power Amplifiers

    DEFF Research Database (Denmark)

    Christensen, Frank Schwartz; Frederiksen, Thomas Mansachs; Andersen, Michael Andreas E.;

    1999-01-01

    The paper presents an in-depth analysis of practical results for Parallel Phase-Shifted Carrier Pulse-Width Modulation (PSCPWM) - amplifier. Spectral analyses of error sources involved in PSCPWM are presented. The analysis is performed both by numerical means in MATLAB and by simulation in PSPICE......, followed by practical verification on a prototype. A toolbox for MATLAB has been developed to ease the complex analysis....

  9. The value of energy carriers

    NARCIS (Netherlands)

    Gool, W. van

    1987-01-01

    The value of energy carriers can be described thermodynamically by the amount of heat (enthalpy method) or work (exergy or availability method) that can be obtained from the carriers. Prices for energy carriers are used in economics to express their values. The prices for energy carriers are often r

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

    Directory of Open Access Journals (Sweden)

    Adit Jamdar

    2015-05-01

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

  11. Adaptive audio watermarking based on SNR in localized regions

    Institute of Scientific and Technical Information of China (English)

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

    2005-01-01

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

  12. “Wrapping” X3DOM around Web Audio API

    Directory of Open Access Journals (Sweden)

    Andreas Stamoulias

    2015-12-01

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

  13. Evaluation of embedded audio feedback on writing assignments.

    Science.gov (United States)

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

    2015-01-01

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

  14. Audio-visual interactions in product sound design

    Science.gov (United States)

    Özcan, Elif; van Egmond, René

    2010-02-01

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

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

    Directory of Open Access Journals (Sweden)

    Yin Bin

    2003-01-01

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

  16. A Robust Zero-Watermarking Algorithm for Audio

    Directory of Open Access Journals (Sweden)

    Jie Zhu

    2008-03-01

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

  17. Audio Sensing Aid based Wireless Microphone Emulation Attacks Detection

    Directory of Open Access Journals (Sweden)

    Wang Shan-shan

    2013-10-01

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

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

    Science.gov (United States)

    Gilchrist, N. H. C.

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

  19. Random Numbers Generated from Audio and Video Sources

    Directory of Open Access Journals (Sweden)

    I-Te Chen

    2013-01-01

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

  20. Audio Quality Assurance : An Application of Cross Correlation

    DEFF Research Database (Denmark)

    Jurik, Bolette Ammitzbøll; Nielsen, Jesper Asbjørn Sindahl

    2012-01-01

    We describe algorithms for automated quality assurance on content of audio files in context of preservation actions and access. The algorithms use cross correlation to compare the sound waves. They are used to do overlap analysis in an access scenario, where preserved radio broadcasts are used...... in research and annotated. They have been applied in a migration scenario, where radio broadcasts are to be migrated for long term preservation....

  1. Audio-magnetotelluric methods in reconnaissance geothermal exploration

    Science.gov (United States)

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

    1976-01-01

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

  2. Audio Signal Generator System Based On State Machines

    Institute of Scientific and Technical Information of China (English)

    王维喜

    2009-01-01

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

  3. Quality and Distortion Evaluation of Audio Signal by Spectrum

    OpenAIRE

    Er. Niranjan Singh; Dr. Bhupendra Verma

    2012-01-01

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

  4. Comparison of Linear Prediction Models for Audio Signals

    Directory of Open Access Journals (Sweden)

    van Waterschoot Toon

    2008-01-01

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

  5. 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.

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

    Directory of Open Access Journals (Sweden)

    N. A. Haleem

    2016-09-01

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

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

    Directory of Open Access Journals (Sweden)

    Rachel N Denison

    2013-01-01

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

  8. Cation disorder and epitaxial strain modulated Drude-Smith type terahertz conductivity and Hall-carrier switching in Ca1-x Ce x RuO3 thin films

    Science.gov (United States)

    Das, Sarmistha; Eswara Phanindra, V.; Santhosh Kumar, K.; Agarwal, Piyush; Dhaker, K. C.; Rana, D. S.

    2017-01-01

    The CaRuO3 is a non-Fermi liquid pseudo-cubic perovskite with a magnetic ground state on the verge of phase transition and it lies in the vicinity of the quantum critical point. To understand the sensitivity of its ground state, the effects of subtle aliovalent chemical disorder on the static and high frequency dynamic conductivity in the coherently strained structures were explored. The Ce-doped Ca1-x Ce x RuO3 (0  ⩽  x  ⩽  0.1) thin films were deposited on LaAlO3 (1 0 0) and SrTiO3 (1 0 0) substrates and studies for low-energy terahertz (THz) carrier dynamics, dc transport and Hall effect. These compositions exhibited a very effective and unusual Hall-carrier switching in both compressive and tensile strain induced epitaxial thin films. The dc resistivity depicts a switching from a non-Fermi liquid to a Fermi liquid behavior without any magnetic phase transition. A discernible and gradual crossover from Drude to Drude-Smith THz dynamic optical conductivity was observed while traversing from pure to 10% Ce-doped CaRuO3 films. Overall, a nearly Fermi liquid behavior, effective carrier switching and unusual features in THz conductivity, were all novel features realized for the first time in physically and/or chemically modified CaRuO3. These new phases highlight the novel subtleties and versatility of the systems lying near the quantum critical point.

  9. Intestinal solute carriers

    DEFF Research Database (Denmark)

    Steffansen, Bente; Nielsen, Carsten Uhd; Brodin, Birger

    2004-01-01

    A large amount of absorptive intestinal membrane transporters play an important part in absorption and distribution of several nutrients, drugs and prodrugs. The present paper gives a general overview on intestinal solute carriers as well as on trends and strategies for targeting drugs and/or pro...

  10. Autonomous component carrier selection

    DEFF Research Database (Denmark)

    Garcia, Luis Guilherme Uzeda; Pedersen, Klaus; Mogensen, Preben

    2009-01-01

    in local areas, basing our study case on LTE-Advanced. We present extensive network simulation results to demonstrate that a simple and robust interference management scheme, called autonomous component carrier selection allows each cell to select the most attractive frequency configuration; improving...

  11. 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.

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

    Directory of Open Access Journals (Sweden)

    Yu Fan

    2013-11-01

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

  13. 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.

  14. MPEG-2/4 Low-Complexity Advanced Audio Coding Optimization and Implementation on DSP

    Science.gov (United States)

    Wu, Bing-Fei; Huang, Hao-Yu; Chen, Yen-Lin; Peng, Hsin-Yuan; Huang, Jia-Hsiung

    This study presents several optimization approaches for the MPEG-2/4 Audio Advanced Coding (AAC) Low Complexity (LC) encoding and decoding processes. Considering the power consumption and the peripherals required for consumer electronics, this study adopts the TI OMAP5912 platform for portable devices. An important optimization issue for implementing AAC codec on embedded and mobile devices is to reduce computational complexity and memory consumption. Due to power saving issues, most embedded and mobile systems can only provide very limited computational power and memory resources for the coding process. As a result, modifying and simplifying only one or two blocks is insufficient for optimizing the AAC encoder and enabling it to work well on embedded systems. It is therefore necessary to enhance the computational efficiency of other important modules in the encoding algorithm. This study focuses on optimizing the Temporal Noise Shaping (TNS), Mid/Side (M/S) Stereo, Modified Discrete Cosine Transform (MDCT) and Inverse Quantization (IQ) modules in the encoder and decoder. Furthermore, we also propose an efficient memory reduction approach that provides a satisfactory balance between the reduction of memory usage and the expansion of the encoded files. In the proposed design, both the AAC encoder and decoder are built with fixed-point arithmetic operations and implemented on a DSP processor combined with an ARM-core for peripheral controlling. Experimental results demonstrate that the proposed AAC codec is computationally effective, has low memory consumption, and is suitable for low-cost embedded and mobile applications.

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

    Science.gov (United States)

    Wang, Fei; Xie, Zhaoxin; Chen, Zuo

    2014-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Gunjan Nehru

    2012-01-01

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

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

    Science.gov (United States)

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

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

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

    Directory of Open Access Journals (Sweden)

    Fei Wang

    2014-01-01

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

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

    Institute of Scientific and Technical Information of China (English)

    Zhou Ping; Liu Zhuo; Xia Liang

    2008-01-01

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

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

    OpenAIRE

    Puidokas, Vytenis

    2011-01-01

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

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

    DEFF Research Database (Denmark)

    Ljusev, Petar; Andersen, Michael Andreas E.

    2005-01-01

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

  2. 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

    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 and the propert......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...

  3. New modulation-based watermarking technique for video

    Science.gov (United States)

    Lemma, Aweke; van der Veen, Michiel; Celik, Mehmet

    2006-02-01

    Successful watermarking algorithms have already been developed for various applications ranging from meta-data tagging to forensic tracking. Nevertheless, it is commendable to develop alternative watermarking techniques that provide a broader basis for meeting emerging services, usage models and security threats. To this end, we propose a new multiplicative watermarking technique for video, which is based on the principles of our successful MASK audio watermark. Audio-MASK has embedded the watermark by modulating the short-time envelope of the audio signal and performed detection using a simple envelope detector followed by a SPOMF (symmetrical phase-only matched filter). Video-MASK takes a similar approach and modulates the image luminance envelope. In addition, it incorporates a simple model to account for the luminance sensitivity of the HVS (human visual system). Preliminary tests show algorithms transparency and robustness to lossy compression.

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

    DEFF Research Database (Denmark)

    Dimitrov, Smilen; Serafin, Stefania

    2011-01-01

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

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

    Science.gov (United States)

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

    2013-07-01

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

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

    Science.gov (United States)

    Buś, Szymon; Jedrzejewski, Konrad

    2015-09-01

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

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

    Science.gov (United States)

    Fletcher, R.

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

  8. Research of Mobile Radio Access Networking which Based on Optical Coherent Modulation Technique to Realize Base Station Carrier Remote via Optical Fiber%基于光学相干调制技术实现基站载波光纤拉远的无线移动接入网络研究与设计

    Institute of Scientific and Technical Information of China (English)

    李广

    2014-01-01

    分析了目前数字基带射频拉远组网架构的优势与弊端,提出了基于光学相干调制技术实现基站载波光纤拉远的无线移动接入网络架构,分析了其组网架构的优越性,给出了该接入网的具体实施方式。%In the paper,we analysed the advantages and disadvantages of the current? radio access networking architecture-dig-ital based band remote via optical fiber. We proposed a mobile radio access networking architecture which based on optical co-herent modulation technique to realize base station carrier remote via optical fiber.

  9. Sinusoidal Analysis-Synthesis of Audio Using Perceptual Criteria

    Directory of Open Access Journals (Sweden)

    Ted Painter

    2003-01-01

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

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

    CERN Document Server

    Moreau, Nicolas

    2013-01-01

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

  11. Inexpensive Audio Activities: Earbud-based Sound Experiments

    Science.gov (United States)

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

    2016-11-01

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

  12. Lost Audio Packets Steganography: The First Practical Evaluation

    CERN Document Server

    Mazurczyk, Wojciech

    2011-01-01

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

  13. Audio Hijack Pro万能录音机

    Institute of Scientific and Technical Information of China (English)

    2004-01-01

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

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

    Institute of Scientific and Technical Information of China (English)

    涂艳; 刘俊

    2016-01-01

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

  15. Audio-haptic interaction in simulated walking experiences

    DEFF Research Database (Denmark)

    Serafin, Stefania

    2011-01-01

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

  16. The complete guide to high-end audio

    CERN Document Server

    Harley, Robert

    2015-01-01

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

  17. A listening test system for automotive audio - listeners

    DEFF Research Database (Denmark)

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

    2007-01-01

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

  18. Development of Passenger Air Carriers

    Directory of Open Access Journals (Sweden)

    Igor Diminik

    2006-09-01

    Full Text Available The work presents the development of carriers in passengerair traffic, and the focus is on the development and operationsof carriers in chartered passenger transport. After the SecondWorld War, there were only scheduled air carriers. The need formass transport of tourists resulted in the development of chartercarriers or usage of scheduled carriers under different commercialconditions acceptable for tourism. Eventually also low-costcarriers appeared and they realize an increasing share in thepassenger transport especially in the aviation developed countries.

  19. 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.

  20. 用于手机的微型声频定向扬声器%Micro Audio Directional Loudspeaker for Mobile Phone

    Institute of Scientific and Technical Information of China (English)

    李学生; 徐利梅; 姜丽峰; 蒲刚; 陈敏; 许亮峰

    2012-01-01

    A method for realizing the micro audio directional loudspeaker in mobile phone is investigated to examine a novel auditory experience in this paper. This novel directional loudspeaker can provide private listening service by emitting the audible sound with high directivity along propagating axis. A prototype using Nokia N73 mobile phone as the package is constructed in which the emitter array is assembled by four PZT ceramic ultrasonic transducers with the diameter of 10mm. The emitter array is driven by a class D amplifier. The modulation algorithm is realized on a DSP platform on which the carrier frequency is equal to the resonant frequency of the emitter array. The testing results of impedance, frequency response and sound pressure level characteristics are described and analyzed. The directivity angle and power consumption of the prototype are measured. The experiment result indicates that the prototype size, power and the acoustic pressure level can satisfy the requirements for the mobile phone, thus the micro audio directional loudspeaker can be used for the portable multimedia devices.%提出了一种用于手机的微型声频定向扬声器实现方法,该新型扬声器可将声音控制在发声单元正前方轴向传播,能为使用者提供免提的局部声音服务.4只直径为(φ)10 mm的锆钛酸铅(PZT)压电陶瓷超声换能器组阵作为发声部件,换能器阵列谐振频率作为超声载波频率,数字信号处理器(DSP)芯片作为声频定向调制算法实现平台,D类超声功率放大器为阵列驱动器,诺基亚N73手机作为外壳,制作出了样机.对换能器单元及阵列阻抗、声压级频率响应等特性进行了测试和分析,对样机作了自解调可听声指向性测试和整机功耗测试.实验结果表明,样机尺寸、功率和声压级基本满足手机中的应用需求,为微型声频定向扬声器在便携式多媒体设备中的应用提供了实践依据.

  1. Zero-Crossing Disturbance Elimination and Spectrum Analysis of Single-Carrier Seven-Level SPWM

    DEFF Research Database (Denmark)

    Wu, Fengjiang; Feng, Fan; Duan, Jiandong;

    2015-01-01

    In this paper, a seven-level single-carrier and multi-modulation-wave sinusoidal pulsewidth modulation (SCMM-SPWM) strategy is proposed. In the negative half cycle of the modulation waves (MWs), dc offsets related to the amplitude of the carrier are set on the three MWs, respectively, to apply...... the same comparison logics of the MWs and carrier during positive and negative half cycles of the MWs. Thus, it is implemented with only one digital signal processor chip without any other attached logical circuit or controller. The reason for generating the zero-crossing voltage pulse disturbance (ZCVPD...

  2. 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.

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

    Energy Technology Data Exchange (ETDEWEB)

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

    2008-07-15

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

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

    Directory of Open Access Journals (Sweden)

    W. Bastiaan Kleijn

    2005-06-01

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

  5. Speech and audio processing for coding, enhancement and recognition

    CERN Document Server

    Togneri, Roberto; Narasimha, Madihally

    2015-01-01

    This book describes the basic principles underlying the generation, coding, transmission and enhancement of speech and audio signals, including advanced statistical and machine learning techniques for speech and speaker recognition with an overview of the key innovations in these areas. Key research undertaken in speech coding, speech enhancement, speech recognition, emotion recognition and speaker diarization are also presented, along with recent advances and new paradigms in these areas. ·         Offers readers a single-source reference on the significant applications of speech and audio processing to speech coding, speech enhancement and speech/speaker recognition. Enables readers involved in algorithm development and implementation issues for speech coding to understand the historical development and future challenges in speech coding research; ·         Discusses speech coding methods yielding bit-streams that are multi-rate and scalable for Voice-over-IP (VoIP) Networks; ·     �...

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

    Directory of Open Access Journals (Sweden)

    Niclas Schult

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

  7. Audio Editing Skills%音频剪辑技巧

    Institute of Scientific and Technical Information of China (English)

    范炜; 杨澍彬; 谭忠凯

    2015-01-01

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

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

    Science.gov (United States)

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

    2015-01-01

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

  9. The KUSC Classical Music Dataset for Audio Key Finding

    Directory of Open Access Journals (Sweden)

    Ching-Hua Chuan

    2014-08-01

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

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

    Directory of Open Access Journals (Sweden)

    Johahn eLeung

    2016-01-01

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

  11. Audio watermarking technologies for automatic cue sheet generation systems

    Science.gov (United States)

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

    2001-08-01

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

  12. Maintainable substrate carrier for electroplating

    Science.gov (United States)

    Chen, Chen-An [Milpitas, CA; Abas, Emmanuel Chua [Laguna, PH; Divino, Edmundo Anida [Cavite, PH; Ermita, Jake Randal G [Laguna, PH; Capulong, Jose Francisco S [Laguna, PH; Castillo, Arnold Villamor [Batangas, PH; Ma,; Xiaobing, Diana [Saratoga, CA

    2012-07-17

    One embodiment relates to a substrate carrier for use in electroplating a plurality of substrates. The carrier includes a non-conductive carrier body on which the substrates are placed and conductive lines embedded within the carrier body. A plurality of conductive clip attachment parts are attached in a permanent manner to the conductive lines embedded within the carrier body. A plurality of contact clips are attached in a removable manner to the clip attachment parts. The contact clips hold the substrates in place and conductively connecting the substrates with the conductive lines. Other embodiments, aspects and features are also disclosed.

  13. Maintainable substrate carrier for electroplating

    Energy Technology Data Exchange (ETDEWEB)

    Chen, Chen-An; Abas, Emmanuel Chua; Divino, Edmundo Anida; Ermita, Jake Randal G.; Capulong, Jose Francisco S.; Castillo, Arnold Villamor; Ma, Diana Xiaobing

    2016-08-02

    One embodiment relates to a substrate carrier for use in electroplating a plurality of substrates. The carrier includes a non-conductive carrier body on which the substrates are placed and conductive lines embedded within the carrier body. A plurality of conductive clip attachment parts are attached in a permanent manner to the conductive lines embedded within the carrier body. A plurality of contact clips are attached in a removable manner to the clip attachment parts. The contact clips hold the substrates in place and conductively connecting the substrates with the conductive lines. Other embodiments, aspects and features are also disclosed.

  14. 19 CFR 4.76 - Procedures and responsibilities of carriers filing outbound vessel manifest information via the AES.

    Science.gov (United States)

    2010-04-01

    ... outbound vessel manifest information via the AES. 4.76 Section 4.76 Customs Duties U.S. CUSTOMS AND BORDER... manifest information via the AES. (a) The sea carrier's module. The Sea Carrier's Module is a component of... information will be transmitted to Customs via AES for each shipment as far in advance of departure...

  15. 多载频相位编码雷达通信一体化研究%Research on Integrated Radar and Communication Based on Multi-carrier Phase Modulation Signal

    Institute of Scientific and Technical Information of China (English)

    胡朗; 薛广然; 唐尧; 杜自成

    2014-01-01

    为了最大程度地利用资源和减小电磁干扰,电子战平台的雷达通信一体化设计有着尤为重要的意义。针对通信中成熟应用且在雷达中有较好前景的正交多载频波形,提出直接序列扩频正交频分复用( OFDM)进行雷达通信一体化的想法。给出了实现方法,重点对该一体化信号模糊函数进行求解和分析,指出增加子载波数目信号分辨性能更佳;随机的信息流仅对模糊函数的邻道干扰项造成影响,为进一步优化一体化信号旁瓣性能提供了理论依据。最终的仿真结果和性能分析表明,该一体化信号能够实现雷达通信一体化的目的。%In order to make full utilization of resources and reduce electromagnetic interference,integration of radar and communication on the electronic warfare platform shows great significance. Since the multi-carrier signal has been widely used in communication and also shows good performances in radar detecting,an inte-grated radar and communication signal based on direct-sequence spread spectrum OFDM ( Orthogonal Fre-quency Division Multiplexing) is put forward. How to achieve the system is discussed afterwards. The ambi-guity function of the integrated signal is worked out to analyze the radar characteristics in detail. The conclu-sions that signal resolution performance grows with the increase of the sub-carrier number and random se-quence only influences adjacent channel interference of the ambiguity function define research directions to improve the signal sidelobe performance. Theoretical analysis and simulation results show that the integrated signal not only satisfies conventional radar detection but also shows good communication performance.

  16. Development of Community College Instructional Modules for Biology and Comparative Vertebrate Anatomy.

    Science.gov (United States)

    Vasiliauskas, Jura B.

    A project was undertaken to: (1) formulate objectives for a biology unit dealing with frog dissection and vertebrate anatomy, (2) on the basis of these objectives, develop self-instructional modules utilizing audio-visual and printed instructional materials, and (3) formulate instruments for the evaluation of the modules. The rationale for the…

  17. Glycosylation of solute carriers

    DEFF Research Database (Denmark)

    Pedersen, Nis Borbye; Carlsson, Michael C; Pedersen, Stine Helene Falsig

    2016-01-01

    as their posttranslational regulation, but only relatively little is known about the role of SLC glycosylation. Glycosylation is one of the most abundant posttranslational modifications of animal proteins and through recent advances in our understanding of protein-glycan interactions, the functional roles of SLC......Solute carriers (SLCs) are one of the largest groups of multi-spanning membrane proteins in mammals and include ubiquitously expressed proteins as well as proteins with highly restricted tissue expression. A vast number of studies have addressed the function and organization of SLCs as well...

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

    DEFF Research Database (Denmark)

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

    2012-01-01

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

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

    Science.gov (United States)

    Branje, Carmen J.; Fels, Deborah I.

    2012-01-01

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

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

    Science.gov (United States)

    Fawcett, Hannah; Oldfield, Jeremy

    2016-01-01

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