WorldWideScience

Sample records for time streaming audio

  1. Real Time Recognition Of Speakers From Internet Audio Stream

    Directory of Open Access Journals (Sweden)

    Weychan Radoslaw

    2015-09-01

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

  2. 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.

  3. Audio stream classification for multimedia database search

    Science.gov (United States)

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

    2013-03-01

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

  4. Streaming Audio and Video: New Challenges and Opportunities for Museums.

    Science.gov (United States)

    Spadaccini, Jim

    Streaming audio and video present new challenges and opportunities for museums. Streaming media is easier to author and deliver to Internet audiences than ever before; digital video editing is commonplace now that the tools--computers, digital video cameras, and hard drives--are so affordable; the cost of serving video files across the Internet…

  5. Parametric time-frequency domain spatial audio

    CERN Document Server

    Delikaris-Manias, Symeon; Politis, Archontis

    2018-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Warrick Roseboom

    2011-04-01

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

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

    Science.gov (United States)

    Egi, Noritsugu; Hayashi, Takanori; Takahashi, Akira

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

  8. 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.

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

    Science.gov (United States)

    Hashimoto, Masahiro; Kumashiro, Masaharu

    2004-06-01

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

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

    Science.gov (United States)

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

    2015-08-01

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

  11. A Bit Stream Scalable Speech/Audio Coder Combining Enhanced Regular Pulse Excitation and Parametric Coding

    Science.gov (United States)

    Riera-Palou, Felip; den Brinker, Albertus C.

    2007-12-01

    This paper introduces a new audio and speech broadband coding technique based on the combination of a pulse excitation coder and a standardized parametric coder, namely, MPEG-4 high-quality parametric coder. After presenting a series of enhancements to regular pulse excitation (RPE) to make it suitable for the modeling of broadband signals, it is shown how pulse and parametric codings complement each other and how they can be merged to yield a layered bit stream scalable coder able to operate at different points in the quality bit rate plane. The performance of the proposed coder is evaluated in a listening test. The major result is that the extra functionality of the bit stream scalability does not come at the price of a reduced performance since the coder is competitive with standardized coders (MP3, AAC, SSC).

  12. A Bit Stream Scalable Speech/Audio Coder Combining Enhanced Regular Pulse Excitation and Parametric Coding

    Directory of Open Access Journals (Sweden)

    Albertus C. den Brinker

    2007-01-01

    Full Text Available This paper introduces a new audio and speech broadband coding technique based on the combination of a pulse excitation coder and a standardized parametric coder, namely, MPEG-4 high-quality parametric coder. After presenting a series of enhancements to regular pulse excitation (RPE to make it suitable for the modeling of broadband signals, it is shown how pulse and parametric codings complement each other and how they can be merged to yield a layered bit stream scalable coder able to operate at different points in the quality bit rate plane. The performance of the proposed coder is evaluated in a listening test. The major result is that the extra functionality of the bit stream scalability does not come at the price of a reduced performance since the coder is competitive with standardized coders (MP3, AAC, SSC.

  13. A continuous-time/discrete-time mixed audio-band sigma delta ADC

    International Nuclear Information System (INIS)

    Liu Yan; Hua Siliang; Wang Donghui; Hou Chaohuan

    2011-01-01

    This paper introduces a mixed continuous-time/discrete-time, single-loop, fourth-order, 4-bit audio-band sigma delta ADC that combines the benefits of continuous-time and discrete-time circuits, while mitigating the challenges associated with continuous-time design. Measurement results show that the peak SNR of this ADC reaches 100 dB and the total power consumption is less than 30 mW. (semiconductor integrated circuits)

  14. Real-time Loudspeaker Distance Estimation with Stereo Audio

    DEFF Research Database (Denmark)

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

    2015-01-01

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

  15. Real-Time Transmission and Storage of Video, Audio, and Health Data in Emergency and Home Care Situations

    Directory of Open Access Journals (Sweden)

    Riccardo Stagnaro

    2007-01-01

    Full Text Available The increase in the availability of bandwidth for wireless links, network integration, and the computational power on fixed and mobile platforms at affordable costs allows nowadays for the handling of audio and video data, their quality making them suitable for medical application. These information streams can support both continuous monitoring and emergency situations. According to this scenario, the authors have developed and implemented the mobile communication system which is described in this paper. The system is based on ITU-T H.323 multimedia terminal recommendation, suitable for real-time data/video/audio and telemedical applications. The audio and video codecs, respectively, H.264 and G723.1, were implemented and optimized in order to obtain high performance on the system target processors. Offline media streaming storage and retrieval functionalities were supported by integrating a relational database in the hospital central system. The system is based on low-cost consumer technologies such as general packet radio service (GPRS and wireless local area network (WLAN or WiFi for lowband data/video transmission. Implementation and testing were carried out for medical emergency and telemedicine application. In this paper, the emergency case study is described.

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

    DEFF Research Database (Denmark)

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

    2017-01-01

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

  17. Using online handwriting and audio streams for mathematical expressions recognition: a bimodal approach

    Science.gov (United States)

    Medjkoune, Sofiane; Mouchère, Harold; Petitrenaud, Simon; Viard-Gaudin, Christian

    2013-01-01

    The work reported in this paper concerns the problem of mathematical expressions recognition. This task is known to be a very hard one. We propose to alleviate the difficulties by taking into account two complementary modalities. The modalities referred to are handwriting and audio ones. To combine the signals coming from both modalities, various fusion methods are explored. Performances evaluated on the HAMEX dataset show a significant improvement compared to a single modality (handwriting) based system.

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

    Science.gov (United States)

    Le, Long N; Jones, Douglas L

    2018-03-01

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

  19. Estudio del streaming de audio y vídeo sobre redes heterogéneas

    OpenAIRE

    Gómez Cruz, María del Carmen

    2009-01-01

    Las operadoras han encontrado nichos de negocio en la integración de múltiples servicios avanzados, como pueden ser Voz sobre IP, Vídeos bajo demanda, datos de alta capacidad, distribución de televisión de alta definición, etc. Esto supone una adaptación constante de sus redes de comunicación de banda ancha para soportar mayores anchos de banda, con mayor alcance, con menores pérdidas, en definitiva con mayores prestaciones; y promueve el desarrollo por parte de los fabricantes de audio y víd...

  20. Effect of Nicotine on Audio and Visual Reaction Time in Dipping ...

    African Journals Online (AJOL)

    Nicotine through blood is harmful and as there are fewer studies in India with respect to nicotines influence on reaction time especially in the smokeless tobacco users we studied this. Reaction time is a measure of the sensorimotor integration in a person. We used a PC 1000 Hz reaction timer to record the audio and visual ...

  1. A centralized audio presentation manager

    Energy Technology Data Exchange (ETDEWEB)

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

    1994-05-16

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

  2. WLAN Technologies for Audio Delivery

    Directory of Open Access Journals (Sweden)

    Nicolas-Alexander Tatlas

    2007-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Umapathy Karthikeyan

    2007-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Karthikeyan Umapathy

    2007-08-01

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

  5. Does listening to music with an audio ski helmet impair reaction time to peripheral stimuli?

    Science.gov (United States)

    Ruedl, G; Pocecco, E; Wolf, M; Schöpf, S; Burtscher, M; Kopp, M

    2012-12-01

    With the recent worldwide increase in ski helmet use, new market trends are developing, including audio helmets for listening to music while skiing or snowboarding. The aim of this study was to evaluate whether listening to music with an audio ski helmet impairs reaction time to peripheral stimuli. A within-subjects design study using the Compensatory-Tracking-Test was performed on 65 subjects (36 males and 29 females) who had a mean age of 23.3 ± 3.9 years. Using repeated measures analysis of variance, we found significant differences in reaction times between the 4 test conditions (p=0.039). The lowest mean reaction time (± SE) was measured for helmet use while listening to music (507.9 ± 13.2 ms), which was not different from helmet use alone (514.6 ± 12.5 ms) (p=0.528). However, compared to helmet use while listening to music, reaction time was significantly longer for helmet and ski goggles used together (535.8 ± 14.2 ms, p=0.005), with a similar trend for helmet and ski goggles used together while listening to music (526.9 ± 13.8 ms) (p=0.094). In conclusion, listening to music with an audio ski helmet did not increase mean reaction time to peripheral stimuli in a laboratory setting. © Georg Thieme Verlag KG Stuttgart · New York.

  6. Time-dependent 2-stream particle transport

    International Nuclear Information System (INIS)

    Corngold, Noel

    2015-01-01

    Highlights: • We consider time-dependent transport in the 2-stream or “rod” model via an attractive matrix formalism. • After reviewing some classical problems in homogeneous media we discuss transport in materials with whose density may vary. • There we achieve a significant contraction of the underlying Telegrapher’s equation. • We conclude with a discussion of stochastics, treated by the “first-order smoothing approximation.” - Abstract: We consider time-dependent transport in the 2-stream or “rod” model via an attractive matrix formalism. After reviewing some classical problems in homogeneous media we discuss transport in materials whose density may vary. There we achieve a significant contraction of the underlying Telegrapher’s equation. We conclude with a discussion of stochastics, treated by the “first-order smoothing approximation.”

  7. Robust and Reversible Audio Watermarking by Modifying Statistical Features in Time Domain

    Directory of Open Access Journals (Sweden)

    Shijun Xiang

    2017-01-01

    Full Text Available Robust and reversible watermarking is a potential technique in many sensitive applications, such as lossless audio or medical image systems. This paper presents a novel robust reversible audio watermarking method by modifying the statistic features in time domain in the way that the histogram of these statistical values is shifted for data hiding. Firstly, the original audio is divided into nonoverlapped equal-sized frames. In each frame, the use of three samples as a group generates a prediction error and a statistical feature value is calculated as the sum of all the prediction errors in the frame. The watermark bits are embedded into the frames by shifting the histogram of the statistical features. The watermark is reversible and robust to common signal processing operations. Experimental results have shown that the proposed method not only is reversible but also achieves satisfactory robustness to MP3 compression of 64 kbps and additive Gaussian noise of 35 dB.

  8. StreamWorks: the live and on-demand audio/video server and its applications in medical information systems

    Science.gov (United States)

    Akrout, Nabil M.; Gordon, Howard; Palisson, Patrice M.; Prost, Remy; Goutte, Robert

    1996-05-01

    Facing a world undergoing fundamental and rapid change, healthcare organizations are seeking ways to increase innovation, quality, productivity, and patient value, keys to more effective care. Individual clinics acting alone can respond in only a limited way, so re- engineering the process key which services are delivered demands real-time collaborative technology that provides immediate information sharing, improving the management and coordination of information in cross-functional teams. StreamWorks is a development stage architecture that uses a distribution technique to deliver an advanced information management system for telemedicine. The challenge of StreamWorks in telemedicine is to enable equity of the quality of Health Care of Telecommunications and Information Technology also to patients in less favored regions, like India or China, where the quality of medical care varies greatly by region, but where there are some very current communications facilities.

  9. A Statistical Method to Predict Flow Permanence in Dryland Streams from Time Series of Stream Temperature

    Directory of Open Access Journals (Sweden)

    Ivan Arismendi

    2017-12-01

    Full Text Available Intermittent and ephemeral streams represent more than half of the length of the global river network. Dryland freshwater ecosystems are especially vulnerable to changes in human-related water uses as well as shifts in terrestrial climates. Yet, the description and quantification of patterns of flow permanence in these systems is challenging mostly due to difficulties in instrumentation. Here, we took advantage of existing stream temperature datasets in dryland streams in the northwest Great Basin desert, USA, to extract critical information on climate-sensitive patterns of flow permanence. We used a signal detection technique, Hidden Markov Models (HMMs, to extract information from daily time series of stream temperature to diagnose patterns of stream drying. Specifically, we applied HMMs to time series of daily standard deviation (SD of stream temperature (i.e., dry stream channels typically display highly variable daily temperature records compared to wet stream channels between April and August (2015–2016. We used information from paired stream and air temperature data loggers as well as co-located stream temperature data loggers with electrical resistors as confirmatory sources of the timing of stream drying. We expanded our approach to an entire stream network to illustrate the utility of the method to detect patterns of flow permanence over a broader spatial extent. We successfully identified and separated signals characteristic of wet and dry stream conditions and their shifts over time. Most of our study sites within the entire stream network exhibited a single state over the entire season (80%, but a portion of them showed one or more shifts among states (17%. We provide recommendations to use this approach based on a series of simple steps. Our findings illustrate a successful method that can be used to rigorously quantify flow permanence regimes in streams using existing records of stream temperature.

  10. A statistical method to predict flow permanence in dryland streams from time series of stream temperature

    Science.gov (United States)

    Arismendi, Ivan; Dunham, Jason B.; Heck, Michael; Schultz, Luke; Hockman-Wert, David

    2017-01-01

    Intermittent and ephemeral streams represent more than half of the length of the global river network. Dryland freshwater ecosystems are especially vulnerable to changes in human-related water uses as well as shifts in terrestrial climates. Yet, the description and quantification of patterns of flow permanence in these systems is challenging mostly due to difficulties in instrumentation. Here, we took advantage of existing stream temperature datasets in dryland streams in the northwest Great Basin desert, USA, to extract critical information on climate-sensitive patterns of flow permanence. We used a signal detection technique, Hidden Markov Models (HMMs), to extract information from daily time series of stream temperature to diagnose patterns of stream drying. Specifically, we applied HMMs to time series of daily standard deviation (SD) of stream temperature (i.e., dry stream channels typically display highly variable daily temperature records compared to wet stream channels) between April and August (2015–2016). We used information from paired stream and air temperature data loggers as well as co-located stream temperature data loggers with electrical resistors as confirmatory sources of the timing of stream drying. We expanded our approach to an entire stream network to illustrate the utility of the method to detect patterns of flow permanence over a broader spatial extent. We successfully identified and separated signals characteristic of wet and dry stream conditions and their shifts over time. Most of our study sites within the entire stream network exhibited a single state over the entire season (80%), but a portion of them showed one or more shifts among states (17%). We provide recommendations to use this approach based on a series of simple steps. Our findings illustrate a successful method that can be used to rigorously quantify flow permanence regimes in streams using existing records of stream temperature.

  11. 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...

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

    Science.gov (United States)

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

    2015-11-01

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

  13. Time-Based Data Streams: Fundamental Concepts for a Data Resource for Streams

    Energy Technology Data Exchange (ETDEWEB)

    Beth A. Plale

    2009-10-10

    Real time data, which we call data streams, are readings from instruments, environmental, bodily or building sensors that are generated at regular intervals and often, due to their volume, need to be processed in real time. Often a single pass is all that can be made on the data, and a decision to discard or keep the instance is made on the spot. Too, the stream is for all practical purposes indefinite, so decisions must be made on incomplete knowledge. This notion of data streams has a different set of issues from a file, for instance, that is byte streamed to a reader. The file is finite, so the byte stream is becomes a processing convenience more than a fundamentally different kind of data. Through the duration of the project we examined three aspects of streaming data: the first, techniques to handle streaming data in a distributed system organized as a collection of web services, the second, the notion of the dashboard and real time controllable analysis constructs in the context of the Fermi Tevatron Beam Position Monitor, and third and finally, we examined provenance collection of stream processing such as might occur as raw observational data flows from the source and undergoes correction, cleaning, and quality control. The impact of this work is severalfold. We were one of the first to advocate that streams had little value unless aggregated, and that notion is now gaining general acceptance. We were one of the first groups to grapple with the notion of provenance of stream data also.

  14. 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.

  15. The evolution of streams in a time-dependent potential

    NARCIS (Netherlands)

    Buist, Hans J. T.; Helmi, Amina

    2015-01-01

    We study the evolution of streams in a time-dependent spherical gravitational potential. Our goal is to establish what are the imprints of this time evolution on the properties of streams as well as their observability. To this end, we have performed a suite of test-particle experiments for a host

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

    Science.gov (United States)

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

    2014-02-01

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

  17. Implementation and Analysis of Real-Time Streaming Protocols.

    Science.gov (United States)

    Santos-González, Iván; Rivero-García, Alexandra; Molina-Gil, Jezabel; Caballero-Gil, Pino

    2017-04-12

    Communication media have become the primary way of interaction thanks to the discovery and innovation of many new technologies. One of the most widely used communication systems today is video streaming, which is constantly evolving. Such communications are a good alternative to face-to-face meetings, and are therefore very useful for coping with many problems caused by distance. However, they suffer from different issues such as bandwidth limitation, network congestion, energy efficiency, cost, reliability and connectivity. Hence, the quality of service and the quality of experience are considered the two most important issues for this type of communication. This work presents a complete comparative study of two of the most used protocols of video streaming, Real Time Streaming Protocol (RTSP) and the Web Real-Time Communication (WebRTC). In addition, this paper proposes two new mobile applications that implement those protocols in Android whose objective is to know how they are influenced by the aspects that most affect the streaming quality of service, which are the connection establishment time and the stream reception time. The new video streaming applications are also compared with the most popular video streaming applications for Android, and the experimental results of the analysis show that the developed WebRTC implementation improves the performance of the most popular video streaming applications with respect to the stream packet delay.

  18. Real-time decreased sensitivity to an audio-visual illusion during goal-directed reaching.

    Directory of Open Access Journals (Sweden)

    Luc Tremblay

    Full Text Available In humans, sensory afferences are combined and integrated by the central nervous system (Ernst MO, Bülthoff HH (2004 Trends Cogn. Sci. 8: 162-169 and appear to provide a holistic representation of the environment. Empirical studies have repeatedly shown that vision dominates the other senses, especially for tasks with spatial demands. In contrast, it has also been observed that sound can strongly alter the perception of visual events. For example, when presented with 2 flashes and 1 beep in a very brief period of time, humans often report seeing 1 flash (i.e. fusion illusion, Andersen TS, Tiippana K, Sams M (2004 Brain Res. Cogn. Brain Res. 21: 301-308. However, it is not known how an unfolding movement modulates the contribution of vision to perception. Here, we used the audio-visual illusion to demonstrate that goal-directed movements can alter visual information processing in real-time. Specifically, the fusion illusion was linearly reduced as a function of limb velocity. These results suggest that cue combination and integration can be modulated in real-time by goal-directed behaviors; perhaps through sensory gating (Chapman CE, Beauchamp E (2006 J. Neurophysiol. 96: 1664-1675 and/or altered sensory noise (Ernst MO, Bülthoff HH (2004 Trends Cogn. Sci. 8: 162-169 during limb movements.

  19. Simulator study of the effect of visual-motion time delays on pilot tracking performance with an audio side task

    Science.gov (United States)

    Riley, D. R.; Miller, G. K., Jr.

    1978-01-01

    The effect of time delay was determined in the visual and motion cues in a flight simulator on pilot performance in tracking a target aircraft that was oscillating sinusoidally in altitude only. An audio side task was used to assure the subject was fully occupied at all times. The results indicate that, within the test grid employed, about the same acceptable time delay (250 msec) was obtained for a single aircraft (fighter type) by each of two subjects for both fixed-base and motion-base conditions. Acceptable time delay is defined as the largest amount of delay that can be inserted simultaneously into the visual and motion cues before performance degradation occurs. A statistical analysis of the data was made to establish this value of time delay. Audio side task provided quantitative data that documented the subject's work level.

  20. Advanced real-time manipulation of video streams

    CERN Document Server

    Herling, Jan

    2014-01-01

    Diminished Reality is a new fascinating technology that removes real-world content from live video streams. This sensational live video manipulation actually removes real objects and generates a coherent video stream in real-time. Viewers cannot detect modified content. Existing approaches are restricted to moving objects and static or almost static cameras and do not allow real-time manipulation of video content. Jan Herling presents a new and innovative approach for real-time object removal with arbitrary camera movements.

  1. Specification and Compilation of Real-Time Stream Processing Applications

    NARCIS (Netherlands)

    Geuns, S.J.

    2015-01-01

    This thesis is concerned with the specification, compilation and corresponding temporal analysis of real-time stream processing applications that are executed on embedded multiprocessor systems. An example of such applications are software defined radio applications. These applications typically

  2. A real time sorting algorithm to time sort any deterministic time disordered data stream

    Science.gov (United States)

    Saini, J.; Mandal, S.; Chakrabarti, A.; Chattopadhyay, S.

    2017-12-01

    In new generation high intensity high energy physics experiments, millions of free streaming high rate data sources are to be readout. Free streaming data with associated time-stamp can only be controlled by thresholds as there is no trigger information available for the readout. Therefore, these readouts are prone to collect large amount of noise and unwanted data. For this reason, these experiments can have output data rate of several orders of magnitude higher than the useful signal data rate. It is therefore necessary to perform online processing of the data to extract useful information from the full data set. Without trigger information, pre-processing on the free streaming data can only be done with time based correlation among the data set. Multiple data sources have different path delays and bandwidth utilizations and therefore the unsorted merged data requires significant computational efforts for real time manifestation of sorting before analysis. Present work reports a new high speed scalable data stream sorting algorithm with its architectural design, verified through Field programmable Gate Array (FPGA) based hardware simulation. Realistic time based simulated data likely to be collected in an high energy physics experiment have been used to study the performance of the algorithm. The proposed algorithm uses parallel read-write blocks with added memory management and zero suppression features to make it efficient for high rate data-streams. This algorithm is best suited for online data streams with deterministic time disorder/unsorting on FPGA like hardware.

  3. 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....

  4. Interpretation of stream programs: characterizing type 2 polynomial time complexity

    OpenAIRE

    Férée , Hugo; Hainry , Emmanuel; Hoyrup , Mathieu; Péchoux , Romain

    2010-01-01

    International audience; We study polynomial time complexity of type 2 functionals. For that purpose, we introduce a first order functional stream language. We give criteria, named well-founded, on such programs relying on second order interpretation that characterize two variants of type 2 polynomial complexity including the Basic Feasible Functions (BFF). These charac- terizations provide a new insight on the complexity of stream programs. Finally, we adapt these results to functions over th...

  5. Interactive real-time media streaming with reliable communication

    Science.gov (United States)

    Pan, Xunyu; Free, Kevin M.

    2014-02-01

    Streaming media is a recent technique for delivering multimedia information from a source provider to an end- user over the Internet. The major advantage of this technique is that the media player can start playing a multimedia file even before the entire file is transmitted. Most streaming media applications are currently implemented based on the client-server architecture, where a server system hosts the media file and a client system connects to this server system to download the file. Although the client-server architecture is successful in many situations, it may not be ideal to rely on such a system to provide the streaming service as users may be required to register an account using personal information in order to use the service. This is troublesome if a user wishes to watch a movie simultaneously while interacting with a friend in another part of the world over the Internet. In this paper, we describe a new real-time media streaming application implemented on a peer-to-peer (P2P) architecture in order to overcome these challenges within a mobile environment. When using the peer-to-peer architecture, streaming media is shared directly between end-users, called peers, with minimal or no reliance on a dedicated server. Based on the proposed software pɛvμa (pronounced [revma]), named for the Greek word meaning stream, we can host a media file on any computer and directly stream it to a connected partner. To accomplish this, pɛvμa utilizes the Microsoft .NET Framework and Windows Presentation Framework, which are widely available on various types of windows-compatible personal computers and mobile devices. With specially designed multi-threaded algorithms, the application can stream HD video at speeds upwards of 20 Mbps using the User Datagram Protocol (UDP). Streaming and playback are handled using synchronized threads that communicate with one another once a connection is established. Alteration of playback, such as pausing playback or tracking to a

  6. Sequential specification of time-aware stream processing applications

    NARCIS (Netherlands)

    Geuns, S.J.; Hausmans, J.P.H.M.; Bekooij, Marco Jan Gerrit

    Automatic parallelization of Nested Loop Programs (NLPs) is an attractive method to create embedded real-time stream processing applications for multi-core systems. However, the description and parallelization of applications with a time dependent functional behavior has not been considered in NLPs.

  7. A Method to Detect AAC Audio Forgery

    Directory of Open Access Journals (Sweden)

    Qingzhong Liu

    2015-08-01

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

  8. Cache Timing Analysis of eStream Finalists

    DEFF Research Database (Denmark)

    Zenner, Erik

    2009-01-01

    Cache Timing Attacks have attracted a lot of cryptographic attention due to their relevance for the AES. However, their applicability to other cryptographic primitives is less well researched. In this talk, we give an overview over our analysis of the stream ciphers that were selected for phase 3...

  9. STREAM

    DEFF Research Database (Denmark)

    Godsk, Mikkel

    This paper presents a flexible model, ‘STREAM’, for transforming higher science education into blended and online learning. The model is inspired by ideas of active and collaborative learning and builds on feedback strategies well-known from Just-in-Time Teaching, Flipped Classroom, and Peer...... Instruction. The aim of the model is to provide both a concrete and comprehensible design toolkit for adopting and implementing educational technologies in higher science teaching practice and at the same time comply with diverse ambitions. As opposed to the above-mentioned feedback strategies, the STREAM...... model supports a relatively diverse use of educational technologies and may also be used to transform teaching into completely online learning. So far both teachers and educational developers have positively received the model and the initial design experiences show promise....

  10. Securing Digital Audio using Complex Quadratic Map

    Science.gov (United States)

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

    2018-03-01

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

  11. Real-time change detection in data streams with FPGAs

    International Nuclear Information System (INIS)

    Vega, J.; Dormido-Canto, S.; Cruz, T.; Ruiz, M.; Barrera, E.; Castro, R.; Murari, A.; Ochando, M.

    2014-01-01

    Highlights: • Automatic recognition of changes in data streams of multidimensional signals. • Detection algorithm based on testing exchangeability on-line. • Real-time and off-line applicability. • Real-time implementation in FPGAs. - Abstract: The automatic recognition of changes in data streams is useful in both real-time and off-line data analyses. This article shows several effective change-detecting algorithms (based on martingales) and describes their real-time applicability in the data acquisition systems through the use of Field Programmable Gate Arrays (FPGA). The automatic event recognition system is absolutely general and it does not depend on either the particular event to detect or the specific data representation (waveforms, images or multidimensional signals). The developed approach provides good results for change detection in both the temporal evolution of profiles and the two-dimensional spatial distribution of volume emission intensity. The average computation time in the FPGA is 210 μs per profile

  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. Balancing Audio

    DEFF Research Database (Denmark)

    Walther-Hansen, Mads

    2016-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Maxime eAmbard

    2015-10-01

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

  15. Portable Audio Design

    DEFF Research Database (Denmark)

    Groth, Sanne Krogh

    2014-01-01

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

  16. A Real-Time Semiautonomous Audio Panning System for Music Mixing

    Directory of Open Access Journals (Sweden)

    Perez_Gonzalez Enrique

    2010-01-01

    Full Text Available A real-time semiautonomous stereo panning system for music mixing has been implemented. The system uses spectral decomposition, constraint rules, and cross-adaptive algorithms to perform real-time placement of sources in a stereo mix. A subjective evaluation test was devised to evaluate its quality against human panning. It was shown that the automatic panning technique performed better than a nonexpert and showed no significant statistical difference to the performance of a professional mixing engineer.

  17. Abstractions for aperiodic multiprocessor scheduling of real-time stream processing applications

    NARCIS (Netherlands)

    Hausmans, J.P.H.M.

    2015-01-01

    Embedded multiprocessor systems are often used in the domain of real-time stream processing applications to keep up with increasing power and performance requirements. Examples of such real-time stream processing applications are digital radio baseband processing and WLAN transceivers. These stream

  18. Audio Restoration

    Science.gov (United States)

    Esquef, Paulo A. A.

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

  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. Real-time WAMI streaming target tracking in fog

    Science.gov (United States)

    Chen, Yu; Blasch, Erik; Chen, Ning; Deng, Anna; Ling, Haibin; Chen, Genshe

    2016-05-01

    Real-time information fusion based on WAMI (Wide-Area Motion Imagery), FMV (Full Motion Video), and Text data is highly desired for many mission critical emergency or security applications. Cloud Computing has been considered promising to achieve big data integration from multi-modal sources. In many mission critical tasks, however, powerful Cloud technology cannot satisfy the tight latency tolerance as the servers are allocated far from the sensing platform, actually there is no guaranteed connection in the emergency situations. Therefore, data processing, information fusion, and decision making are required to be executed on-site (i.e., near the data collection). Fog Computing, a recently proposed extension and complement for Cloud Computing, enables computing on-site without outsourcing jobs to a remote Cloud. In this work, we have investigated the feasibility of processing streaming WAMI in the Fog for real-time, online, uninterrupted target tracking. Using a single target tracking algorithm, we studied the performance of a Fog Computing prototype. The experimental results are very encouraging that validated the effectiveness of our Fog approach to achieve real-time frame rates.

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

    DEFF Research Database (Denmark)

    Dimitrov, Smilen; Serafin, Stefania

    2014-01-01

    This article investigates how lacking suitable platforms for laboratory exercises becomes a learning problem, limiting the practical experience students gain. In engineering education, laboratory demonstration difficulty of issues like real-time streaming in digital signal and audio processing...... afforded by such laboratories, and their open nature, could testably improve the diversity of demonstrated practical topics, while maintaining engineering students' motivation....

  2. Huffman coding in advanced audio coding standard

    Science.gov (United States)

    Brzuchalski, Grzegorz

    2012-05-01

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

  3. Simulation of acoustic streaming by means of the finite-difference time-domain method

    DEFF Research Database (Denmark)

    Santillan, Arturo Orozco

    2012-01-01

    Numerical simulations of acoustic streaming generated by a standing wave in a narrow twodimensional cavity are presented. In this case, acoustic streaming arises from the viscous boundary layers set up at the surfaces of the walls. It is known that streaming vortices inside the boundary layer have...... directions of rotation that are opposite to those of the outer streaming vortices (Rayleigh streaming). The general objective of the work described in this paper has been to study the extent to which it is possible to simulate both the outer streaming vortices and the inner boundary layer vortices using...... the finite-difference time-domain method. To simplify the problem, thermal effects are not considered. The motivation of the described investigation has been the possibility of using the numerical method to study acoustic streaming, particularly under non-steady conditions. Results are discussed for channels...

  4. Analysis of sound data streamed over the network

    Directory of Open Access Journals (Sweden)

    Jiří Fejfar

    2013-01-01

    Full Text Available In this paper we inspect a difference between original sound recording and signal captured after streaming this original recording over a network loaded with a heavy traffic. There are several kinds of failures occurring in the captured recording caused by network congestion. We try to find a method how to evaluate correctness of streamed audio. Usually there are metrics based on a human perception of a signal such as “signal is clear, without audible failures”, “signal is having some failures but it is understandable”, or “signal is inarticulate”. These approaches need to be statistically evaluated on a broad set of respondents, which is time and resource consuming. We try to propose some metrics based on signal properties allowing us to compare the original and captured recording. We use algorithm called Dynamic Time Warping (Müller, 2007 commonly used for time series comparison in this paper. Some other time series exploration approaches can be found in (Fejfar, 2011 and (Fejfar, 2012. The data was acquired in our network laboratory simulating network traffic by downloading files, streaming audio and video simultaneously. Our former experiment inspected Quality of Service (QoS and its impact on failures of received audio data stream. This experiment is focused on the comparison of sound recordings rather than network mechanism.We focus, in this paper, on a real time audio stream such as a telephone call, where it is not possible to stream audio in advance to a “pool”. Instead it is necessary to achieve as small delay as possible (between speaker voice recording and listener voice replay. We are using RTP protocol for streaming audio.

  5. 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 ...

  6. PRESEE: an MDL/MML algorithm to time-series stream segmenting.

    Science.gov (United States)

    Xu, Kaikuo; Jiang, Yexi; Tang, Mingjie; Yuan, Changan; Tang, Changjie

    2013-01-01

    Time-series stream is one of the most common data types in data mining field. It is prevalent in fields such as stock market, ecology, and medical care. Segmentation is a key step to accelerate the processing speed of time-series stream mining. Previous algorithms for segmenting mainly focused on the issue of ameliorating precision instead of paying much attention to the efficiency. Moreover, the performance of these algorithms depends heavily on parameters, which are hard for the users to set. In this paper, we propose PRESEE (parameter-free, real-time, and scalable time-series stream segmenting algorithm), which greatly improves the efficiency of time-series stream segmenting. PRESEE is based on both MDL (minimum description length) and MML (minimum message length) methods, which could segment the data automatically. To evaluate the performance of PRESEE, we conduct several experiments on time-series streams of different types and compare it with the state-of-art algorithm. The empirical results show that PRESEE is very efficient for real-time stream datasets by improving segmenting speed nearly ten times. The novelty of this algorithm is further demonstrated by the application of PRESEE in segmenting real-time stream datasets from ChinaFLUX sensor networks data stream.

  7. Gas stream analysis using voltage-current time differential operation of electrochemical sensors

    Science.gov (United States)

    Woo, Leta Yar-Li; Glass, Robert Scott; Fitzpatrick, Joseph Jay; Wang, Gangqiang; Henderson, Brett Tamatea; Lourdhusamy, Anthoniraj; Steppan, James John; Allmendinger, Klaus Karl

    2018-01-02

    A method for analysis of a gas stream. The method includes identifying an affected region of an affected waveform signal corresponding to at least one characteristic of the gas stream. The method also includes calculating a voltage-current time differential between the affected region of the affected waveform signal and a corresponding region of an original waveform signal. The affected region and the corresponding region of the waveform signals have a sensitivity specific to the at least one characteristic of the gas stream. The method also includes generating a value for the at least one characteristic of the gas stream based on the calculated voltage-current time differential.

  8. Smartphone audio port data collection cookbook

    Directory of Open Access Journals (Sweden)

    Kyle Forinash

    2018-06-01

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

  9. Using Streaming Analytics for Effective Real Time Network Visibility -

    Science.gov (United States)

    on in your network right now. Certainly the other thing that we talked about on the big data side was [inaudible] data. So now we'll drill into - so this is all the traffic from the internal network to the taking a streaming analytics approach to network traffic analysis. So we can go to the next - there we go

  10. Towards a characterization of real-time streaming systems

    NARCIS (Netherlands)

    Weffers-Albu, M.A.; Lukkien, J.J.; Stok, van der P.D.V.; Puaut, I.

    2005-01-01

    In this article we provide a model for the dynamic behavior of a single video streaming chain, by formulating a theorem describing the stable behavior. This stable behavior is characterized in terms of the elementary actions of the components in the chain, from which standard performance measures

  11. Open Source Initiative Powers Real-Time Data Streams

    Science.gov (United States)

    2014-01-01

    Under an SBIR contract with Dryden Flight Research Center, Creare Inc. developed a data collection tool called the Ring Buffered Network Bus. The technology has now been released under an open source license and is hosted by the Open Source DataTurbine Initiative. DataTurbine allows anyone to stream live data from sensors, labs, cameras, ocean buoys, cell phones, and more.

  12. Real-time lossless compression of depth streams

    KAUST Repository

    Schneider, Jens

    2017-08-17

    Various examples are provided for lossless compression of data streams. In one example, a Z-lossless (ZLS) compression method includes generating compacted depth information by condensing information of a depth image and a compressed binary representation of the depth image using histogram compaction and decorrelating the compacted depth information to produce bitplane slicing of residuals by spatial prediction. In another example, an apparatus includes imaging circuitry that can capture one or more depth images and processing circuitry that can generate compacted depth information by condensing information of a captured depth image and a compressed binary representation of the captured depth image using histogram compaction; decorrelate the compacted depth information to produce bitplane slicing of residuals by spatial prediction; and generate an output stream based upon the bitplane slicing.

  13. Real-time lossless compression of depth streams

    KAUST Repository

    Schneider, Jens

    2017-01-01

    Various examples are provided for lossless compression of data streams. In one example, a Z-lossless (ZLS) compression method includes generating compacted depth information by condensing information of a depth image and a compressed binary representation of the depth image using histogram compaction and decorrelating the compacted depth information to produce bitplane slicing of residuals by spatial prediction. In another example, an apparatus includes imaging circuitry that can capture one or more depth images and processing circuitry that can generate compacted depth information by condensing information of a captured depth image and a compressed binary representation of the captured depth image using histogram compaction; decorrelate the compacted depth information to produce bitplane slicing of residuals by spatial prediction; and generate an output stream based upon the bitplane slicing.

  14. Overcoming equifinality: Leveraging long time series for stream metabolism estimation

    Science.gov (United States)

    Appling, Alison; Hall, Robert O.; Yackulic, Charles B.; Arroita, Maite

    2018-01-01

    The foundational ecosystem processes of gross primary production (GPP) and ecosystem respiration (ER) cannot be measured directly but can be modeled in aquatic ecosystems from subdaily patterns of oxygen (O2) concentrations. Because rivers and streams constantly exchange O2 with the atmosphere, models must either use empirical estimates of the gas exchange rate coefficient (K600) or solve for all three parameters (GPP, ER, and K600) simultaneously. Empirical measurements of K600 require substantial field work and can still be inaccurate. Three-parameter models have suffered from equifinality, where good fits to O2 data are achieved by many different parameter values, some unrealistic. We developed a new three-parameter, multiday model that ensures similar values for K600 among days with similar physical conditions (e.g., discharge). Our new model overcomes the equifinality problem by (1) flexibly relating K600 to discharge while permitting moderate daily deviations and (2) avoiding the oft-violated assumption that residuals in O2 predictions are uncorrelated. We implemented this hierarchical state-space model and several competitor models in an open-source R package, streamMetabolizer. We then tested the models against both simulated and field data. Our new model reduces error by as much as 70% in daily estimates of K600, GPP, and ER. Further, accuracy benefits of multiday data sets require as few as 3 days of data. This approach facilitates more accurate metabolism estimates for more streams and days, enabling researchers to better quantify carbon fluxes, compare streams by their metabolic regimes, and investigate controls on aquatic activity.

  15. Audio Conferencing Enhancements

    OpenAIRE

    VESTERINEN, LEENA

    2006-01-01

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

  16. Scalable Video Streaming Adaptive to Time-Varying IEEE 802.11 MAC Parameters

    Science.gov (United States)

    Lee, Kyung-Jun; Suh, Doug-Young; Park, Gwang-Hoon; Huh, Jae-Doo

    This letter proposes a QoS control method for video streaming service over wireless networks. Based on statistical analysis, the time-varying MAC parameters highly related to channel condition are selected to predict available bitrate. Adaptive bitrate control of scalably-encoded video guarantees continuity in streaming service even if the channel condition changes abruptly.

  17. 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

  18. Tune in the Net with RealAudio.

    Science.gov (United States)

    Buchanan, Larry

    1997-01-01

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

  19. Semantic Context Detection Using Audio Event Fusion

    Directory of Open Access Journals (Sweden)

    Cheng Wen-Huang

    2006-01-01

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

  20. Time course of auditory streaming: Do CI users differ from normal-hearing listeners?

    Directory of Open Access Journals (Sweden)

    Martin eBöckmann-Barthel

    2014-07-01

    Full Text Available In a complex acoustical environment with multiple sound sources the auditory system uses streaming as a tool to organize the incoming sounds in one or more streams depending on the stimulus parameters. Streaming is commonly studied by alternating sequences of signals. These are often tones with different frequencies. The present study investigates stream segregation in cochlear implant (CI users, where hearing is restored by electrical stimulation of the auditory nerve. CI users listened to 30-s long sequences of alternating A and B harmonic complexes at four different fundamental frequency separations, ranging from 2 to 14 semitones. They had to indicate as promptly as possible after sequence onset, if they perceived one stream or two streams and, in addition, any changes of the percept throughout the rest of the sequence. The conventional view is that the initial percept is always that of a single stream which may after some time change to a percept of two streams. This general build-up hypothesis has recently been challenged on the basis of a new analysis of data of normal-hearing listeners which showed a build-up response only for an intermediate frequency separation. Using the same experimental paradigm and analysis, the present study found that the results of CI users agree with those of the normal-hearing listeners: (i the probability of the first decision to be a one-stream percept decreased and that of a two-stream percept increased as Δf increased, and (ii a build-up was only found for 6 semitones. Only the time elapsed before the listeners made their first decision of the percept was prolonged as compared to normal-hearing listeners. The similarity in the data of the CI user and the normal-hearing listeners indicates that the quality of stream formation is similar in these groups of listeners.

  1. Radioactive Decay: Audio Data Collection

    Science.gov (United States)

    Struthers, Allan

    2009-01-01

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

  2. A Joint Audio-Visual Approach to Audio Localization

    DEFF Research Database (Denmark)

    Jensen, Jesper Rindom; Christensen, Mads Græsbøll

    2015-01-01

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

  3. Publicación de materiales audiovisuales a través de un servidor de video-streaming Publication of audio-visual materials through a streaming video server

    OpenAIRE

    Acevedo Clavijo Edwin Jovanny; Parra Toloza Dina Julieth; Winlker Hernadez Walter

    2010-01-01

    Esta propuesta tiene como objetivo estudiar varias alternativas de servidores Streaming para determinar la mejor herramienta para el desarrollo de la publicación de material audiovisual educativo. Se evaluaron las plataformas más utilizadas teniendo en cuenta sus características y beneficios que tiene cada servidor entre las los cuales están: Hélix Universal Server, Windows Media Server de Microsoft, Peer Cast y Darwin Server. implementando un servidor con mayores capacidades y beneficios par...

  4. Publicación de materiales audiovisuales a través de un servidor de video-streaming Publication of audio-visual materials through a streaming video server

    Directory of Open Access Journals (Sweden)

    Acevedo Clavijo Edwin Jovanny

    2010-07-01

    Full Text Available Esta propuesta tiene como objetivo estudiar varias alternativas de servidores Streaming para determinar la mejor herramienta para el desarrollo de la publicación de material audiovisual educativo. Se evaluaron las plataformas más utilizadas teniendo en cuenta sus características y beneficios que tiene cada servidor entre las los cuales están: Hélix Universal Server, Windows Media Server de Microsoft, Peer Cast y Darwin Server. implementando un servidor con mayores capacidades y beneficios para la publicación de videos con fines académicos a través de la intranet de la Universidad Cooperativa de Colombia seccional Barrancabermeja This proposal has as an principal objective to study different alternatives for streaming servers to determine the best tool in the project’s development. Platforms most used were evaluated features and benefits in each served such as: Helix Universal Server, Microsoft Windows Media Server, Peer Cast and Darwin Server. Implementing a server with more capabilities and benefits for the publication of videos for academic purposes through the intranet of the Cooperative University of Colombia Barrancabermeja’s sectional

  5. Audio Recording of Children with Dyslalia

    Directory of Open Access Journals (Sweden)

    Stefan Gheorghe Pentiuc

    2008-01-01

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

  6. Audio Recording of Children with Dyslalia

    OpenAIRE

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

    2008-01-01

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

  7. Categorizing Video Game Audio

    DEFF Research Database (Denmark)

    Westerberg, Andreas Rytter; Schoenau-Fog, Henrik

    2015-01-01

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

  8. Adjusting patients streaming initiated by a wait time threshold in emergency department for minimizing opportunity cost.

    Science.gov (United States)

    Kim, Byungjoon B J; Delbridge, Theodore R; Kendrick, Dawn B

    2017-07-10

    Purpose Two different systems for streaming patients were considered to improve efficiency measures such as waiting times (WTs) and length of stay (LOS) for a current emergency department (ED). A typical fast track area (FTA) and a fast track with a wait time threshold (FTW) were designed and compared effectiveness measures from the perspective of total opportunity cost of all patients' WTs in the ED. The paper aims to discuss these issues. Design/methodology/approach This retrospective case study used computerized ED patient arrival to discharge time logs (between July 1, 2009 and June 30, 2010) to build computer simulation models for the FTA and fast track with wait time threshold systems. Various wait time thresholds were applied to stream different acuity-level patients. National average wait time for each acuity level was considered as a threshold to stream patients. Findings The fast track with a wait time threshold (FTW) showed a statistically significant shorter total wait time than the current system or a typical FTA system. The patient streaming management would improve the service quality of the ED as well as patients' opportunity costs by reducing the total LOS in the ED. Research limitations/implications The results of this study were based on computer simulation models with some assumptions such as no transfer times between processes, an arrival distribution of patients, and no deviation of flow pattern. Practical implications When the streaming of patient flow can be managed based on the wait time before being seen by a physician, it is possible for patients to see a physician within a tolerable wait time, which would result in less crowded in the ED. Originality/value A new streaming scheme of patients' flow may improve the performance of fast track system.

  9. Mining Frequent Item Sets in Asynchronous Transactional Data Streams over Time Sensitive Sliding Windows Model

    International Nuclear Information System (INIS)

    Javaid, Q.; Memon, F.; Talpur, S.; Arif, M.; Awan, M.D.

    2016-01-01

    EPs (Extracting Frequent Patterns) from the continuous transactional data streams is a challenging and critical task in some of the applications, such as web mining, data analysis and retail market, prediction and network monitoring, or analysis of stock market exchange data. Many algorithms have been developed previously for mining FPs (Frequent Patterns) from a data stream. Such algorithms are currently highly required to develop new solutions and approaches to the precise handling of data streams. New techniques, solutions, or approaches are developed to address unbounded, ordered, and continuous sequences of data and for the generation of data at a rapid speed from data streams. Hence, extracting FPs using fresh or recent data involves the high-level analysis of data streams. We have suggested an efficient technique for the window sliding model; this technique extracts new and fresh FPs from high-speed data streams. In this study, a CPILT (Compacted Tree Compact Pattern Tree) is developed to capture the latest contents in the stream and to efficiently remove outdated contents from the data stream. The main concept introduced in this work on CPILT is the dynamic restructuring of a tree, which is helpful in producing a compacted tree and the frequency descending structure of a tree on runtime. With the help of the mining technique of FP growth, a complete list of new and fresh FPs is obtained from a CPILT using an existing window. The memory usage and time complexity of the latest FPs in high-speed data streams can efficiently be determined through proper experimentation and analysis. (author)

  10. Theoretical study of time-dependent, ultrasound-induced acoustic streaming in microchannels.

    Science.gov (United States)

    Muller, Peter Barkholt; Bruus, Henrik

    2015-12-01

    Based on first- and second-order perturbation theory, we present a numerical study of the temporal buildup and decay of unsteady acoustic fields and acoustic streaming flows actuated by vibrating walls in the transverse cross-sectional plane of a long straight microchannel under adiabatic conditions and assuming temperature-independent material parameters. The unsteady streaming flow is obtained by averaging the time-dependent velocity field over one oscillation period, and as time increases, it is shown to converge towards the well-known steady time-averaged solution calculated in the frequency domain. Scaling analysis reveals that the acoustic resonance builds up much faster than the acoustic streaming, implying that the radiation force may dominate over the drag force from streaming even for small particles. However, our numerical time-dependent analysis indicates that pulsed actuation does not reduce streaming significantly due to its slow decay. Our analysis also shows that for an acoustic resonance with a quality factor Q, the amplitude of the oscillating second-order velocity component is Q times larger than the usual second-order steady time-averaged velocity component. Consequently, the well-known criterion v(1)≪c(s) for the validity of the perturbation expansion is replaced by the more restrictive criterion v(1)≪c(s)/Q. Our numerical model is available as supplemental material in the form of comsol model files and matlab scripts.

  11. 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.

  12. A Fast Density-Based Clustering Algorithm for Real-Time Internet of Things Stream

    Science.gov (United States)

    Ying Wah, Teh

    2014-01-01

    Data streams are continuously generated over time from Internet of Things (IoT) devices. The faster all of this data is analyzed, its hidden trends and patterns discovered, and new strategies created, the faster action can be taken, creating greater value for organizations. Density-based method is a prominent class in clustering data streams. It has the ability to detect arbitrary shape clusters, to handle outlier, and it does not need the number of clusters in advance. Therefore, density-based clustering algorithm is a proper choice for clustering IoT streams. Recently, several density-based algorithms have been proposed for clustering data streams. However, density-based clustering in limited time is still a challenging issue. In this paper, we propose a density-based clustering algorithm for IoT streams. The method has fast processing time to be applicable in real-time application of IoT devices. Experimental results show that the proposed approach obtains high quality results with low computation time on real and synthetic datasets. PMID:25110753

  13. A fast density-based clustering algorithm for real-time Internet of Things stream.

    Science.gov (United States)

    Amini, Amineh; Saboohi, Hadi; Wah, Teh Ying; Herawan, Tutut

    2014-01-01

    Data streams are continuously generated over time from Internet of Things (IoT) devices. The faster all of this data is analyzed, its hidden trends and patterns discovered, and new strategies created, the faster action can be taken, creating greater value for organizations. Density-based method is a prominent class in clustering data streams. It has the ability to detect arbitrary shape clusters, to handle outlier, and it does not need the number of clusters in advance. Therefore, density-based clustering algorithm is a proper choice for clustering IoT streams. Recently, several density-based algorithms have been proposed for clustering data streams. However, density-based clustering in limited time is still a challenging issue. In this paper, we propose a density-based clustering algorithm for IoT streams. The method has fast processing time to be applicable in real-time application of IoT devices. Experimental results show that the proposed approach obtains high quality results with low computation time on real and synthetic datasets.

  14. Online Class Review: Using Streaming-Media Technology

    Science.gov (United States)

    Loudon, Marc; Sharp, Mark

    2006-01-01

    We present an automated system that allows students to replay both audio and video from a large nonmajors' organic chemistry class as streaming RealMedia. Once established, this system requires no technical intervention and is virtually transparent to the instructor. This gives students access to online class review at any time. Assessment has…

  15. Real-time analytics techniques to analyze and visualize streaming data

    CERN Document Server

    Ellis, Byron

    2014-01-01

    Construct a robust end-to-end solution for analyzing and visualizing streaming data Real-time analytics is the hottest topic in data analytics today. In Real-Time Analytics: Techniques to Analyze and Visualize Streaming Data, expert Byron Ellis teaches data analysts technologies to build an effective real-time analytics platform. This platform can then be used to make sense of the constantly changing data that is beginning to outpace traditional batch-based analysis platforms. The author is among a very few leading experts in the field. He has a prestigious background in research, development,

  16. Cache Timing Analysis of LFSR-based Stream Ciphers

    DEFF Research Database (Denmark)

    Zenner, Erik; Leander, Gregor; Hawkes, Philip

    2009-01-01

    Cache timing attacks are a class of side-channel attacks that is applicable against certain software implementations. They have generated significant interest when demonstrated against the Advanced Encryption Standard (AES), but have more recently also been applied against other cryptographic...

  17. Real-time skin feature identification in a time-sequential video stream

    Science.gov (United States)

    Kramberger, Iztok

    2005-04-01

    Skin color can be an important feature when tracking skin-colored objects. Particularly this is the case for computer-vision-based human-computer interfaces (HCI). Humans have a highly developed feeling of space and, therefore, it is reasonable to support this within intelligent HCI, where the importance of augmented reality can be foreseen. Joining human-like interaction techniques within multimodal HCI could, or will, gain a feature for modern mobile telecommunication devices. On the other hand, real-time processing plays an important role in achieving more natural and physically intuitive ways of human-machine interaction. The main scope of this work is the development of a stereoscopic computer-vision hardware-accelerated framework for real-time skin feature identification in the sense of a single-pass image segmentation process. The hardware-accelerated preprocessing stage is presented with the purpose of color and spatial filtering, where the skin color model within the hue-saturation-value (HSV) color space is given with a polyhedron of threshold values representing the basis of the filter model. An adaptive filter management unit is suggested to achieve better segmentation results. This enables the adoption of filter parameters to the current scene conditions in an adaptive way. Implementation of the suggested hardware structure is given at the level of filed programmable system level integrated circuit (FPSLIC) devices using an embedded microcontroller as their main feature. A stereoscopic clue is achieved using a time-sequential video stream, but this shows no difference for real-time processing requirements in terms of hardware complexity. The experimental results for the hardware-accelerated preprocessing stage are given by efficiency estimation of the presented hardware structure using a simple motion-detection algorithm based on a binary function.

  18. 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.

  19. Real-time video streaming in mobile cloud over heterogeneous wireless networks

    Science.gov (United States)

    Abdallah-Saleh, Saleh; Wang, Qi; Grecos, Christos

    2012-06-01

    Recently, the concept of Mobile Cloud Computing (MCC) has been proposed to offload the resource requirements in computational capabilities, storage and security from mobile devices into the cloud. Internet video applications such as real-time streaming are expected to be ubiquitously deployed and supported over the cloud for mobile users, who typically encounter a range of wireless networks of diverse radio access technologies during their roaming. However, real-time video streaming for mobile cloud users across heterogeneous wireless networks presents multiple challenges. The network-layer quality of service (QoS) provision to support high-quality mobile video delivery in this demanding scenario remains an open research question, and this in turn affects the application-level visual quality and impedes mobile users' perceived quality of experience (QoE). In this paper, we devise a framework to support real-time video streaming in this new mobile video networking paradigm and evaluate the performance of the proposed framework empirically through a lab-based yet realistic testing platform. One particular issue we focus on is the effect of users' mobility on the QoS of video streaming over the cloud. We design and implement a hybrid platform comprising of a test-bed and an emulator, on which our concept of mobile cloud computing, video streaming and heterogeneous wireless networks are implemented and integrated to allow the testing of our framework. As representative heterogeneous wireless networks, the popular WLAN (Wi-Fi) and MAN (WiMAX) networks are incorporated in order to evaluate effects of handovers between these different radio access technologies. The H.264/AVC (Advanced Video Coding) standard is employed for real-time video streaming from a server to mobile users (client nodes) in the networks. Mobility support is introduced to enable continuous streaming experience for a mobile user across the heterogeneous wireless network. Real-time video stream packets

  20. Analysis and optimization techniques for real-time streaming image processing software on general purpose systems

    NARCIS (Netherlands)

    Westmijze, Mark

    2018-01-01

    Commercial Off The Shelf (COTS) Chip Multi-Processor (CMP) systems are for cost reasons often used in industry for soft real-time stream processing. COTS CMP systems typically have a low timing predictability, which makes it difficult to develop software applications for these systems with tight

  1. Automatic dataflow model extraction from modal real-time stream processing applications

    NARCIS (Netherlands)

    Geuns, S.J.; Hausmans, J.P.H.M.; Bekooij, Marco Jan Gerrit

    2013-01-01

    Many real-time stream processing applications are initially described as a sequential application containing while-loops, which execute for an unknown number of iterations. These modal applications have to be executed in parallel on an MPSoC system in order to meet their real-time throughput

  2. Defining and measuring the mean residence time of lateral surface transient storage zones in small streams

    Science.gov (United States)

    T.R. Jackson; R. Haggerty; S.V. Apte; A. Coleman; K.J. Drost

    2012-01-01

    Surface transient storage (STS) has functional significance in stream ecosystems because it increases solute interaction with sediments. After volume, mean residence time is the most important metric of STS, but it is unclear how this can be measured accurately or related to other timescales and field-measureable parameters. We studied mean residence time of lateral...

  3. Sequential Specification of Time-aware Stream Processing Applications (Extended Abstract)

    NARCIS (Netherlands)

    Geuns, S.J.; Hausmans, J.P.H.M.; Bekooij, Marco Jan Gerrit

    2012-01-01

    Automatic parallelization of Nested Loop Programs (NLPs) is an attractive method to create embedded real-time stream processing applications for multi-core systems. However, the description and parallelization of applications with a time dependent functional behavior has not been considered in NLPs.

  4. Size-selective sorting in bubble streaming flows: Particle migration on fast time scales

    Science.gov (United States)

    Thameem, Raqeeb; Rallabandi, Bhargav; Hilgenfeldt, Sascha

    2015-11-01

    Steady streaming from ultrasonically driven microbubbles is an increasingly popular technique in microfluidics because such devices are easily manufactured and generate powerful and highly controllable flows. Combining streaming and Poiseuille transport flows allows for passive size-sensitive sorting at particle sizes and selectivities much smaller than the bubble radius. The crucial particle deflection and separation takes place over very small times (milliseconds) and length scales (20-30 microns) and can be rationalized using a simplified geometric mechanism. A quantitative theoretical description is achieved through the application of recent results on three-dimensional streaming flow field contributions. To develop a more fundamental understanding of the particle dynamics, we use high-speed photography of trajectories in polydisperse particle suspensions, recording the particle motion on the time scale of the bubble oscillation. Our data reveal the dependence of particle displacement on driving phase, particle size, oscillatory flow speed, and streaming speed. With this information, the effective repulsive force exerted by the bubble on the particle can be quantified, showing for the first time how fast, selective particle migration is effected in a streaming flow. We acknowledge support by the National Science Foundation under grant number CBET-1236141.

  5. Real Time Adaptive Stream-oriented Geo-data Filtering

    Directory of Open Access Journals (Sweden)

    A. A. Golovkov

    2016-01-01

    Full Text Available The cutting-edge engineering maintenance software systems of various objects are aimed at processing of geo-location data coming from the employees’ mobile devices in real time. To reduce the amount of transmitted data such systems, usually, use various filtration methods of geo-coordinates recorded directly on mobile devices.The paper identifies the reasons for errors of geo-data coming from different sources, and proposes an adaptive dynamic method to filter geo-location data. Compared with the static method previously described in the literature [1] the approach offers to align adaptively the filtering threshold with changing characteristics of coordinates from many sources of geo-location data.To evaluate the efficiency of the developed filter method have been involved about 400 thousand points, representing motion paths of different type (on foot, by car and high-speed train and parking (indoors, outdoors, near high-rise buildings to take data from different mobile devices. Analysis of results has shown that the benefits of the proposed method are the more precise location of long parking (up to 6 hours and coordinates when user is in motion, the capability to provide steam-oriented filtering of data from different sources that allows to use the approach in geo-information systems, providing continuous monitoring of the location in streamoriented data processing in real time. The disadvantage is a little bit more computational complexity and increasing amount of points of the final track as compared to other filtration techniques.In general, the developed approach enables a significant quality improvement of displayed paths of moving mobile objects.

  6. Structure Learning in Audio

    DEFF Research Database (Denmark)

    Nielsen, Andreas Brinch

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

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

    Science.gov (United States)

    Romero-Fresco, Pablo; Fryer, Louise

    2013-01-01

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

  8. Efficient Buffer Capacity and Scheduler Setting Computation for Soft Real-Time Stream Processing Applications

    NARCIS (Netherlands)

    Bekooij, Marco; Bekooij, Marco Jan Gerrit; Wiggers, M.H.; van Meerbergen, Jef

    2007-01-01

    Soft real-time applications that process data streams can often be intuitively described as dataflow process networks. In this paper we present a novel analysis technique to compute conservative estimates of the required buffer capacities in such process networks. With the same analysis technique

  9. Hierarchical programming language for modal multi-rate real-time stream processing applications

    NARCIS (Netherlands)

    Geuns, S.J.; Hausmans, J.P.H.M.; Bekooij, Marco Jan Gerrit

    2014-01-01

    Modal multi-rate stream processing applications with real-time constraints which are executed on multi-core embedded systems often cannot be conveniently specified using current programming languages. An important issue is that sequential programming languages do not allow for convenient programming

  10. Unsupervised deep learning for real-time assessment of video streaming services

    NARCIS (Netherlands)

    Torres Vega, M.; Mocanu, D.C.; Liotta, A.

    2017-01-01

    Evaluating quality of experience in video streaming services requires a quality metric that works in real time and for a broad range of video types and network conditions. This means that, subjective video quality assessment studies, or complex objective video quality assessment metrics, which would

  11. Real-Time Clinical Decision Support System with Data Stream Mining

    Directory of Open Access Journals (Sweden)

    Yang Zhang

    2012-01-01

    Full Text Available This research aims to describe a new design of data stream mining system that can analyze medical data stream and make real-time prediction. The motivation of the research is due to a growing concern of combining software technology and medical functions for the development of software application that can be used in medical field of chronic disease prognosis and diagnosis, children healthcare, diabetes diagnosis, and so forth. Most of the existing software technologies are case-based data mining systems. They only can analyze finite and structured data set and can only work well in their early years and can hardly meet today's medical requirement. In this paper, we describe a clinical-support-system based data stream mining technology; the design has taken into account all the shortcomings of the existing clinical support systems.

  12. Mutual Information Based Dynamic Integration of Multiple Feature Streams for Robust Real-Time LVCSR

    Science.gov (United States)

    Sato, Shoei; Kobayashi, Akio; Onoe, Kazuo; Homma, Shinichi; Imai, Toru; Takagi, Tohru; Kobayashi, Tetsunori

    We present a novel method of integrating the likelihoods of multiple feature streams, representing different acoustic aspects, for robust speech recognition. The integration algorithm dynamically calculates a frame-wise stream weight so that a higher weight is given to a stream that is robust to a variety of noisy environments or speaking styles. Such a robust stream is expected to show discriminative ability. A conventional method proposed for the recognition of spoken digits calculates the weights front the entropy of the whole set of HMM states. This paper extends the dynamic weighting to a real-time large-vocabulary continuous speech recognition (LVCSR) system. The proposed weight is calculated in real-time from mutual information between an input stream and active HMM states in a searchs pace without an additional likelihood calculation. Furthermore, the mutual information takes the width of the search space into account by calculating the marginal entropy from the number of active states. In this paper, we integrate three features that are extracted through auditory filters by taking into account the human auditory system's ability to extract amplitude and frequency modulations. Due to this, features representing energy, amplitude drift, and resonant frequency drifts, are integrated. These features are expected to provide complementary clues for speech recognition. Speech recognition experiments on field reports and spontaneous commentary from Japanese broadcast news showed that the proposed method reduced error words by 9.2% in field reports and 4.7% in spontaneous commentaries relative to the best result obtained from a single stream.

  13. Adaptive DCTNet for Audio Signal Classification

    OpenAIRE

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

    2016-01-01

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

  14. A real-time remote video streaming platform for ultrasound imaging.

    Science.gov (United States)

    Ahmadi, Mehdi; Gross, Warren J; Kadoury, Samuel

    2016-08-01

    Ultrasound is a viable imaging technology in remote and resources-limited areas. Ultrasonography is a user-dependent skill which depends on a high degree of training and hands-on experience. However, there is a limited number of skillful sonographers located in remote areas. In this work, we aim to develop a real-time video streaming platform which allows specialist physicians to remotely monitor ultrasound exams. To this end, an ultrasound stream is captured and transmitted through a wireless network into remote computers, smart-phones and tablets. In addition, the system is equipped with a camera to track the position of the ultrasound probe. The main advantage of our work is using an open source platform for video streaming which gives us more control over streaming parameters than the available commercial products. The transmission delays of the system are evaluated for several ultrasound video resolutions and the results show that ultrasound videos close to the high-definition (HD) resolution can be received and displayed on an Android tablet with the delay of 0.5 seconds which is acceptable for accurate real-time diagnosis.

  15. Change Semantic Constrained Online Data Cleaning Method for Real-Time Observational Data Stream

    Science.gov (United States)

    Ding, Yulin; Lin, Hui; Li, Rongrong

    2016-06-01

    Recent breakthroughs in sensor networks have made it possible to collect and assemble increasing amounts of real-time observational data by observing dynamic phenomena at previously impossible time and space scales. Real-time observational data streams present potentially profound opportunities for real-time applications in disaster mitigation and emergency response, by providing accurate and timeliness estimates of environment's status. However, the data are always subject to inevitable anomalies (including errors and anomalous changes/events) caused by various effects produced by the environment they are monitoring. The "big but dirty" real-time observational data streams can rarely achieve their full potential in the following real-time models or applications due to the low data quality. Therefore, timely and meaningful online data cleaning is a necessary pre-requisite step to ensure the quality, reliability, and timeliness of the real-time observational data. In general, a straightforward streaming data cleaning approach, is to define various types of models/classifiers representing normal behavior of sensor data streams and then declare any deviation from this model as normal or erroneous data. The effectiveness of these models is affected by dynamic changes of deployed environments. Due to the changing nature of the complicated process being observed, real-time observational data is characterized by diversity and dynamic, showing a typical Big (Geo) Data characters. Dynamics and diversity is not only reflected in the data values, but also reflected in the complicated changing patterns of the data distributions. This means the pattern of the real-time observational data distribution is not stationary or static but changing and dynamic. After the data pattern changed, it is necessary to adapt the model over time to cope with the changing patterns of real-time data streams. Otherwise, the model will not fit the following observational data streams, which may led

  16. CHANGE SEMANTIC CONSTRAINED ONLINE DATA CLEANING METHOD FOR REAL-TIME OBSERVATIONAL DATA STREAM

    Directory of Open Access Journals (Sweden)

    Y. Ding

    2016-06-01

    Full Text Available Recent breakthroughs in sensor networks have made it possible to collect and assemble increasing amounts of real-time observational data by observing dynamic phenomena at previously impossible time and space scales. Real-time observational data streams present potentially profound opportunities for real-time applications in disaster mitigation and emergency response, by providing accurate and timeliness estimates of environment’s status. However, the data are always subject to inevitable anomalies (including errors and anomalous changes/events caused by various effects produced by the environment they are monitoring. The “big but dirty” real-time observational data streams can rarely achieve their full potential in the following real-time models or applications due to the low data quality. Therefore, timely and meaningful online data cleaning is a necessary pre-requisite step to ensure the quality, reliability, and timeliness of the real-time observational data. In general, a straightforward streaming data cleaning approach, is to define various types of models/classifiers representing normal behavior of sensor data streams and then declare any deviation from this model as normal or erroneous data. The effectiveness of these models is affected by dynamic changes of deployed environments. Due to the changing nature of the complicated process being observed, real-time observational data is characterized by diversity and dynamic, showing a typical Big (Geo Data characters. Dynamics and diversity is not only reflected in the data values, but also reflected in the complicated changing patterns of the data distributions. This means the pattern of the real-time observational data distribution is not stationary or static but changing and dynamic. After the data pattern changed, it is necessary to adapt the model over time to cope with the changing patterns of real-time data streams. Otherwise, the model will not fit the following observational

  17. Design for real-time data acquisition based on streaming technology

    International Nuclear Information System (INIS)

    Nakanishi, Hideya; Kojima, Mamoru

    2001-04-01

    For the LHD project a long-pulse plasma experiment of one-hour duration is planned. In this quasi steady-state operation, the data acquisition system will be required to continuously transfer the diagnostic data from the digitizer front-end and display them in real-time. The Compact PCI standard is used to replace the conventional CAMAC digitizers in LHD, because it provides good functionality for real-time data streaming and also a connectivity with modern PC technology. The digitizer scheme, interface to the host computer, adoption of data compression, and downstream applications are discussed in detail to design and implement this new real-time data streaming system for LHD plasma diagnostics. (author)

  18. Design of an audio advertisement dataset

    Science.gov (United States)

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

    2015-12-01

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

  19. 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...

  20. CERN automatic audio-conference service

    CERN Document Server

    Sierra Moral, R

    2010-01-01

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

  1. Detection Of Alterations In Audio Files Using Spectrograph Analysis

    Directory of Open Access Journals (Sweden)

    Anandha Krishnan G

    2015-08-01

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

  2. Applying the EBU R128 loudness standard in live-streaming sound sculptures

    DEFF Research Database (Denmark)

    Højlund, Marie Koldkjær; Riis, Morten S.; Rothmann, Daniel

    2017-01-01

    to preserve a natural sounding dynamic image from the varying sound sources that can be played back under varying conditions, an adaptation of the EBU R128 loudness measurement recommendation, originally developed for levelling non-real-time broadcast material, has been applied. The paper describes the Pure......This paper describes the development of a loudness-based compressor for live audio streams. The need for this device arose while developing the public sound art project The Overheard, which involves mixing together several live audio streams through a web based mixing interface. In order...

  3. DAFX Digital Audio Effects

    CERN Document Server

    2011-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Valenzise G

    2009-01-01

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

  5. 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…

  6. A distributed approach for optimizing cascaded classifier topologies in real-time stream mining systems.

    Science.gov (United States)

    Foo, Brian; van der Schaar, Mihaela

    2010-11-01

    In this paper, we discuss distributed optimization techniques for configuring classifiers in a real-time, informationally-distributed stream mining system. Due to the large volume of streaming data, stream mining systems must often cope with overload, which can lead to poor performance and intolerable processing delay for real-time applications. Furthermore, optimizing over an entire system of classifiers is a difficult task since changing the filtering process at one classifier can impact both the feature values of data arriving at classifiers further downstream and thus, the classification performance achieved by an ensemble of classifiers, as well as the end-to-end processing delay. To address this problem, this paper makes three main contributions: 1) Based on classification and queuing theoretic models, we propose a utility metric that captures both the performance and the delay of a binary filtering classifier system. 2) We introduce a low-complexity framework for estimating the system utility by observing, estimating, and/or exchanging parameters between the inter-related classifiers deployed across the system. 3) We provide distributed algorithms to reconfigure the system, and analyze the algorithms based on their convergence properties, optimality, information exchange overhead, and rate of adaptation to non-stationary data sources. We provide results using different video classifier systems.

  7. Modeling the time--varying subjective quality of HTTP video streams with rate adaptations.

    Science.gov (United States)

    Chen, Chao; Choi, Lark Kwon; de Veciana, Gustavo; Caramanis, Constantine; Heath, Robert W; Bovik, Alan C

    2014-05-01

    Newly developed hypertext transfer protocol (HTTP)-based video streaming technologies enable flexible rate-adaptation under varying channel conditions. Accurately predicting the users' quality of experience (QoE) for rate-adaptive HTTP video streams is thus critical to achieve efficiency. An important aspect of understanding and modeling QoE is predicting the up-to-the-moment subjective quality of a video as it is played, which is difficult due to hysteresis effects and nonlinearities in human behavioral responses. This paper presents a Hammerstein-Wiener model for predicting the time-varying subjective quality (TVSQ) of rate-adaptive videos. To collect data for model parameterization and validation, a database of longer duration videos with time-varying distortions was built and the TVSQs of the videos were measured in a large-scale subjective study. The proposed method is able to reliably predict the TVSQ of rate adaptive videos. Since the Hammerstein-Wiener model has a very simple structure, the proposed method is suitable for online TVSQ prediction in HTTP-based streaming.

  8. A reference web architecture and patterns for real-time visual analytics on large streaming data

    Science.gov (United States)

    Kandogan, Eser; Soroker, Danny; Rohall, Steven; Bak, Peter; van Ham, Frank; Lu, Jie; Ship, Harold-Jeffrey; Wang, Chun-Fu; Lai, Jennifer

    2013-12-01

    Monitoring and analysis of streaming data, such as social media, sensors, and news feeds, has become increasingly important for business and government. The volume and velocity of incoming data are key challenges. To effectively support monitoring and analysis, statistical and visual analytics techniques need to be seamlessly integrated; analytic techniques for a variety of data types (e.g., text, numerical) and scope (e.g., incremental, rolling-window, global) must be properly accommodated; interaction, collaboration, and coordination among several visualizations must be supported in an efficient manner; and the system should support the use of different analytics techniques in a pluggable manner. Especially in web-based environments, these requirements pose restrictions on the basic visual analytics architecture for streaming data. In this paper we report on our experience of building a reference web architecture for real-time visual analytics of streaming data, identify and discuss architectural patterns that address these challenges, and report on applying the reference architecture for real-time Twitter monitoring and analysis.

  9. Sound stream segregation: a neuromorphic approach to solve the "cocktail party problem" in real-time.

    Science.gov (United States)

    Thakur, Chetan Singh; Wang, Runchun M; Afshar, Saeed; Hamilton, Tara J; Tapson, Jonathan C; Shamma, Shihab A; van Schaik, André

    2015-01-01

    The human auditory system has the ability to segregate complex auditory scenes into a foreground component and a background, allowing us to listen to specific speech sounds from a mixture of sounds. Selective attention plays a crucial role in this process, colloquially known as the "cocktail party effect." It has not been possible to build a machine that can emulate this human ability in real-time. Here, we have developed a framework for the implementation of a neuromorphic sound segregation algorithm in a Field Programmable Gate Array (FPGA). This algorithm is based on the principles of temporal coherence and uses an attention signal to separate a target sound stream from background noise. Temporal coherence implies that auditory features belonging to the same sound source are coherently modulated and evoke highly correlated neural response patterns. The basis for this form of sound segregation is that responses from pairs of channels that are strongly positively correlated belong to the same stream, while channels that are uncorrelated or anti-correlated belong to different streams. In our framework, we have used a neuromorphic cochlea as a frontend sound analyser to extract spatial information of the sound input, which then passes through band pass filters that extract the sound envelope at various modulation rates. Further stages include feature extraction and mask generation, which is finally used to reconstruct the targeted sound. Using sample tonal and speech mixtures, we show that our FPGA architecture is able to segregate sound sources in real-time. The accuracy of segregation is indicated by the high signal-to-noise ratio (SNR) of the segregated stream (90, 77, and 55 dB for simple tone, complex tone, and speech, respectively) as compared to the SNR of the mixture waveform (0 dB). This system may be easily extended for the segregation of complex speech signals, and may thus find various applications in electronic devices such as for sound segregation and

  10. Type and timing of stream flow changes in urbanizing watersheds in the Eastern U.S.

    Directory of Open Access Journals (Sweden)

    Kristina G. Hopkins

    2015-06-01

    Full Text Available Abstract Linking the type and timing of hydrologic changes with patterns of urban growth is essential to identifying the underlying mechanisms that drive declines in urban aquatic ecosystems. In six urbanizing watersheds surrounding three U.S. cities (Baltimore, MD, Boston, MA, and Pittsburgh, PA, we reconstructed the history of development patterns since 1900 and assessed the magnitude and timing of stream flow changes during watershed development. Development reconstructions indicated that the majority of watershed development occurred during a period of peak population growth, typically between 1950 and 1970. Stream flow records indicated significant increases in annual frequency of high-flow events in all six watersheds and increases in annual runoff efficiency in five watersheds. Annual development intensity during the peak growth period had the strongest association with the magnitude of changes in high-flow frequency from the pre- to post-development periods. Results suggest the timing of the peak growth period is particularly important to understanding hydrologic changes, because it can set the type of stormwater infrastructure installed within a watershed. In three watersheds there was a rapid (∼10-15 years shift toward more frequent high-flow events, and in four watersheds there was a shift toward higher runoff efficiency. Breakpoint analyses indicated these shifts occurred between 1969 and 1976 for high-flow frequency and between 1962 and 1984 for runoff efficiency. Results indicated that the timing of high-flow changes were mainly driven by the development trajectory of each watershed, whereas the timing of runoff-efficiency changes were driven by a combination of development trajectories and extreme weather events. Our results underscore the need to refine the causes of urban stream degradation to incorporate the impact of gradual versus rapid urbanization on hydrologic changes and aquatic ecosystem function, as well as to

  11. Residence-time framework for modeling multicomponent reactive transport in stream hyporheic zones

    Science.gov (United States)

    Painter, S. L.; Coon, E. T.; Brooks, S. C.

    2017-12-01

    Process-based models for transport and transformation of nutrients and contaminants in streams require tractable representations of solute exchange between the stream channel and biogeochemically active hyporheic zones. Residence-time based formulations provide an alternative to detailed three-dimensional simulations and have had good success in representing hyporheic exchange of non-reacting solutes. We extend the residence-time formulation for hyporheic transport to accommodate general multicomponent reactive transport. To that end, the integro-differential form of previous residence time models is replaced by an equivalent formulation based on a one-dimensional advection dispersion equation along the channel coupled at each channel location to a one-dimensional transport model in Lagrangian travel-time form. With the channel discretized for numerical solution, the associated Lagrangian model becomes a subgrid model representing an ensemble of streamlines that are diverted into the hyporheic zone before returning to the channel. In contrast to the previous integro-differential forms of the residence-time based models, the hyporheic flowpaths have semi-explicit spatial representation (parameterized by travel time), thus allowing coupling to general biogeochemical models. The approach has been implemented as a stream-corridor subgrid model in the open-source integrated surface/subsurface modeling software ATS. We use bedform-driven flow coupled to a biogeochemical model with explicit microbial biomass dynamics as an example to show that the subgrid representation is able to represent redox zonation in sediments and resulting effects on metal biogeochemical dynamics in a tractable manner that can be scaled to reach scales.

  12. The audio expert everything you need to know about audio

    CERN Document Server

    Winer, Ethan

    2012-01-01

    The Audio Expert is a comprehensive reference that covers all aspects of audio, with many practical, as well as theoretical, explanations. Providing in-depth descriptions of how audio really works, using common sense plain-English explanations and mechanical analogies with minimal math, the book is written for people who want to understand audio at the deepest, most technical level, without needing an engineering degree. It's presented in an easy-to-read, conversational tone, and includes more than 400 figures and photos augmenting the text.The Audio Expert takes th

  13. Real-Time Management of Multimodal Streaming Data for Monitoring of Epileptic Patients.

    Science.gov (United States)

    Triantafyllopoulos, Dimitrios; Korvesis, Panagiotis; Mporas, Iosif; Megalooikonomou, Vasileios

    2016-03-01

    New generation of healthcare is represented by wearable health monitoring systems, which provide real-time monitoring of patient's physiological parameters. It is expected that continuous ambulatory monitoring of vital signals will improve treatment of patients and enable proactive personal health management. In this paper, we present the implementation of a multimodal real-time system for epilepsy management. The proposed methodology is based on a data streaming architecture and efficient management of a big flow of physiological parameters. The performance of this architecture is examined for varying spatial resolution of the recorded data.

  14. Data Centric Sensor Stream Reduction for Real-Time Applications in Wireless Sensor Networks

    Science.gov (United States)

    Aquino, Andre Luiz Lins; Nakamura, Eduardo Freire

    2009-01-01

    This work presents a data-centric strategy to meet deadlines in soft real-time applications in wireless sensor networks. This strategy considers three main aspects: (i) The design of real-time application to obtain the minimum deadlines; (ii) An analytic model to estimate the ideal sample size used by data-reduction algorithms; and (iii) Two data-centric stream-based sampling algorithms to perform data reduction whenever necessary. Simulation results show that our data-centric strategies meet deadlines without loosing data representativeness. PMID:22303145

  15. Real-Time Earthquake Intensity Estimation Using Streaming Data Analysis of Social and Physical Sensors

    Science.gov (United States)

    Kropivnitskaya, Yelena; Tiampo, Kristy F.; Qin, Jinhui; Bauer, Michael A.

    2017-06-01

    Earthquake intensity is one of the key components of the decision-making process for disaster response and emergency services. Accurate and rapid intensity calculations can help to reduce total loss and the number of casualties after an earthquake. Modern intensity assessment procedures handle a variety of information sources, which can be divided into two main categories. The first type of data is that derived from physical sensors, such as seismographs and accelerometers, while the second type consists of data obtained from social sensors, such as witness observations of the consequences of the earthquake itself. Estimation approaches using additional data sources or that combine sources from both data types tend to increase intensity uncertainty due to human factors and inadequate procedures for temporal and spatial estimation, resulting in precision errors in both time and space. Here we present a processing approach for the real-time analysis of streams of data from both source types. The physical sensor data is acquired from the U.S. Geological Survey (USGS) seismic network in California and the social sensor data is based on Twitter user observations. First, empirical relationships between tweet rate and observed Modified Mercalli Intensity (MMI) are developed using data from the M6.0 South Napa, CAF earthquake that occurred on August 24, 2014. Second, the streams of both data types are analyzed together in simulated real-time to produce one intensity map. The second implementation is based on IBM InfoSphere Streams, a cloud platform for real-time analytics of big data. To handle large processing workloads for data from various sources, it is deployed and run on a cloud-based cluster of virtual machines. We compare the quality and evolution of intensity maps from different data sources over 10-min time intervals immediately following the earthquake. Results from the joint analysis shows that it provides more complete coverage, with better accuracy and higher

  16. Wavelet-based audio embedding and audio/video compression

    Science.gov (United States)

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

    2001-12-01

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

  17. Real-Time Joint Streaming Data Processing from Social and Physical Sensors

    Science.gov (United States)

    Kropivnitskaya, Y. Y.; Qin, J.; Tiampo, K. F.; Bauer, M.

    2014-12-01

    The results of the technological breakthroughs in computing that have taken place over the last few decades makes it possible to achieve emergency management objectives that focus on saving human lives and decreasing economic effects. In particular, the integration of a wide variety of information sources, including observations from spatially-referenced physical sensors and new social media sources, enables better real-time seismic hazard analysis through distributed computing networks. The main goal of this work is to utilize innovative computational algorithms for better real-time seismic risk analysis by integrating different data sources and processing tools into streaming and cloud computing applications. The Geological Survey of Canada operates the Canadian National Seismograph Network (CNSN) with over 100 high-gain instruments and 60 low-gain or strong motion seismographs. The processing of the continuous data streams from each station of the CNSN provides the opportunity to detect possible earthquakes in near real-time. The information from physical sources is combined to calculate a location and magnitude for an earthquake. The automatically calculated results are not always sufficiently precise and prompt that can significantly reduce the response time to a felt or damaging earthquake. Social sensors, here represented as Twitter users, can provide information earlier to the general public and more rapidly to the emergency planning and disaster relief agencies. We introduce joint streaming data processing from social and physical sensors in real-time based on the idea that social media observations serve as proxies for physical sensors. By using the streams of data in the form of Twitter messages, each of which has an associated time and location, we can extract information related to a target event and perform enhanced analysis by combining it with physical sensor data. Results of this work suggest that the use of data from social media, in conjunction

  18. Predicting the Overall Spatial Quality of Automotive Audio Systems

    Science.gov (United States)

    Koya, Daisuke

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

  19. 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.

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

    International Nuclear Information System (INIS)

    Kwiatek, M

    2008-01-01

    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

  1. 106-17 Telemetry Standards Digitized Audio Telemetry Standard Chapter 5

    Science.gov (United States)

    2017-07-01

    Digitized Audio Telemetry Standard 5.1 General This chapter defines continuously variable slope delta (CVSD) modulation as the standard for digitizing...audio signal. The CVSD modulator is, in essence , a 1-bit analog-to-digital converter. The output of this 1-bit encoder is a serial bit stream, where

  2. Relationship between coronal holes and high speed streams at L1: arrival times, durations, and intensities

    Science.gov (United States)

    Luo, B.; Bu, X.; Liu, S.; Gong, J.

    2017-12-01

    Coronal holes are sources of high-speed steams (HSS) of solar wind. When coronal holes appear at mid/low latitudes on the Sun, consequential HSSs may impact Earth and cause recurrent geospace environment disturbances, such as geomagnetic storms, relativistic electron enhancements at the geosynchronous orbit, and thermosphere density enhancements. Thus, it is of interests for space weather forecasters to predict when (arrival times), how long (time durations), and how severe (intensities) HSSs may impact Earth when they notice coronal holes on the sun and are anticipating their geoeffectiveness. In this study, relationship between coronal holes and high speed streams will be statistically investigated. Several coronal hole parameters, including passage times of solar central meridian, coronal hole longitudinal widths, intensities reflected by mean brightness, are derived using Solar Dynamics Observatory (SDO)/Atmospheric Imaging Assembly (AIA) images for years 2011 to 2016. These parameters will be correlated with in-situ solar wind measurements measured at the L1 point by the ACE spacecraft, which can give some results that are useful for space weather forecaster in predicting the arrival times, durations, and intensities of coronal hole high-speed streams in about 3 days advance.

  3. Residence times and nitrate transport in ground water discharging to streams in the Chesapeake Bay Watershed

    Science.gov (United States)

    Lindsey, Bruce D.; Phillips, Scott; Donnelly, Colleen A.; Speiran, Gary K.; Plummer, Niel; Bohlke, John Karl; Focazio, Michael J.; Burton, William C.; Busenberg, Eurybiades

    2003-01-01

    One of the major water-quality problems in the Chesapeake Bay is an overabundance of nutrients from the streams and rivers that discharge to the Bay. Some of these nutrients are from nonpoint sources such as atmospheric deposition, agricultural manure and fertilizer, and septic systems. The effects of efforts to control nonpoint sources, however, can be difficult to quantify because of the lag time between changes at the land surface and the response in the base-flow (ground water) component of streams. To help resource managers understand the lag time between implementation of management practices and subsequent response in the nutrient concentrations in the base-flow component of streamflow, a study of ground-water discharge, residence time, and nitrate transport in springs throughout the Chesapeake Bay Watershed and in four smaller watersheds in selected hydrogeomorphic regions (HGMRs) was conducted. The four watersheds were in the Coastal Plain Uplands, Piedmont crystalline, Valley and Ridge carbonate, and Valley and Ridge siliciclastic HGMRs.A study of springs to estimate an apparent age of the ground water was based on analyses for concentrations of chlorofluorocarbons in water samples collected from 48 springs in the Chesapeake Bay Watershed. Results of the analysis indicate that median age for all the samples was 10 years, with the 25th percentile having an age of 7 years and the 75th percentile having an age of 13 years. Although the number of samples collected in each HGMR was limited, there did not appear to be distinct differences in the ages between the HGMRs. The ranges were similar between the major HGMRs above the Fall Line (modern to about 50 years), with only two HGMRs of small geographic extent (Piedmont carbonate and Mesozoic Lowland) having ranges of modern to about 10 years. The median values of all the HGMRs ranged from 7 to 11 years. Not enough samples were collected in the Coastal Plain for comparison. Spring samples showed slightly younger

  4. 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.

  5. Numerical and machine learning simulation of parametric distributions of groundwater residence time in streams and wells

    Science.gov (United States)

    Starn, J. J.; Belitz, K.; Carlson, C.

    2017-12-01

    Groundwater residence-time distributions (RTDs) are critical for assessing susceptibility of water resources to contamination. This novel approach for estimating regional RTDs was to first simulate groundwater flow using existing regional digital data sets in 13 intermediate size watersheds (each an average of 7,000 square kilometers) that are representative of a wide range of glacial systems. RTDs were simulated with particle tracking. We refer to these models as "general models" because they are based on regional, as opposed to site-specific, digital data. Parametric RTDs were created from particle RTDs by fitting 1- and 2-component Weibull, gamma, and inverse Gaussian distributions, thus reducing a large number of particle travel times to 3 to 7 parameters (shape, location, and scale for each component plus a mixing fraction) for each modeled area. The scale parameter of these distributions is related to the mean exponential age; the shape parameter controls departure from the ideal exponential distribution and is partly a function of interaction with bedrock and with drainage density. Given the flexible shape and mathematical similarity of these distributions, any of them are potentially a good fit to particle RTDs. The 1-component gamma distribution provided a good fit to basin-wide particle RTDs. RTDs at monitoring wells and streams often have more complicated shapes than basin-wide RTDs, caused in part by heterogeneity in the model, and generally require 2-component distributions. A machine learning model was trained on the RTD parameters using features derived from regionally available watershed characteristics such as recharge rate, material thickness, and stream density. RTDs appeared to vary systematically across the landscape in relation to watershed features. This relation was used to produce maps of useful metrics with respect to risk-based thresholds, such as the time to first exceedance, time to maximum concentration, time above the threshold

  6. Face customization in a real-time digiTV stream

    Science.gov (United States)

    Lugmayr, Artur R.; Creutzburg, Reiner; Kalli, Seppo; Tsoumanis, Andreas

    2002-03-01

    The challenge in digital, interactive TV (digiTV) is to move the consumer from the refiguration state to the configuration state, where he can influence the story flow, the choice of characters and other narrative elements. Besides restructuring narrative and interactivity methodologies, one major task is content manipulation to provide the auditorium the ability to predefine actors that it wants to have in its virtual story universe. Current solutions in broadcasting video provide content as monolithic structure, composed of graphics, narration, special effects, etc. compressed into one high bit rate MPEG-2 stream. More personalized and interactive TV requires a contemporary approach to segment video data in real-time to customize contents. Our research work emphasizes techniques for interchanging faces/bodies against virtual anchors in real-time constrained broadcasted video streams. The aim of our research paper is to show and point out solutions for realizing real-time face and avatar customization. The major task for the broadcaster is metadata extraction by applying face detection/tracking/recognition algorithms, and transmission of the information to the client side. At the client side, our system shall provide the facility to pre-select virtual avatars stored in a local database, and synchronize movements and expressions with the current digiTV contents.

  7. Machine-learning-based Brokers for Real-time Classification of the LSST Alert Stream

    Science.gov (United States)

    Narayan, Gautham; Zaidi, Tayeb; Soraisam, Monika D.; Wang, Zhe; Lochner, Michelle; Matheson, Thomas; Saha, Abhijit; Yang, Shuo; Zhao, Zhenge; Kececioglu, John; Scheidegger, Carlos; Snodgrass, Richard T.; Axelrod, Tim; Jenness, Tim; Maier, Robert S.; Ridgway, Stephen T.; Seaman, Robert L.; Evans, Eric Michael; Singh, Navdeep; Taylor, Clark; Toeniskoetter, Jackson; Welch, Eric; Zhu, Songzhe; The ANTARES Collaboration

    2018-05-01

    The unprecedented volume and rate of transient events that will be discovered by the Large Synoptic Survey Telescope (LSST) demand that the astronomical community update its follow-up paradigm. Alert-brokers—automated software system to sift through, characterize, annotate, and prioritize events for follow-up—will be critical tools for managing alert streams in the LSST era. The Arizona-NOAO Temporal Analysis and Response to Events System (ANTARES) is one such broker. In this work, we develop a machine learning pipeline to characterize and classify variable and transient sources only using the available multiband optical photometry. We describe three illustrative stages of the pipeline, serving the three goals of early, intermediate, and retrospective classification of alerts. The first takes the form of variable versus transient categorization, the second a multiclass typing of the combined variable and transient data set, and the third a purity-driven subtyping of a transient class. Although several similar algorithms have proven themselves in simulations, we validate their performance on real observations for the first time. We quantitatively evaluate our pipeline on sparse, unevenly sampled, heteroskedastic data from various existing observational campaigns, and demonstrate very competitive classification performance. We describe our progress toward adapting the pipeline developed in this work into a real-time broker working on live alert streams from time-domain surveys.

  8. Machine Learning-based Transient Brokers for Real-time Classification of the LSST Alert Stream

    Science.gov (United States)

    Narayan, Gautham; Zaidi, Tayeb; Soraisam, Monika; ANTARES Collaboration

    2018-01-01

    The number of transient events discovered by wide-field time-domain surveys already far outstrips the combined followup resources of the astronomical community. This number will only increase as we progress towards the commissioning of the Large Synoptic Survey Telescope (LSST), breaking the community's current followup paradigm. Transient brokers - software to sift through, characterize, annotate and prioritize events for followup - will be a critical tool for managing alert streams in the LSST era. Developing the algorithms that underlie the brokers, and obtaining simulated LSST-like datasets prior to LSST commissioning, to train and test these algorithms are formidable, though not insurmountable challenges. The Arizona-NOAO Temporal Analysis and Response to Events System (ANTARES) is a joint project of the National Optical Astronomy Observatory and the Department of Computer Science at the University of Arizona. We have been developing completely automated methods to characterize and classify variable and transient events from their multiband optical photometry. We describe the hierarchical ensemble machine learning algorithm we are developing, and test its performance on sparse, unevenly sampled, heteroskedastic data from various existing observational campaigns, as well as our progress towards incorporating these into a real-time event broker working on live alert streams from time-domain surveys.

  9. Internet Teleoperation of a Robot with Streaming Buffer System under Varying Time Delays

    Science.gov (United States)

    Park, Jahng-Hyon; Shin, Wanjae

    It is known that existence of irregular transmission time delay is a major bottleneck for application of advanced robot control schemes to internet telerobotic systems. In the internet teleoperation system, the irregular transmission time delay causes a critical problem, which includes instability and inaccuracy. This paper suggests a practical internet teleoperation system with streaming buffer system, which consists of a buffer, a buffer manager, and a control timer. The proposed system converts the irregular transmission time delay to a constant. So, the system effectively transmits the control input to a remote site to operate a robot stably and accurately. This feature enables short control input intervals. That means the entire system has a large control bandwidth. The validity of the proposed method is demonstrated by experiments of teleoperation from USC (University of Southern California in U. S.A.) to HYU (Hanyang Univ. in Korea) through the Internet. The proposed method is also demonstrated by experiments of teleoperation through the wireless internet.

  10. Sarcastic sentiment detection in tweets streamed in real time: a big data approach

    Directory of Open Access Journals (Sweden)

    S.K. Bharti

    2016-08-01

    Full Text Available Sarcasm is a type of sentiment where people express their negative feelings using positive or intensified positive words in the text. While speaking, people often use heavy tonal stress and certain gestural clues like rolling of the eyes, hand movement, etc. to reveal sarcastic. In the textual data, these tonal and gestural clues are missing, making sarcasm detection very difficult for an average human. Due to these challenges, researchers show interest in sarcasm detection of social media text, especially in tweets. Rapid growth of tweets in volume and its analysis pose major challenges. In this paper, we proposed a Hadoop based framework that captures real time tweets and processes it with a set of algorithms which identifies sarcastic sentiment effectively. We observe that the elapse time for analyzing and processing under Hadoop based framework significantly outperforms the conventional methods and is more suited for real time streaming tweets.

  11. Mahanaxar: quality of service guarantees in high-bandwidth, real-time streaming data storage

    Energy Technology Data Exchange (ETDEWEB)

    Bigelow, David [Los Alamos National Laboratory; Bent, John [Los Alamos National Laboratory; Chen, Hsing-Bung [Los Alamos National Laboratory; Brandt, Scott [UCSC

    2010-04-05

    Large radio telescopes, cyber-security systems monitoring real-time network traffic, and others have specialized data storage needs: guaranteed capture of an ultra-high-bandwidth data stream, retention of the data long enough to determine what is 'interesting,' retention of interesting data indefinitely, and concurrent read/write access to determine what data is interesting, without interrupting the ongoing capture of incoming data. Mahanaxar addresses this problem. Mahanaxar guarantees streaming real-time data capture at (nearly) the full rate of the raw device, allows concurrent read and write access to the device on a best-effort basis without interrupting the data capture, and retains data as long as possible given the available storage. It has built in mechanisms for reliability and indexing, can scale to meet arbitrary bandwidth requirements, and handles both small and large data elements equally well. Results from our prototype implementation shows that Mahanaxar provides both better guarantees and better performance than traditional file systems.

  12. Intelligent Stale-Frame Discards for Real-Time Video Streaming over Wireless Ad Hoc Networks

    Directory of Open Access Journals (Sweden)

    Sheu Tsang-Ling

    2009-01-01

    Full Text Available Abstract This paper presents intelligent early packet discards (I-EPD for real-time video streaming over a multihop wireless ad hoc network. In a multihop wireless ad hoc network, the quality of transferring real-time video streams could be seriously degraded, since every intermediate node (IN functionally like relay device does not possess large buffer and sufficient bandwidth. Even worse, a selected relay node could leave or power off unexpectedly, which breaks the route to destination. Thus, a stale video frame is useless even if it can reach destination after network traffic becomes smooth or failed route is reconfigured. In the proposed I-EPD, an IN can intelligently determine whether a buffered video packet should be early discarded. For the purpose of validation, we implement the I-EPD on Linux-based embedded systems. Via the comparisons of performance metrics (packet/frame discards ratios, PSNR, etc., we demonstrate that video quality over a wireless ad hoc network can be substantially improved and unnecessary bandwidth wastage is greatly reduced.

  13. Threshold responses of Amazonian stream fishes to timing and extent of deforestation.

    Science.gov (United States)

    Brejão, Gabriel L; Hoeinghaus, David J; Pérez-Mayorga, María Angélica; Ferraz, Silvio F B; Casatti, Lilian

    2017-12-06

    Deforestation is a primary driver of biodiversity change through habitat loss and fragmentation. Stream biodiversity may not respond to deforestation in a simple linear relationship. Rather, threshold responses to extent and timing of deforestation may occur. Identification of critical deforestation thresholds is needed for effective conservation and management. We tested for threshold responses of fish species and functional groups to degree of watershed and riparian zone deforestation and time since impact in 75 streams in the western Brazilian Amazon. We used remote sensing to assess deforestation from 1984 to 2011. Fish assemblages were sampled with seines and dip nets in a standardized manner. Fish species (n = 84) were classified into 20 functional groups based on ecomorphological traits associated with habitat use, feeding, and locomotion. Threshold responses were quantified using threshold indicator taxa analysis. Negative threshold responses to deforestation were common and consistently occurred at very low levels of deforestation (70% deforestation and >10 years after impact. Findings were similar at the community level for both taxonomic and functional analyses. Because most negative threshold responses occurred at low levels of deforestation and soon after impact, even minimal change is expected to negatively affect biodiversity. Delayed positive threshold responses to extreme deforestation by a few species do not offset the loss of sensitive taxa and likely contribute to biotic homogenization. © 2017 Society for Conservation Biology.

  14. Use of NTRIP for Optimizing the Decoding Algorithm for Real-Time Data Streams

    Directory of Open Access Journals (Sweden)

    Zhanke He

    2014-10-01

    Full Text Available As a network transmission protocol, Networked Transport of RTCM via Internet Protocol (NTRIP is widely used in GPS and Global Orbiting Navigational Satellite System (GLONASS Augmentation systems, such as Continuous Operational Reference System (CORS, Wide Area Augmentation System (WAAS and Satellite Based Augmentation Systems (SBAS. With the deployment of BeiDou Navigation Satellite system(BDS to serve the Asia-Pacific region, there are increasing needs for ground monitoring of the BeiDou Navigation Satellite system and the development of the high-precision real-time BeiDou products. This paper aims to optimize the decoding algorithm of NTRIP Client data streams and the user authentication strategies of the NTRIP Caster based on NTRIP. The proposed method greatly enhances the handling efficiency and significantly reduces the data transmission delay compared with the Federal Agency for Cartography and Geodesy (BKG NTRIP. Meanwhile, a transcoding method is proposed to facilitate the data transformation from the BINary EXchange (BINEX format to the RTCM format. The transformation scheme thus solves the problem of handing real-time data streams from Trimble receivers in the BeiDou Navigation Satellite System indigenously developed by China.

  15. Use of NTRIP for optimizing the decoding algorithm for real-time data streams.

    Science.gov (United States)

    He, Zhanke; Tang, Wenda; Yang, Xuhai; Wang, Liming; Liu, Jihua

    2014-10-10

    As a network transmission protocol, Networked Transport of RTCM via Internet Protocol (NTRIP) is widely used in GPS and Global Orbiting Navigational Satellite System (GLONASS) Augmentation systems, such as Continuous Operational Reference System (CORS), Wide Area Augmentation System (WAAS) and Satellite Based Augmentation Systems (SBAS). With the deployment of BeiDou Navigation Satellite system(BDS) to serve the Asia-Pacific region, there are increasing needs for ground monitoring of the BeiDou Navigation Satellite system and the development of the high-precision real-time BeiDou products. This paper aims to optimize the decoding algorithm of NTRIP Client data streams and the user authentication strategies of the NTRIP Caster based on NTRIP. The proposed method greatly enhances the handling efficiency and significantly reduces the data transmission delay compared with the Federal Agency for Cartography and Geodesy (BKG) NTRIP. Meanwhile, a transcoding method is proposed to facilitate the data transformation from the BINary EXchange (BINEX) format to the RTCM format. The transformation scheme thus solves the problem of handing real-time data streams from Trimble receivers in the BeiDou Navigation Satellite System indigenously developed by China.

  16. Modified DCTNet for audio signals classification

    Science.gov (United States)

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

    2016-10-01

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

  17. 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...

  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. Intensity Maps Production Using Real-Time Joint Streaming Data Processing From Social and Physical Sensors

    Science.gov (United States)

    Kropivnitskaya, Y. Y.; Tiampo, K. F.; Qin, J.; Bauer, M.

    2015-12-01

    Intensity is one of the most useful measures of earthquake hazard, as it quantifies the strength of shaking produced at a given distance from the epicenter. Today, there are several data sources that could be used to determine intensity level which can be divided into two main categories. The first category is represented by social data sources, in which the intensity values are collected by interviewing people who experienced the earthquake-induced shaking. In this case, specially developed questionnaires can be used in addition to personal observations published on social networks such as Twitter. These observations are assigned to the appropriate intensity level by correlating specific details and descriptions to the Modified Mercalli Scale. The second category of data sources is represented by observations from different physical sensors installed with the specific purpose of obtaining an instrumentally-derived intensity level. These are usually based on a regression of recorded peak acceleration and/or velocity amplitudes. This approach relates the recorded ground motions to the expected felt and damage distribution through empirical relationships. The goal of this work is to implement and evaluate streaming data processing separately and jointly from both social and physical sensors in order to produce near real-time intensity maps and compare and analyze their quality and evolution through 10-minute time intervals immediately following an earthquake. Results are shown for the case study of the M6.0 2014 South Napa, CA earthquake that occurred on August 24, 2014. The using of innovative streaming and pipelining computing paradigms through IBM InfoSphere Streams platform made it possible to read input data in real-time for low-latency computing of combined intensity level and production of combined intensity maps in near-real time. The results compare three types of intensity maps created based on physical, social and combined data sources. Here we correlate

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

    DEFF Research Database (Denmark)

    Korhonen, Jari; Reiter, Ulrich; Myakotnykh, Eugene

    2010-01-01

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

  1. Exploring inter-frame correlation analysis and wavelet-domain modeling for real-time caption detection in streaming video

    Science.gov (United States)

    Li, Jia; Tian, Yonghong; Gao, Wen

    2008-01-01

    In recent years, the amount of streaming video has grown rapidly on the Web. Often, retrieving these streaming videos offers the challenge of indexing and analyzing the media in real time because the streams must be treated as effectively infinite in length, thus precluding offline processing. Generally speaking, captions are important semantic clues for video indexing and retrieval. However, existing caption detection methods often have difficulties to make real-time detection for streaming video, and few of them concern on the differentiation of captions from scene texts and scrolling texts. In general, these texts have different roles in streaming video retrieval. To overcome these difficulties, this paper proposes a novel approach which explores the inter-frame correlation analysis and wavelet-domain modeling for real-time caption detection in streaming video. In our approach, the inter-frame correlation information is used to distinguish caption texts from scene texts and scrolling texts. Moreover, wavelet-domain Generalized Gaussian Models (GGMs) are utilized to automatically remove non-text regions from each frame and only keep caption regions for further processing. Experiment results show that our approach is able to offer real-time caption detection with high recall and low false alarm rate, and also can effectively discern caption texts from the other texts even in low resolutions.

  2. Real-time alpha monitoring of a radioactive liquid waste stream at Los Alamos National Laboratory

    Energy Technology Data Exchange (ETDEWEB)

    Johnson, J.D.; Whitley, C.R.; Rawool-Sullivan, M. [Los Alamos National Lab., NM (United States)

    1995-12-31

    This poster display concerns the development, installation, and testing of a real-time radioactive liquid waste monitor at Los Alamos National Laboratory (LANL). The detector system was designed for the LANL Radioactive Liquid Waste Treatment Facility so that influent to the plant could be monitored in real time. By knowing the activity of the influent, plant operators can better monitor treatment, better segregate waste (potentially), and monitor the regulatory compliance of users of the LANL Radioactive Liquid Waste Collection System. The detector system uses long-range alpha detection technology, which is a nonintrusive method of characterization that determines alpha activity on the liquid surface by measuring the ionization of ambient air. Extensive testing has been performed to ensure long-term use with a minimal amount of maintenance. The final design was a simple cost-effective alpha monitor that could be modified for monitoring influent waste streams at various points in the LANL Radioactive Liquid Waste Collection System.

  3. Null stream analysis of Pulsar Timing Array data: localisation of resolvable gravitational wave sources

    Science.gov (United States)

    Goldstein, Janna; Veitch, John; Sesana, Alberto; Vecchio, Alberto

    2018-04-01

    Super-massive black hole binaries are expected to produce a gravitational wave (GW) signal in the nano-Hertz frequency band which may be detected by pulsar timing arrays (PTAs) in the coming years. The signal is composed of both stochastic and individually resolvable components. Here we develop a generic Bayesian method for the analysis of resolvable sources based on the construction of `null-streams' which cancel the part of the signal held in common for each pulsar (the Earth-term). For an array of N pulsars there are N - 2 independent null-streams that cancel the GW signal from a particular sky location. This method is applied to the localisation of quasi-circular binaries undergoing adiabatic inspiral. We carry out a systematic investigation of the scaling of the localisation accuracy with signal strength and number of pulsars in the PTA. Additionally, we find that source sky localisation with the International PTA data release one is vastly superior than what is achieved by its constituent regional PTAs.

  4. A hierarchical model of daily stream temperature using air-water temperature synchronization, autocorrelation, and time lags

    Directory of Open Access Journals (Sweden)

    Benjamin H. Letcher

    2016-02-01

    Full Text Available Water temperature is a primary driver of stream ecosystems and commonly forms the basis of stream classifications. Robust models of stream temperature are critical as the climate changes, but estimating daily stream temperature poses several important challenges. We developed a statistical model that accounts for many challenges that can make stream temperature estimation difficult. Our model identifies the yearly period when air and water temperature are synchronized, accommodates hysteresis, incorporates time lags, deals with missing data and autocorrelation and can include external drivers. In a small stream network, the model performed well (RMSE = 0.59°C, identified a clear warming trend (0.63 °C decade−1 and a widening of the synchronized period (29 d decade−1. We also carefully evaluated how missing data influenced predictions. Missing data within a year had a small effect on performance (∼0.05% average drop in RMSE with 10% fewer days with data. Missing all data for a year decreased performance (∼0.6 °C jump in RMSE, but this decrease was moderated when data were available from other streams in the network.

  5. 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.

  6. A GIS-based groundwater travel time model to evaluate stream nitrate concentration reductions from land use change

    Science.gov (United States)

    Schilling, K.E.; Wolter, C.F.

    2007-01-01

    Excessive nitrate-nitrogen (nitrate) loss from agricultural watersheds is an environmental concern. A common conservation practice to improve stream water quality is to retire vulnerable row croplands to grass. In this paper, a groundwater travel time model based on a geographic information system (GIS) analysis of readily available soil and topographic variables was used to evaluate the time needed to observe stream nitrate concentration reductions from conversion of row crop land to native prairie in Walnut Creek watershed, Iowa. Average linear groundwater velocity in 5-m cells was estimated by overlaying GIS layers of soil permeability, land slope (surrogates for hydraulic conductivity and gradient, respectively) and porosity. Cells were summed backwards from the stream network to watershed divide to develop a travel time distribution map. Results suggested that groundwater from half of the land planted in prairie has reached the stream network during the 10 years of ongoing water quality monitoring. The mean travel time for the watershed was estimated to be 10.1 years, consistent with results from a simple analytical model. The proportion of land in the watershed and subbasins with prairie groundwater reaching the stream (10-22%) was similar to the measured reduction of stream nitrate (11-36%). Results provide encouragement that additional nitrate reductions in Walnut Creek are probable in the future as reduced nitrate groundwater from distal locations discharges to the stream network in the coming years. The high spatial resolution of the model (5-m cells) and its simplicity may make it potentially applicable for land managers interested in communicating lag time issues to the public, particularly related to nitrate concentration reductions over time. ?? 2007 Springer-Verlag.

  7. DEVELOPMENT OF AN ON-LINE, REAL-TIME ALPHA RADIATION MEASURING INSTRUMENT FOR LIQUID STREAMS

    International Nuclear Information System (INIS)

    Unknown

    1999-01-01

    The US Department of Energy (DOE) has expressed a need for an on-line, real-time instrument for assaying alpha-emitting radionuclides (uranium and the transuranics) in effluent waters leaving DOE sites to ensure compliance with regulatory limits. Due to the short range of alpha particles in water (approximately40 Im), it is necessary now to intermittently collect samples of water and send them to a central laboratory for analysis. A lengthy and costly procedure is used to separate and measure the radionuclides from each sample. Large variations in radionuclide concentrations in the water may go undetected due to the sporadic sampling. Even when detected, the reading may not be representative of the actual stream concentration. To address these issues, the Advanced Technologies Group of Thermo Power Corporation (a Thermo Electron company) is developing a real-time, field-deployable alpha monitor based on a solid-state silicon wafer semiconductor (US Patent 5,652,013 and pending, assigned to the US Department of Energy). The Thermo Water Alpha Monitor will serve to monitor effluent water streams (Subsurface Contaminants Focus Area) and will be suitable for process control of remediation as well as decontamination and decommissioning (D and D) operations, such as monitoring scrubber or rinse water radioactivity levels (Mixed Waste, Plutonium, and D and D Focus Area). It would be applicable for assaying other liquids, such as oil, or solids after proper preconditioning. Rapid isotopic alpha air monitoring is also possible using this technology. This report details the program's accomplishments to date. Most significantly, the Alpha Monitoring Instrument was successfully field demonstrated on water 100X below the Environmental Protection Agency's proposed safe drinking water limit--down to under 1 pCi/1. During the Field Test, the Alpha Monitoring Instrument successfully analyzed isotopic uranium levels on a total of five different surface water, process water, and

  8. DEVELOPMENT OF AN ON-LINE, REAL-TIME ALPHA RADIATION MEASURING INSTRUMENT FOR LIQUID STREAMS

    International Nuclear Information System (INIS)

    1996-01-01

    The Department of Energy (DOE) has expressed a need for an on-line, real-time instrument for assaying alpha-emitting radionuclides (uranium and the transuranics) in effluent waters leaving DOE sites to ensure compliance with regulatory limits. Due to the short range of alpha particles in water (approximately40 Tm), it is necessary now to intermittently collect samples of water and send them to a central laboratory for analysis. A lengthy and costly procedure is used to separate and measure the radionuclides from each sample. Large variations in radionuclide concentrations in the water may go undetected due to the sporadic sampling. Even when detected, the reading may not be representative of the actual stream concentration. To address these issues, Tecogen, a division of Thermo Power Corporation, a Thermo Electron company, is developing a real-time, field-deployable, alpha monitor based on a solid-state silicon wafer semiconductor (patent pending, to be assigned to the Department of Energy). The Thermo Alpha Monitor (TAM) will serve to monitor effluent water streams (Subsurface Contaminants Focus Area) and will be suitable for process control of remediation as well as decontamination and decommissioning operations, such as monitoring scrubber or rinse water radioactivity levels (Mixed Waste Focus Area and D and D Focus Area). It would be applicable for assaying other liquids, such as oil, or solids after proper preconditioning. Rapid isotopic alpha air monitoring is also possible using this technology. This instrument for direct counting of alpha-emitters in aqueous streams is presently being developed by Thermo Power under a development program funded by the DOE Environmental Management program (DOE-EM), administered by the Morgantown Energy Technology Center (METC). Under this contract, Thermo Power has demonstrated a solid-state, silicon-based semiconductor instrument, which uses a proprietary film-based collection system to quantitatively extract the

  9. Determining the times and distances of particle transit in a mountain stream using fallout radionuclides

    Science.gov (United States)

    Bonniwell, Everett C.; Matisoff, Gerald; Whiting, Peter J.

    1999-02-01

    Targeting of erosion and pollution control programs is much more effective if the time for fine particles to be transported through a watershed, the travel distance, the proportions of old and new sediment in suspension, and the rate of erosion of the landscape can be estimated. In this paper we present a novel technique for tracing suspended sediment in a mountain stream using fallout radionuclides sorbed to sediment. Atmospherically-delivered 7Be, 210Pb, and 137Cs accumulate in the snowpack, are released with its melting and sorb to fine particulates, a portion of which are carried downslope into stream channels. The half-life of cosmogenic 7Be is short (53.4 days), thus, sediment residing on the stream bed should contain little of the radionuclide. The different signatures of newly delivered sediment from the landscape with its 7Be tag and older untagged sediment from the channel is the basis for the tracing. The total flux of such radionuclides, compared to the inventory in the soil, permits estimates of the rates of erosion of the landscape. Fine suspended particulates in the Gold Fork River, ID, are transported downstream through the drainage in one or more steps having lengths of tens of kilometers. Length of the step decreases from about 60 km near the peak of the hydrograph to about 12 km near baseflow. The percent of sediment in suspension that is `new' (i.e., recently delivered from the landscape) ranges from 96 to 12%. The remaining sediment is resuspended older channel sediment. Residence times for particulates range from 1.6 days, early in the hydrograph at the upper site, to 103 days late in the hydrograph at the lowest elevation location. Rates of erosion of fine sediment calculated from the flux of radionuclides average 0.0023 cm/year. The long distance transport of fine particles suggests that delivery through the Gold Fork drainage to the basin outlet is fairly rapid once particles reach the channel and perhaps is also rapid in similar and

  10. 5G Terminals with Multi-Streaming Features for Real-Time Mobile Broadband Applications

    Directory of Open Access Journals (Sweden)

    T. Shuminoski

    2017-06-01

    Full Text Available In this paper we present a novel QoS framework on the network layer for 5G terminals with vertical multi-homing and multi-streaming capabilities by using radio networks aggregation. The proposed framework is leading to high performance utility networks with QoS provisioning for real-time multimedia services by achieving low packet delays, stochastic queuing network stability and highest mobile broadband capabilities i.e. bitrates. The proposed QoS algorithm is implemented within the mobile terminals on one side, and in dedicated proxy servers on mobile core network side. It is based on Lyapunov optimization techniques and it is targeted to handle simultaneously multiple multimedia service flows via multiple radio network interfaces in parallel.

  11. Time-Efficiency of Sorting Chironomidae Surface-Floating Pupal Exuviae Samples from Urban Trout Streams in Northeast Minnesota, USA

    Directory of Open Access Journals (Sweden)

    Alyssa M Anderson

    2012-10-01

    Full Text Available Collections of Chironomidae surface-floating pupal exuviae (SFPE provide an effective means of assessing water quality in streams. Although not widely used in the United States, the technique is not new and has been shown to be more cost-efficient than traditional dip-net sampling techniques in organically enriched stream in an urban landscape. The intent of this research was to document the efficiency of sorting SFPE samples relative to dip-net samples in trout streams with catchments varying in amount of urbanization and differences in impervious surface. Samples of both SFPE and dip-nets were collected from 17 sample sites located on 12 trout streams in Duluth, MN, USA. We quantified time needed to sort subsamples of 100 macroinvertebrates from dip-net samples, and less than or greater than 100 chironomid exuviae from SFPE samples. For larger samples of SFPE, the time required to subsample up to 300 exuviae was also recorded. The average time to sort subsamples of 100 specimens was 22.5 minutes for SFPE samples, compared to 32.7 minutes for 100 macroinvertebrates in dip-net samples. Average time to sort up to 300 exuviae was 37.7 minutes. These results indicate that sorting SFPE samples is more time-efficient than traditional dip-net techniques in trout streams with varying catchment characteristics.doi: 10.5324/fn.v31i0.1380.Published online: 17 October 2012.

  12. Improving Wait Times to Care for Individuals with Multimorbidities and Complex Conditions Using Value Stream Mapping

    Directory of Open Access Journals (Sweden)

    Tara Sampalli

    2015-07-01

    Full Text Available Background Recognizing the significant impact of wait times for care for individuals with complex chronic conditions, we applied a LEAN methodology, namely – an adaptation of Value Stream Mapping (VSM to meet the needs of people with multiple chronic conditions and to improve wait times without additional resources or funding. Methods Over an 18-month time period, staff applied a patient-centric approach that included LEAN methodology of VSM to improve wait times to care. Our framework of evaluation was grounded in the needs and perspectives of patients and individuals waiting to receive care. Patient centric views were obtained through surveys such as Patient Assessment of Chronic Illness Care (PACIC and process engineering based questions. In addition, LEAN methodology, VSM was added to identify non-value added processes contributing to wait times. Results The care team successfully reduced wait times to 2 months in 2014 with no wait times for care anticipated in 2015. Increased patient engagement and satisfaction are also outcomes of this innovative initiative. In addition, successful transformations and implementation have resulted in resource efficiencies without increase in costs. Patients have shown significant improvements in functional health following Integrated Chronic Care Service (ICCS intervention. The methodology will be applied to other chronic disease management areas in Capital Health and the province. Conclusion Wait times to care in the management of multimoribidities and other complex conditions can add a significant burden not only on the affected individuals but also on the healthcare system. In this study, a novel and modified LEAN methodology has been applied to embed the voice of the patient in care delivery processes and to reduce wait times to care in the management of complex chronic conditions.

  13. Efficient Audio Power Amplification - Challenges

    DEFF Research Database (Denmark)

    Andersen, Michael Andreas E.

    2005-01-01

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

  14. 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)

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

    DEFF Research Database (Denmark)

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

    2017-01-01

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

  16. DEVELOPMENT OF AN ON-LINE, REAL-TIME ALPHA RADIATION MEASURING INSTRUMENT FOR LIQUID STREAMS

    International Nuclear Information System (INIS)

    Unknown

    1999-01-01

    Thermo Power Corporation has proven the technical viability of an on-line, real-time alpha radionuclide instrument for aqueous sample analysis through laboratory and initial field tests of the instrument. The instrument has been shown to be isotonically sensitive to extremely low (ten parts per trillion, or femto Curies per liter) levels of a broad range of radioisotopes. Performance enhancement and other scaling data obtained during the course of this investigation have shown that on-line, real-time operation is possible, with a sub 30-minute response time analyzing 20 ppb (30 pCi/1) natural uranium. Now that these initial field tests in Oak Ridge, Tennessee have been successfully completed, Thermo Power plans to conduct comprehensive field tests of the instrument. The purpose of these endurance tests will be to determine the endurance characteristics of the Thermo Alpha Monitor for Water when it is used by non-Thermo Power personnel in a series of one or more extended field tests. Such endurance testing is the vital next step towards the commercialization of the Alpha Monitor. Subsequently, it will be possible to provide the DOE with an instrument that has the capability of obtaining rapid feedback about the concentrations of alpha-emitting isotope contamination in effluent water streams (Subsurface Contaminants Focus Area). It will also be useful for process control of remediation and D and D operations such as monitoring scrubber/rinse water radioactivity levels (Mixed Waste, Plutonium and D and D Focus Areas)

  17. Reducing door-to-needle times using Toyota's lean manufacturing principles and value stream analysis.

    Science.gov (United States)

    Ford, Andria L; Williams, Jennifer A; Spencer, Mary; McCammon, Craig; Khoury, Naim; Sampson, Tomoko R; Panagos, Peter; Lee, Jin-Moo

    2012-12-01

    Earlier tissue-type plasminogen activator (tPA) treatment for acute ischemic stroke increases efficacy, prompting national efforts to reduce door-to-needle times. We used lean process improvement methodology to develop a streamlined intravenous tPA protocol. In early 2011, a multidisciplinary team analyzed the steps required to treat patients with acute ischemic stroke with intravenous tPA using value stream analysis (VSA). We directly compared the tPA-treated patients in the "pre-VSA" epoch with the "post-VSA" epoch with regard to baseline characteristics, protocol metrics, and clinical outcomes. The VSA revealed several tPA protocol inefficiencies: routing of patients to room, then to CT, then back to the room; serial processing of workflow; and delays in waiting for laboratory results. On March 1, 2011, a new protocol incorporated changes to minimize delays: routing patients directly to head CT before the patient room, using parallel process workflow, and implementing point-of-care laboratories. In the pre and post-VSA epochs, 132 and 87 patients were treated with intravenous tPA, respectively. Compared with pre-VSA, door-to-needle times and percent of patients treated ≤60 minutes from hospital arrival were improved in the post-VSA epoch: 60 minutes versus 39 minutes (PLean process improvement methodology can expedite time-dependent stroke care without compromising safety.

  18. MAC-Layer Active Dropping for Real-Time Video Streaming in 4G Access Networks

    KAUST Repository

    She, James

    2010-12-01

    This paper introduces a MAC-layer active dropping scheme to achieve effective resource utilization, which can satisfy the application-layer delay for real-time video streaming in time division multiple access based 4G broadband wireless access networks. When a video frame is not likely to be reconstructed within the application-layer delay bound at a receiver for the minimum decoding requirement, the MAC-layer protocol data units of such video frame will be proactively dropped before the transmission. An analytical model is developed to evaluate how confident a video frame can be delivered within its application-layer delay bound by jointly considering the effects of time-varying wireless channel, minimum decoding requirement of each video frame, data retransmission, and playback buffer. Extensive simulations with video traces are conducted to prove the effectiveness of the proposed scheme. When compared to conventional cross-layer schemes using prioritized-transmission/retransmission, the proposed scheme is practically implementable for more effective resource utilization, avoiding delay propagation, and achieving better video qualities under certain conditions.

  19. A new time-space accounting scheme to predict stream water residence time and hydrograph source components at the watershed scale

    Science.gov (United States)

    Takahiro Sayama; Jeffrey J. McDonnell

    2009-01-01

    Hydrograph source components and stream water residence time are fundamental behavioral descriptors of watersheds but, as yet, are poorly represented in most rainfall-runoff models. We present a new time-space accounting scheme (T-SAS) to simulate the pre-event and event water fractions, mean residence time, and spatial source of streamflow at the watershed scale. We...

  20. CERN automatic audio-conference service

    International Nuclear Information System (INIS)

    Sierra Moral, Rodrigo

    2010-01-01

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

  1. CERN automatic audio-conference service

    Energy Technology Data Exchange (ETDEWEB)

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

    2010-04-01

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

  2. CERN automatic audio-conference service

    Science.gov (United States)

    Sierra Moral, Rodrigo

    2010-04-01

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

  3. 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.

  4. Characteristic time series and operation region of the system of two tank reactors (CSTR) with variable division of recirculation stream

    International Nuclear Information System (INIS)

    Merta, Henryk

    2006-01-01

    The paper deals with a system of a cascade of two tank reactors, being characterized by the variable stream of recirculating fluid at each stage. The assumed mathematical model enables one to determine the system's dynamics for the case when there is no time delay and for the opposite case. The time series of the conversion degree and of the dimensionless fluid temperature, characteristic for the system considered as well as the operation regions-the latter-basing on Feingenbaum diagrams with respect to the division ratio of the recirculating stream are presented

  5. Response and recovery of water yield and timing, stream sediment, abiotic parameters, and stream chemistry following logging

    Science.gov (United States)

    Wayne Swank; Jennifer Knoepp; James Vose; Stephanie Laseter; Jackson Webster

    2014-01-01

    Watershed ecosystem analysis provides a scientific approach to quantify and integrate resource responses to management (Hornbeck and Swank 1992) and also to address issues of resource sustainability (Christensen et. al. 1996). Philosophical components of the research approach at Coweeta are 1) the quantity, timing, and quality of streamflow provides an integrated...

  6. Autonomous watersheds: Reducing flooding and stream erosion through real-time control

    Science.gov (United States)

    Kerkez, B.; Wong, B. P.

    2017-12-01

    We introduce an analytical toolchain, based on dynamical system theory and feedback control, to determine how many control points (valves, gates, pumps, etc.) are needed to transform urban watersheds from static to adaptive. Advances and distributed sensing and control stand to fundamentally change how we manage urban watersheds. In lieu of new and costly infrastructure, the real-time control of stormwater systems will reduce flooding, mitigate stream erosion, and improve the treatment of polluted runoff. We discuss the how open source technologies, in the form of wireless sensor nodes and remotely-controllable valves (open-storm.org), have been deployed to build "smart" stormwater systems in the Midwestern US. Unlike "static" infrastructure, which cannot readily adapt to changing inputs and land uses, these distributed control assets allow entire watersheds to be reconfigured on a storm-by-storm basis. Our results show how the control of even just a few valves within urban catchments (1-10km^2) allows for the real-time "shaping" of hydrographs, which reduces downstream erosion and flooding. We also introduce an equivalence framework that can be used by decision-makers to objectively compare investments into "smart" system to more traditional solutions, such as gray and green stormwater infrastructure.

  7. Learning a Continuous-Time Streaming Video QoE Model.

    Science.gov (United States)

    Ghadiyaram, Deepti; Pan, Janice; Bovik, Alan C

    2018-05-01

    Over-the-top adaptive video streaming services are frequently impacted by fluctuating network conditions that can lead to rebuffering events (stalling events) and sudden bitrate changes. These events visually impact video consumers' quality of experience (QoE) and can lead to consumer churn. The development of models that can accurately predict viewers' instantaneous subjective QoE under such volatile network conditions could potentially enable the more efficient design of quality-control protocols for media-driven services, such as YouTube, Amazon, Netflix, and so on. However, most existing models only predict a single overall QoE score on a given video and are based on simple global video features, without accounting for relevant aspects of human perception and behavior. We have created a QoE evaluator, called the time-varying QoE Indexer, that accounts for interactions between stalling events, analyzes the spatial and temporal content of a video, predicts the perceptual video quality, models the state of the client-side data buffer, and consequently predicts continuous-time quality scores that agree quite well with human opinion scores. The new QoE predictor also embeds the impact of relevant human cognitive factors, such as memory and recency, and their complex interactions with the video content being viewed. We evaluated the proposed model on three different video databases and attained standout QoE prediction performance.

  8. Flood and Weather Monitoring Using Real-time Twitter Data Streams

    Science.gov (United States)

    Demir, I.; Sit, M. A.; Sermet, M. Y.

    2016-12-01

    Social media data is a widely used source to making inference within public crisis periods and events in disaster times. Specifically, since Twitter provides large-scale data publicly in real-time, it is one of the most extensive resources with location information. This abstract provides an overview of a real-time Twitter analysis system to support flood preparedness and response using a comprehensive information-centric flood ontology and natural language processing. Within the scope of this project, we deal with acquisition and processing of real-time Twitter data streams. System fetches the tweets with specified keywords and classifies them as related to flooding or heavy weather conditions. The system uses machine learning algorithms to discover patterns using the correlation between tweets and Iowa Flood Information System's (IFIS) extensive resources. The system uses these patterns to forecast the formation and progress of a potential future flood event. While fetching tweets, predefined hashtags are used for filtering and enhancing the relevancy for selected tweets. With this project, tweets can also be used as an alternative data source where other data sources are not sufficient for specific tasks. During the disasters, the photos that people upload alongside their tweets can be collected and placed to appropriate locations on a mapping system. This allows decision making authorities and communities to see the most recent outlook of the disaster interactively. In case of an emergency, concentration of tweets can help the authorities to determine a strategy on how to reach people most efficiently while providing them the supplies they need. Thanks to the extendable nature of the flood ontology and framework, results from this project will be a guide for other natural disasters, and will be shared with the community.

  9. An empirical method for approximating stream baseflow time series using groundwater table fluctuations

    Science.gov (United States)

    Meshgi, Ali; Schmitter, Petra; Babovic, Vladan; Chui, Ting Fong May

    2014-11-01

    Developing reliable methods to estimate stream baseflow has been a subject of interest due to its importance in catchment response and sustainable watershed management. However, to date, in the absence of complex numerical models, baseflow is most commonly estimated using statistically derived empirical approaches that do not directly incorporate physically-meaningful information. On the other hand, Artificial Intelligence (AI) tools such as Genetic Programming (GP) offer unique capabilities to reduce the complexities of hydrological systems without losing relevant physical information. This study presents a simple-to-use empirical equation to estimate baseflow time series using GP so that minimal data is required and physical information is preserved. A groundwater numerical model was first adopted to simulate baseflow for a small semi-urban catchment (0.043 km2) located in Singapore. GP was then used to derive an empirical equation relating baseflow time series to time series of groundwater table fluctuations, which are relatively easily measured and are physically related to baseflow generation. The equation was then generalized for approximating baseflow in other catchments and validated for a larger vegetation-dominated basin located in the US (24 km2). Overall, this study used GP to propose a simple-to-use equation to predict baseflow time series based on only three parameters: minimum daily baseflow of the entire period, area of the catchment and groundwater table fluctuations. It serves as an alternative approach for baseflow estimation in un-gauged systems when only groundwater table and soil information is available, and is thus complementary to other methods that require discharge measurements.

  10. Retailer's optimal credit period and cycle time in a supply chain for deteriorating items with up-stream and down-stream trade credits

    Science.gov (United States)

    Mahata, Gour Chandra

    2015-09-01

    In practice, the supplier often offers the retailers a trade credit period and the retailer in turn provides a trade credit period to her/his customer to stimulate sales and reduce inventory. From the retailer's perspective, granting trade credit not only increases sales and revenue but also increases opportunity cost (i.e., the capital opportunity loss during credit period) and default risk (i.e., the percentage that the customer will not be able to pay off his/her debt obligations). Hence, how to determine credit period is increasingly recognized as an important strategy to increase retailer's profitability. Also, the selling items such as fruits, fresh fishes, gasoline, photographic films, pharmaceuticals and volatile liquids deteriorate continuously due to evaporation, obsolescence and spoilage. In this paper, we propose an economic order quantity model for the retailer where (1) the supplier provides an up-stream trade credit and the retailer also offers a down-stream trade credit, (2) the retailer's down-stream trade credit to the buyer not only increases sales and revenue but also opportunity cost and default risk, and (3) the selling items are perishable. Under these conditions, we model the retailer's inventory system as a profit maximization problem to determine the retailer's optimal replenishment decisions under the supply chain management. We then show that the retailer's optimal credit period and cycle time not only exist but also are unique. We deduce some previously published results of other researchers as special cases. Finally, we use some numerical examples to illustrate the theoretical results.

  11. Time dependent emission line profiles in the radially streaming particle model of Seyfert galaxy nuclei and quasi-stellar objects

    Science.gov (United States)

    Hubbard, R.

    1974-01-01

    The radially-streaming particle model for broad quasar and Seyfert galaxy emission features is modified to include sources of time dependence. The results are suggestive of reported observations of multiple components, variability, and transient features in the wings of Seyfert and quasi-stellar emission lines.

  12. Instrumental Landing Using Audio Indication

    Science.gov (United States)

    Burlak, E. A.; Nabatchikov, A. M.; Korsun, O. N.

    2018-02-01

    The paper proposes an audio indication method for presenting to a pilot the information regarding the relative positions of an aircraft in the tasks of precision piloting. The implementation of the method is presented, the use of such parameters of audio signal as loudness, frequency and modulation are discussed. To confirm the operability of the audio indication channel the experiments using modern aircraft simulation facility were carried out. The simulated performed the instrument landing using the proposed audio method to indicate the aircraft deviations in relation to the slide path. The results proved compatible with the simulated instrumental landings using the traditional glidescope pointers. It inspires to develop the method in order to solve other precision piloting tasks.

  13. ENERGY STAR Certified Audio Video

    Data.gov (United States)

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

  14. Realtime Audio with Garbage Collection

    OpenAIRE

    Matheussen, Kjetil Svalastog

    2010-01-01

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

  15. Audio localization for mobile robots

    OpenAIRE

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

    2009-01-01

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

  16. 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....

  17. Perancangan Sistem Audio Mobil Berbasiskan Sistem Pakar dan Web

    Directory of Open Access Journals (Sweden)

    Djunaidi Santoso

    2011-12-01

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

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

    Directory of Open Access Journals (Sweden)

    Petar S. Aleksic

    2002-11-01

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

  19. Age and admission times as predictive factors for failure of admissions to discharge-stream short-stay units.

    Science.gov (United States)

    Shetty, Amith L; Shankar Raju, Savitha Banagar; Hermiz, Arsalan; Vaghasiya, Milan; Vukasovic, Matthew

    2015-02-01

    Discharge-stream emergency short-stay units (ESSU) improve ED and hospital efficiency. Age of patients and time of hospital presentations have been shown to correlate with increasing complexity of care. We aim to determine whether an age and time cut-off could be derived to subsequently improve short-stay unit success rates. We conducted a retrospective audit on 6703 (5522 inclusions) patients admitted to our discharge-stream short-stay unit. Patients were classified as appropriate or inappropriate admissions, and deemed successful if discharged out of the unit within 24 h; and failures if they needed inpatient admission into the hospital. We calculated short-stay unit length of stay for patients in each of these groups. A 15% failure rate was deemed as acceptable key performance indicator (KPI) for our unit. There were 197 out of 4621 (4.3%, 95% CI 3.7-4.9%) patients up to the age of 70 who failed admission to ESSU compared with 67 out of 901 (7.4%, 95% CI 5.9-9.3%, P 70 years of age have higher rates of failure after admission to discharge-stream ESSU. Although in appropriately selected discharge-stream patients, no age group or time-band of presentation was associated with increased failure rate beyond the stipulated KPI. © 2014 Australasian College for Emergency Medicine and Australasian Society for Emergency Medicine.

  20. 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,

  1. 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

  2. Importance of neutral processes varies in time and space: Evidence from dryland stream ecosystems.

    Directory of Open Access Journals (Sweden)

    Xiaoli Dong

    Full Text Available Many ecosystems experience strong temporal variability in environmental conditions; yet, a clear picture of how niche and neutral processes operate to determine community assembly in temporally variable systems remains elusive. In this study, we constructed neutral metacommunity models to assess the relative importance of neutral processes in a spatially and temporally variable ecosystem. We analyzed macroinvertebrate community data spanning multiple seasons and years from 20 sites in a Sonoran Desert river network in Arizona. The model goodness-of-fit was used to infer the importance of neutral processes. Averaging over eight stream flow conditions across three years, we found that neutral processes were more important in perennial streams than in non-perennial streams (intermittent and ephemeral streams. Averaging across perennial and non-perennial streams, we found that neutral processes were more important during very high flow and in low flow periods; whereas, at very low flows, the relative importance of neutral processes varied greatly. These findings were robust to the choice of model parameter values. Our study suggested that the net effect of disturbance on the relative importance of niche and neutral processes in community assembly varies non-monotonically with the severity of disturbance. In contrast to the prevailing view that disturbance promotes niche processes, we found that neutral processes could become more important when the severity of disturbance is beyond a certain threshold such that all organisms are adversely affected regardless of their biological traits and strategies.

  3. About subjective evaluation of adaptive video streaming

    Science.gov (United States)

    Tavakoli, Samira; Brunnström, Kjell; Garcia, Narciso

    2015-03-01

    The usage of HTTP Adaptive Streaming (HAS) technology by content providers is increasing rapidly. Having available the video content in multiple qualities, using HAS allows to adapt the quality of downloaded video to the current network conditions providing smooth video-playback. However, the time-varying video quality by itself introduces a new type of impairment. The quality adaptation can be done in different ways. In order to find the best adaptation strategy maximizing users perceptual quality it is necessary to investigate about the subjective perception of adaptation-related impairments. However, the novelties of these impairments and their comparably long time duration make most of the standardized assessment methodologies fall less suited for studying HAS degradation. Furthermore, in traditional testing methodologies, the quality of the video in audiovisual services is often evaluated separated and not in the presence of audio. Nevertheless, the requirement of jointly evaluating the audio and the video within a subjective test is a relatively under-explored research field. In this work, we address the research question of determining the appropriate assessment methodology to evaluate the sequences with time-varying quality due to the adaptation. This was done by studying the influence of different adaptation related parameters through two different subjective experiments using a methodology developed to evaluate long test sequences. In order to study the impact of audio presence on quality assessment by the test subjects, one of the experiments was done in the presence of audio stimuli. The experimental results were subsequently compared with another experiment using the standardized single stimulus Absolute Category Rating (ACR) methodology.

  4. Impact of meander geometry and stream flow events on residence times and solute transport in the intra-meander flow

    Science.gov (United States)

    Nasir Mahmood, Muhammad; Schmidt, Christian; Trauth, Nico

    2017-04-01

    Stream morphological features, in combination with hydrological variability play a key role in water and solute exchange across surface and subsurface waters. Meanders are prominent morphological features within stream systems which exhibit unique hydrodynamics. The water surface elevation difference across the inner bank of a meander induces lateral hyporheic exchange within the intra-meander region. This hyporheic flow is characterized by considerably prolonged flow paths and residence times (RT) compared to smaller scales of hyporheic exchange. In this study we examine the impact of different meander geometries on the intra-meander hyporheic flow field and solute mobilization under both steady state and transient flow conditions. We developed a number of artificial meander shape scenarios, representing various meander evolution stages, ranging from a typical initial to advanced stage (near cut off ) meander. Three dimensional steady state numerical groundwater flow simulations including the unsaturated zone were performed for the intra-meander region. The meandering stream was implemented in the model by adjusting the top layers of the modelling domain to the streambed elevation and assigning linearly decreasing head boundary conditions to the streambed cells. Residence times for the intra-meander region were computed by advective particle tracking across the inner bank of meander. Selected steady state cases were extended to transient flow simulations to evaluate the impact of stream discharge events on the temporal behavior of the water exchange and solute transport in the intra-meander region. The transient stream discharge was simulated for a number of discharge events of variable duration and peak height using the surface water model HEC-RAS. Transient hydraulic heads obtained from the surface water model were applied as transient head boundary conditions to the streambed cells of the groundwater model. A solute concentration source was added in the

  5. Aurally Aided Visual Search Performance Comparing Virtual Audio Systems

    DEFF Research Database (Denmark)

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

    2014-01-01

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

  6. Aurally Aided Visual Search Performance Comparing Virtual Audio Systems

    DEFF Research Database (Denmark)

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

    2014-01-01

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

  7. Real-Time Detection Methods to Monitor TRU Compositions in UREX+Process Streams

    Energy Technology Data Exchange (ETDEWEB)

    McDeavitt, Sean; Charlton, William; Indacochea, J Ernesto; taleyarkhan, Rusi; Pereira, Candido

    2013-03-01

    The U.S. Department of Energy has developed advanced methods for reprocessing spent nuclear fuel. The majority of this development was accomplished under the Advanced Fuel Cycle Initiative (AFCI), building on the strong legacy of process development R&D over the past 50 years. The most prominent processing method under development is named UREX+. The name refers to a family of processing methods that begin with the Uranium Extraction (UREX) process and incorporate a variety of other methods to separate uranium, selected fission products, and the transuranic (TRU) isotopes from dissolved spent nuclear fuel. It is important to consider issues such as safeguards strategies and materials control and accountability methods. Monitoring of higher actinides during aqueous separations is a critical research area. By providing on-line materials accountability for the processes, covert diversion of the materials streams becomes much more difficult. The importance of the nuclear fuel cycle continues to rise on national and international agendas. The U.S. Department of Energy is evaluating and developing advanced methods for safeguarding nuclear materials along with instrumentation in various stages of the fuel cycle, especially in material balance areas (MBAs) and during reprocessing of used nuclear fuel. One of the challenges related to the implementation of any type of MBA and/or reprocessing technology (e.g., PUREX or UREX) is the real-time quantification and control of the transuranic (TRU) isotopes as they move through the process. Monitoring of higher actinides from their neutron emission (including multiplicity) and alpha signatures during transit in MBAs and in aqueous separations is a critical research area. By providing on-line real-time materials accountability, diversion of the materials becomes much more difficult. The objective of this consortium was to develop real time detection methods to monitor the efficacy of the UREX+ process and to safeguard the separated

  8. Snow Cover, Snowmelt Timing and Stream Power in the Wind River Range, Wyoming

    Science.gov (United States)

    Hall, Dorothy K.; Foster, James L.; DiGirolamo, Nicolo E.; Riggs, George A.

    2011-01-01

    Earlier onset of springtime weather, including earlier snowmelt, has been documented in the western United States over at least the last 50 years. Because the majority (is greater than 70%) of the water supply in the western U.S. comes from snowmelt, analysis of the declining spring snowpack (and shrinking glaciers) has important implications for the management of streamflow. The amount of water in a snowpack influences stream discharge which can also influence erosion and sediment transport by changing stream power, or the rate at which a stream can do work, such as move sediment and erode the stream bed. The focus of this work is the Wind River Range (WRR) in west-central Wyoming. Ten years of Moderate-Resolution Imaging Spectroradiometer (MODIS) snow-cover, cloud-gap-filled (CGF) map products and 30 years of discharge and meteorological station data are studied. Streamflow data from streams in WRR drainage basins show lower annual discharge and earlier snowmelt in the decade of the 2000s than in the previous three decades, though no trend of either lower streamflow or earlier snowmelt was observed within the decade of the 2000s. Results show a statistically-significant trend at the 95% confidence level (or higher) of increasing weekly maximum air temperature (for three out of the five meteorological stations studied) in the decade of the 1970s, and also for the 40-year study period as a whole. The extent of snow-cover (percent of basin covered) derived from the lowest elevation zone (2500-3000 m) of the WRR, using MODIS CGF snow-cover maps, is strongly correlated with maximum monthly discharge on 30 April, where Spearman's Rank correlation, rs,=0.89 for the decade of the 2000s. We also investigated stream power for Bull Lake Creek above Bull Lake; and found a trend (significant at the 90% confidence level) toward reduced stream power from 1970 to 2009. Observed changes in streamflow and stream power may be related to increasing weekly maximum air temperature

  9. Video Streaming in Online Learning

    Science.gov (United States)

    Hartsell, Taralynn; Yuen, Steve Chi-Yin

    2006-01-01

    The use of video in teaching and learning is a common practice in education today. As learning online becomes more of a common practice in education, streaming video and audio will play a bigger role in delivering course materials to online learners. This form of technology brings courses alive by allowing online learners to use their visual and…

  10. 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...

  11. Precision Scaling of Neural Networks for Efficient Audio Processing

    OpenAIRE

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

    2017-01-01

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

  12. 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

  13. 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

  14. Theoretical study of time-dependent, ultrasound-induced acoustic streaming in microchannels

    DEFF Research Database (Denmark)

    Muller, Peter Barkholt; Bruus, Henrik

    2015-01-01

    Based on first- and second-order perturbation theory, we present a numerical study of the temporal buildup and decay of unsteady acoustic fields and acoustic streaming flows actuated by vibrating walls in the transverse cross-sectional plane of a long straight microchannel under adiabatic...

  15. Long-term monitoring of streambed sedimentation and scour in a dynamic stream based on streambed temperature time series.

    Science.gov (United States)

    Sebok, Eva; Engesgaard, Peter; Duque, Carlos

    2017-08-24

    This study presented the monitoring and quantification of streambed sedimentation and scour in a stream with dynamically changing streambed based on measured phase and amplitude of the diurnal signal of sediment temperature time series. With the applied method, changes in streambed elevation were estimated on a sub-daily scale with 2-h intervals without continuous maintenance of the measurement system, thus making both high temporal resolution and long-term monitoring of streambed elevations possible. Estimates of streambed elevation showed that during base flow conditions streambed elevation fluctuates by 2-3 cm. Following high stream stages, scouring of 2-5 cm can be observed even at areas with low stream flow and weak currents. Our results demonstrate that weather variability can induce significant changes in the stream water and consequently sediment temperatures influencing the diurnal temperature signal in such an extent that the sediment thickness between paired temperature sensors were overestimated by up to 8 cm. These observations have significant consequences on the design of vertical sensor spacing in high-flux environments and in climates with reduced diurnal variations in air temperature.

  16. Development and test of a free-streaming readout chain for the CBM time of flight wall

    International Nuclear Information System (INIS)

    Loizeau, Pierre-Alain

    2014-01-01

    This thesis presents the development and test of a free-streaming readout chain for the Time of Flight (TOF) Wall of the Compressed Baryonic Matter (CBM) experiment. In order to contribute to the exploration of the phase diagram of strongly interacting matter, CBM aims at the measurement of rare probes, whose yields and phase space distributions are significantly influenced by their environment. Many of the possible signals, of which the antiprotons was investigated within this thesis, require an excellent Particle Identification (PID) and a new readout paradigm called free-streaming. In CBM, the PID for charged particles is provided by a TOF wall based on Multi-gap Resistive Plate Chambers (MRPC). Within the thesis, a central component of the TOF readout chain, the free-streaming ASIC-TDC, was evaluated and pushed from the prototype level to a close to final design, for which it could be demonstrated that it fulfill all the CBM requirements: resolution, rate capability and stability. Additionally, the CBM TOF software in the CBMROOT software framework was reorganized to merge the processing and analysis of real and simulated data. A data unpacker and a realistic digitizer were implemented with a common output data format. The digitizer was used to estimate the data rates and number of components in a free-streaming readout chain for the full wall.

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

    Science.gov (United States)

    George, Rohini

    function could be approximated to a normal distribution function. A statistical analysis was also performed to investigate if a patient's physical, tumor or general characteristics played a role in identifying whether he/she responded positively to the coaching type---signified by a reduction in the variability of respiratory motion. The analysis demonstrated that, although there were some characteristics like disease type and dose per fraction that were significant with respect to time-independent analysis, there were no significant time trends observed for the inter-session or intra-session analysis. Based on patient feedback with the existing audio-visual biofeedback system used for the study and research performed on other feedback systems, an improved audio-visual biofeedback system was designed. It is hoped the widespread clinical implementation of audio-visual biofeedback for radiotherapy will improve the accuracy of lung cancer radiotherapy.

  18. 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

  19. Advances in audio source seperation and multisource audio content retrieval

    Science.gov (United States)

    Vincent, Emmanuel

    2012-06-01

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

  20. Beyond Rating Curves: Time Series Models for in-Stream Turbidity Prediction

    Science.gov (United States)

    Wang, L.; Mukundan, R.; Zion, M.; Pierson, D. C.

    2012-12-01

    The New York City Department of Environmental Protection (DEP) manages New York City's water supply, which is comprised of over 20 reservoirs and supplies over 1 billion gallons of water per day to more than 9 million customers. DEP's "West of Hudson" reservoirs located in the Catskill Mountains are unfiltered per a renewable filtration avoidance determination granted by the EPA. While water quality is usually pristine, high volume storm events occasionally cause the reservoirs to become highly turbid. A logical strategy for turbidity control is to temporarily remove the turbid reservoirs from service. While effective in limiting delivery of turbid water and reducing the need for in-reservoir alum flocculation, this strategy runs the risk of negatively impacting water supply reliability. Thus, it is advantageous for DEP to understand how long a particular turbidity event will affect their system. In order to understand the duration, intensity and total load of a turbidity event, predictions of future in-stream turbidity values are important. Traditionally, turbidity predictions have been carried out by applying streamflow observations/forecasts to a flow-turbidity rating curve. However, predictions from rating curves are often inaccurate due to inter- and intra-event variability in flow-turbidity relationships. Predictions can be improved by applying an autoregressive moving average (ARMA) time series model in combination with a traditional rating curve. Since 2003, DEP and the Upstate Freshwater Institute have compiled a relatively consistent set of 15-minute turbidity observations at various locations on Esopus Creek above Ashokan Reservoir. Using daily averages of this data and streamflow observations at nearby USGS gauges, flow-turbidity rating curves were developed via linear regression. Time series analysis revealed that the linear regression residuals may be represented using an ARMA(1,2) process. Based on this information, flow-turbidity regressions with

  1. 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…

  2. 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...

  3. Sound stream segregation: a neuromorphic approach to solve the “cocktail party problem” in real-time

    Science.gov (United States)

    Thakur, Chetan Singh; Wang, Runchun M.; Afshar, Saeed; Hamilton, Tara J.; Tapson, Jonathan C.; Shamma, Shihab A.; van Schaik, André

    2015-01-01

    The human auditory system has the ability to segregate complex auditory scenes into a foreground component and a background, allowing us to listen to specific speech sounds from a mixture of sounds. Selective attention plays a crucial role in this process, colloquially known as the “cocktail party effect.” It has not been possible to build a machine that can emulate this human ability in real-time. Here, we have developed a framework for the implementation of a neuromorphic sound segregation algorithm in a Field Programmable Gate Array (FPGA). This algorithm is based on the principles of temporal coherence and uses an attention signal to separate a target sound stream from background noise. Temporal coherence implies that auditory features belonging to the same sound source are coherently modulated and evoke highly correlated neural response patterns. The basis for this form of sound segregation is that responses from pairs of channels that are strongly positively correlated belong to the same stream, while channels that are uncorrelated or anti-correlated belong to different streams. In our framework, we have used a neuromorphic cochlea as a frontend sound analyser to extract spatial information of the sound input, which then passes through band pass filters that extract the sound envelope at various modulation rates. Further stages include feature extraction and mask generation, which is finally used to reconstruct the targeted sound. Using sample tonal and speech mixtures, we show that our FPGA architecture is able to segregate sound sources in real-time. The accuracy of segregation is indicated by the high signal-to-noise ratio (SNR) of the segregated stream (90, 77, and 55 dB for simple tone, complex tone, and speech, respectively) as compared to the SNR of the mixture waveform (0 dB). This system may be easily extended for the segregation of complex speech signals, and may thus find various applications in electronic devices such as for sound segregation

  4. Sound stream segregation: a neuromorphic approach to solve the ‘cocktail party problem’ in real-time

    Directory of Open Access Journals (Sweden)

    Chetan Singh Thakur

    2015-09-01

    Full Text Available The human auditory system has the ability to segregate complex auditory scenes into a foreground component and a background, allowing us to listen to specific speech sounds from a mixture of sounds. Selective attention plays a crucial role in this process, colloquially known as the ‘cocktail party effect’. It has not been possible to build a machine that can emulate this human ability in real-time. Here, we have developed a framework for the implementation of a neuromorphic sound segregation algorithm in a Field Programmable Gate Array (FPGA. This algorithm is based on the principles of temporal coherence and uses an attention signal to separate a target sound stream from background noise. Temporal coherence implies that auditory features belonging to the same sound source are coherently modulated and evoke highly correlated neural response patterns. The basis for this form of sound segregation is that responses from pairs of channels that are strongly positively correlated belong to the same stream, while channels that are uncorrelated or anti-correlated belong to different streams. In our framework, we have used a neuromorphic cochlea as a frontend sound analyser to extract spatial information of the sound input, which then passes through band pass filters that extract the sound envelope at various modulation rates. Further stages include feature extraction and mask generation, which is finally used to reconstruct the targeted sound. Using sample tonal and speech mixtures, we show that our FPGA architecture is able to segregate sound sources in real-time. The accuracy of segregation is indicated by the high signal-to-noise ratio (SNR of the segregated stream (90, 77 and 55 dB for simple tone, complex tone and speech, respectively as compared to the SNR of the mixture waveform (0 dB. This system may be easily extended for the segregation of complex speech signals, and may thus find various applications in electronic devices such as for

  5. Perancangan Radio Streaming Edukasi (Studi Kasus Balai Pengembangan Media Radio YOGYAKARTA)

    OpenAIRE

    Nurwulan, Ayu Isni; Paputungan, Irving Vitra

    2009-01-01

    Pendidikan berkualitas sudah sewajarnya bisa dinikmati secara merata oleh semua orang. Mediapembelajaran secara audio yang selama ini disampaikan masih memiliki banyak keterbatasan, terutama padalingkup wilayah penyampaian. Dalam makalah ini, sebuah media pendidikan berbasis audio dengan cara laindiusulkan. Media tersebut bernama radio streaming. Pembuatan radio streaming memerlukan banyak analisissehingga perancangannya tepat. Hasil analisis dan perancangan yang disampaikan dalam makalah ini...

  6. Streaming Media Seminar--Effective Development and Distribution of Streaming Multimedia in Education

    Science.gov (United States)

    Mainhart, Robert; Gerraughty, James; Anderson, Kristine M.

    2004-01-01

    Concisely defined, "streaming media" is moving video and/or audio transmitted over the Internet for immediate viewing/listening by an end user. However, at Saint Francis University's Center of Excellence for Remote and Medically Under-Served Areas (CERMUSA), streaming media is approached from a broader perspective. The working definition includes…

  7. Real-time video streaming system for LHD experiment using IP multicast

    International Nuclear Information System (INIS)

    Emoto, Masahiko; Yamamoto, Takashi; Yoshida, Masanobu; Nagayama, Yoshio; Hasegawa, Makoto

    2009-01-01

    In order to accomplish smooth cooperation research, remote participation plays an important role. For this purpose, the authors have been developing various applications for remote participation for the LHD (Large Helical Device) experiments, such as Web interface for visualization of acquired data. The video streaming system is one of them. It is useful to grasp the status of the ongoing experiment remotely, and we provide the video images displayed in the control room to the remote users. However, usual streaming servers cannot send video images without delay. The delay changes depending on how to send the images, but even a little delay might become critical if the researchers use the images to adjust the diagnostic devices. One of the main causes of delay is the procedure of compressing and decompressing the images. Furthermore, commonly used video compression method is lossy; it removes less important information to reduce the size. However, lossy images cannot be used for physical analysis because the original information is lost. Therefore, video images for remote participation should be sent without compression in order to minimize the delay and to supply high quality images durable for physical analysis. However, sending uncompressed video images requires large network bandwidth. For example, sending 5 frames of 16bit color SXGA images a second requires 100Mbps. Furthermore, the video images must be sent to several remote sites simultaneously. It is hard for a server PC to handle such a large data. To cope with this problem, the authors adopted IP multicast to send video images to several remote sites at once. Because IP multicast packets are sent only to the network on which the clients want the data; the load of the server does not depend on the number of clients and the network load is reduced. In this paper, the authors discuss the feasibility of high bandwidth video streaming system using IP multicast. (author)

  8. Real-time detection and classification of anomalous events in streaming data

    Science.gov (United States)

    Ferragut, Erik M.; Goodall, John R.; Iannacone, Michael D.; Laska, Jason A.; Harrison, Lane T.

    2016-04-19

    A system is described for receiving a stream of events and scoring the events based on anomalousness and maliciousness (or other classification). The events can be displayed to a user in user-defined groupings in an animated fashion. The system can include a plurality of anomaly detectors that together implement an algorithm to identify low probability events and detect atypical traffic patterns. The atypical traffic patterns can then be classified as being of interest or not. In one particular example, in a network environment, the classification can be whether the network traffic is malicious or not.

  9. Application of Genetic Programing to Develop a Modular Model for the Simulation of Stream Flow Time Series

    Science.gov (United States)

    Meshgi, A.; Babovic, V.; Chui, T. F. M.; Schmitter, P.

    2014-12-01

    Developing reliable methods to estimate stream flow has been a subject of interest due to its importance in planning, design and management of water resources within a basin. Machine learning tools such as Artificial Neural Network (ANN) and Genetic Programming (GP) have been widely applied for rainfall-runoff modeling as they require less computational time as compared to physically-based models. As GP is able to generate a function with understandable structure, it may offer advantages over other data driven techniques and therefore has been used in different studies to generate rainfall-runoff functions. However, to date, proposed formulations only contain rainfall and/or streamflow data and consequently are local and cannot be generalized and adopted in other catchments which have different physical characteristics. This study investigated the capability of GP in developing a physically interpretable model with understandable structure to simulate stream flow based on hydrological parameters (e.g. precipitation) and catchment conditions (e.g., initial groundwater table elevation and area of the catchment) by following a modular approach. The modular model resulted in two sub-models where the baseflow was first predicted and the direct runoff was then estimated for a semi-urban catchment in Singapore. The simulated results matched very well with observed data in both the training and the testing of data sets, giving NSEs of 0.97 and 0.96 respectively demonstrated the successful estimation of stream flow using the modular model derived in this study. The results of this study indicate that GP is an effective tool in developing a physically interpretable model with understandable structure to simulate stream flow that can be transferred to other catchments.

  10. Bit rates in audio source coding

    NARCIS (Netherlands)

    Veldhuis, Raymond N.J.

    1992-01-01

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

  11. Audio Frequency Analysis in Mobile Phones

    Science.gov (United States)

    Aguilar, Horacio Munguía

    2016-01-01

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

  12. Medical education in India: Time to encourage cross-talk between different streams

    Directory of Open Access Journals (Sweden)

    Kishor Patwardhan

    2013-01-01

    Full Text Available Currently, India recognizes five different healthcare systems, collectively known as AYUSH (Ayurveda, Yoga, Unani, Siddha, Homeopathy, along with the conventional biomedicine. These systems have their own institutionalized structure for monitoring medical education and practice. However, because of the ′parallel′ kind of policy model that is followed in India, there is no formal provision for any cross-talk between the professionals belonging to these different streams. This situation has not only given rise to mutual misgivings among these professionals regarding the strengths and weaknesses of each other, but also has led to a poor appreciation of the historical and socio-cultural connections these streams share with the community at large. To tackle these issues and to promote adequate participation of biomedicine experts in AYUSH-related research projects, ′introduction of an AYUSH module in the current curriculum of MBBS (Bachelor of Medicine and Bachelor of Surgery program′ has been proposed in this communication along with a possible roadmap for its implementation. It is also suggested that the experts in biomedicine be engaged for training AYUSH graduates in their respective specialties so that quality AYUSH education may be ensured.

  13. A time-varying subjective quality model for mobile streaming videos with stalling events

    Science.gov (United States)

    Ghadiyaram, Deepti; Pan, Janice; Bovik, Alan C.

    2015-09-01

    Over-the-top mobile video streaming is invariably influenced by volatile network conditions which cause playback interruptions (stalling events), thereby impairing users' quality of experience (QoE). Developing models that can accurately predict users' QoE could enable the more efficient design of quality-control protocols for video streaming networks that reduce network operational costs while still delivering high-quality video content to the customers. Existing objective models that predict QoE are based on global video features, such as the number of stall events and their lengths, and are trained and validated on a small pool of ad hoc video datasets, most of which are not publicly available. The model we propose in this work goes beyond previous models as it also accounts for the fundamental effect that a viewer's recent level of satisfaction or dissatisfaction has on their overall viewing experience. In other words, the proposed model accounts for and adapts to the recency, or hysteresis effect caused by a stall event in addition to accounting for the lengths, frequency of occurrence, and the positions of stall events - factors that interact in a complex way to affect a user's QoE. On the recently introduced LIVE-Avvasi Mobile Video Database, which consists of 180 distorted videos of varied content that are afflicted solely with over 25 unique realistic stalling events, we trained and validated our model to accurately predict the QoE, attaining standout QoE prediction performance.

  14. The influence of stream bed geomorphology on chemical species within the hyporheic zone over time and space

    Science.gov (United States)

    Quick, A. M.; Reeder, W. J.; Farrell, T. B.; Benner, S. G.; Tonina, D.; Feris, K. P.

    2017-12-01

    The hyporheic zone is well established as an important zone of biogeochemical activity in streams and rivers. Multiple large scale flume experiments were carried out to mimic bedform-controlled hyporheic zones in small streams. The laboratory setting allowed for geochemical measurement resolution and replicates that would not be possible in a natural setting. Two flume experiments that consisted of three small streams with variable sizes of bedform dunes were carried out in which chemical species were measured in the surface water and along hyporheic flow lines in the subsurface. The species measured included dissolved oxygen, pH, alkalinity, major cations (Na+, Mg2+, Ca2+, K+, Si4+, Al3+), anions (NO3-, NO2-, SO42-, PO43-, Cl-), and many trace elements (As, Sr, Co, Ni, Cu, Zn, Pb, U, V). Observed spatial and temporal trends reflect microbiological processes, changing redox conditions, and chemical weathering. In general, microbial respiration causes DO to decrease with residence time, leading to aerobic and anaerobic zones that influence redox-sensitive species and pH gradients that influence mineral solubility. Most other species concentrations, including those of major cations and trace elements, increase with residence time and generally decrease over time elapsed during the experiment. The different dune morphologies dictate flow velocities in the hyporheic zone; for most species, steeper dunes with higher velocities had lower concentrations at the end of the experiment, indicating the role of dune shape in the weathering rates of minerals in hyporheic sediment and the concentrations of dissolved species entering the surface water over time. Many of the observed trends can be applied, at least qualitatively, to understanding how these species will behave in natural settings. This insight will contribute to the understanding of many of the applications of the hyporheic zone (e.g. bioremediation, habitat, greenhouse gas emissions, etc.).

  15. Task-oriented quality assessment and adaptation in real-time mission critical video streaming applications

    Science.gov (United States)

    Nightingale, James; Wang, Qi; Grecos, Christos

    2015-02-01

    In recent years video traffic has become the dominant application on the Internet with global year-on-year increases in video-oriented consumer services. Driven by improved bandwidth in both mobile and fixed networks, steadily reducing hardware costs and the development of new technologies, many existing and new classes of commercial and industrial video applications are now being upgraded or emerging. Some of the use cases for these applications include areas such as public and private security monitoring for loss prevention or intruder detection, industrial process monitoring and critical infrastructure monitoring. The use of video is becoming commonplace in defence, security, commercial, industrial, educational and health contexts. Towards optimal performances, the design or optimisation in each of these applications should be context aware and task oriented with the characteristics of the video stream (frame rate, spatial resolution, bandwidth etc.) chosen to match the use case requirements. For example, in the security domain, a task-oriented consideration may be that higher resolution video would be required to identify an intruder than to simply detect his presence. Whilst in the same case, contextual factors such as the requirement to transmit over a resource-limited wireless link, may impose constraints on the selection of optimum task-oriented parameters. This paper presents a novel, conceptually simple and easily implemented method of assessing video quality relative to its suitability for a particular task and dynamically adapting videos streams during transmission to ensure that the task can be successfully completed. Firstly we defined two principle classes of tasks: recognition tasks and event detection tasks. These task classes are further subdivided into a set of task-related profiles, each of which is associated with a set of taskoriented attributes (minimum spatial resolution, minimum frame rate etc.). For example, in the detection class

  16. Identifying Unsafe Videos on Online Public Media using Real-time Crowdsourcing

    OpenAIRE

    Mridha, Sankar Kumar; Sarkar, Braznev; Chatterjee, Sujoy; Bhattacharyya, Malay

    2017-01-01

    Due to the significant growth of social networking and human activities through the web in recent years, attention to analyzing big data using real-time crowdsourcing has increased. This data may appear in the form of streaming images, audio or videos. In this paper, we address the problem of deciding the appropriateness of streaming videos in public media with the help of crowdsourcing in real-time.

  17. The Effect of Cell Phone Conversation on Drivers’ Reaction Time to Audio Stimulus: Investigating the Theory of Multiple Resources and Central Resource of Attention

    Directory of Open Access Journals (Sweden)

    Seyed Kazem Mousavi-Sadati

    2011-01-01

    Full Text Available Objective: This research was aimed at investigating the theory of multiple resources and central resource of attention on secondary task performance of talking with two types of cell phone during driving. Materials & Methods: Using disposal sampling, 25 male participants were selected and their reaction to auditory stimulus in three different driving conditions (no conversation with phone, conversation with handheld phone and hands-free phone were recorded. Driving conditions have been changed from a participant to another participant in order to control the sequence of tests and participants familiarity with the test conditions. Results: the results of data analysis with descriptive statistics and Mauchly’s Test of Sphericity, One- factor repeated measures ANOVA and Paired-Samples T test showed that different driving conditions can affect the reaction time (P0.001. Phone Conversation with hands-free phone increases drivers’ simple reaction time to auditory stimulus (P<0.001. Using handheld phone does not increase drivers’ reaction time to auditory stimulus over hands-free phone (P<0.001. Conclusion: The results confirmed that the performance quality of dual tasks and multiple tasks can be predicted by Four-dimensional multiple resources model of attention and all traffic laws in connection with the handheld phone also have to be spread to the use of hands-free phone.

  18. Multiple frequency audio signal communication as a mechanism for neurophysiology and video data synchronization.

    Science.gov (United States)

    Topper, Nicholas C; Burke, Sara N; Maurer, Andrew Porter

    2014-12-30

    Current methods for aligning neurophysiology and video data are either prepackaged, requiring the additional purchase of a software suite, or use a blinking LED with a stationary pulse-width and frequency. These methods lack significant user interface for adaptation, are expensive, or risk a misalignment of the two data streams. A cost-effective means to obtain high-precision alignment of behavioral and neurophysiological data is obtained by generating an audio-pulse embedded with two domains of information, a low-frequency binary-counting signal and a high, randomly changing frequency. This enabled the derivation of temporal information while maintaining enough entropy in the system for algorithmic alignment. The sample to frame index constructed using the audio input correlation method described in this paper enables video and data acquisition to be aligned at a sub-frame level of precision. Traditionally, a synchrony pulse is recorded on-screen via a flashing diode. The higher sampling rate of the audio input of the camcorder enables the timing of an event to be detected with greater precision. While on-line analysis and synchronization using specialized equipment may be the ideal situation in some cases, the method presented in the current paper presents a viable, low cost alternative, and gives the flexibility to interface with custom off-line analysis tools. Moreover, the ease of constructing and implements this set-up presented in the current paper makes it applicable to a wide variety of applications that require video recording. Copyright © 2014 Elsevier B.V. All rights reserved.

  19. Promoting smoke-free homes: a novel behavioral intervention using real-time audio-visual feedback on airborne particle levels.

    Directory of Open Access Journals (Sweden)

    Neil E Klepeis

    Full Text Available Interventions are needed to protect the health of children who live with smokers. We pilot-tested a real-time intervention for promoting behavior change in homes that reduces second hand tobacco smoke (SHS levels. The intervention uses a monitor and feedback system to provide immediate auditory and visual signals triggered at defined thresholds of fine particle concentration. Dynamic graphs of real-time particle levels are also shown on a computer screen. We experimentally evaluated the system, field-tested it in homes with smokers, and conducted focus groups to obtain general opinions. Laboratory tests of the monitor demonstrated SHS sensitivity, stability, precision equivalent to at least 1 µg/m(3, and low noise. A linear relationship (R(2 = 0.98 was observed between the monitor and average SHS mass concentrations up to 150 µg/m(3. Focus groups and interviews with intervention participants showed in-home use to be acceptable and feasible. The intervention was evaluated in 3 homes with combined baseline and intervention periods lasting 9 to 15 full days. Two families modified their behavior by opening windows or doors, smoking outdoors, or smoking less. We observed evidence of lower SHS levels in these homes. The remaining household voiced reluctance to changing their smoking activity and did not exhibit lower SHS levels in main smoking areas or clear behavior change; however, family members expressed receptivity to smoking outdoors. This study established the feasibility of the real-time intervention, laying the groundwork for controlled trials with larger sample sizes. Visual and auditory cues may prompt family members to take immediate action to reduce SHS levels. Dynamic graphs of SHS levels may help families make decisions about specific mitigation approaches.

  20. The concept of value stream mapping to reduce of work-time waste as applied the smart construction management

    Science.gov (United States)

    Elizar, Suripin, Wibowo, Mochamad Agung

    2017-11-01

    Delays in construction sites occur due to systematic additions of time waste in various activities that are part of the construction process. Work-time waste is non-adding value activity which used to differentiate between physical construction waste found on site and other waste which occurs during the construction process. The aim of this study is identification using the concept of Value Stream Mapping (VSM) to reduce of work-time waste as applied the smart construction management.VSM analysis is a method of business process improvement. The application of VSM began in the manufacturing community. The research method base on theoretically informed case study and literature review. The data have collected using questionnaire through personal interviews from 383 respondents on construction project in Indonesia. The results show that concept of VSM can identify causes of work-time waste. Base on result of questioners and quantitative approach analysis was obtained 29 variables that influence of work-time waste or non-value-adding activities. Base on three cases of construction project founded that average 14.88% of working time was classified as waste. Finally, the concept of VSM can recommend to identification of systematic for reveal current practices and opportunities for improvement towards global challenges. The concept of value stream mapping can help optimize to reduce work-time waste and improve quality standard of construction management. The concept is also can help manager to make a decision to reduce work-time waste so as to obtain of result in more efficient for performance and sustainable construction project.

  1. Studi Implementasi Lean Six Sigma dengan Pendekatan Value Stream Mapping untuk Mereduksi Idle Time Material pada Gudang Pelat dan Profil

    Directory of Open Access Journals (Sweden)

    Wawan Widiatmoko

    2013-03-01

    Full Text Available Peningkatan volume kegiatan industri maritim di Indonesia menuntut industri perkapalan di daerah Surabaya untuk lebih meningkatkan pelayanan baik berupa bangunan baru maupun reparasi kapal. Berdasarkan hal tersebut galangan harus mampu mengelola proses produksi dengan baik sehingga menghasilkan keuntungan yang maksimum. Salah satunya adalah proses inventory dan transport of materials yang efektif. Tugas akhir bertujuan untuk mengetahui sistem inventori yang diterapkan oleh perusahaan yang dijadikan sampel serta idle time material pelat dan profil yang ada di gudang bahan baku dengan menggunakan metode lean six sigma dengan pendekatan value stream mapping. Dari hasil perhitungan menggunakan diperoleh nilai sigma perhitungan idle time sebesar 0.1976 sehingga perlu dilakukan upaya peningkatan nilai sigma pengadaan material itu sendiri. Berdasarkan hasil analisa penyebab adanya idle time dengan menggunakan RCA diperoleh beberapa faktor yaitu : rendahnya nilai sigma penggunaan material, tidak tercapainya target pengerjaan pada proses fabrikasi, proses pengadaan material yang tidak mempertimbangkan strategi proses pembangunan kapal. Dengan penerapan lean six sigma dengan pendekatan value stream mapping dihasilkan usulan perbaikan proses inventori di perusahaan antara lain : meningkatkan nilai sigma penggunaan material, melakukan strategi pembelian material sesuai strategi pembangunan kapal berdasarkan zona, memperbaiki kerjasama dengan supplier material pelat dan profil. Pembuatan future state mapping mendapatkan usulan perbaikan dengan pembuatan perencanaan pengadaan material dengan mempertimbangkan strategi pembangunan kapal berdasarkan zona pembangunannya. Diperoleh strategi pengadaan material yang dilakukan sebanyak 4 kali order.

  2. In-line near real time monitoring of fluid streams in separation processes for used nuclear fuel - 5146

    International Nuclear Information System (INIS)

    Nee, K.; Nilsson, M.

    2015-01-01

    Applying spectroscopic tools for chemical processes has been intensively studied in various industries owing to its rapid and non-destructive analysis for detecting chemical components and determine physical characteristic in a process stream. The general complexity of separation processes for used nuclear fuel, e.g., chemical speciation, temperature variations, and prominent process security and safety concerns, require a well-secured and robust monitoring system to provide precise information of the process streams at real time without interference. Multivariate analysis accompanied with spectral measurements is a powerful statistic technique that can be used to monitor this complex chemical system. In this work, chemometric models that respond to the chemical components in the fluid samples were calibrated and validated to establish an in-line near real time monitoring system. The models show good prediction accuracy using partial least square regression analysis on the spectral data obtained from UV/Vis/NIR spectroscopies. The models were tested on a solvent extraction process using a single stage centrifugal contactor in our laboratory to determine the performance of an in-line near real time monitoring system. (authors)

  3. Class-D audio amplifiers with negative feedback

    OpenAIRE

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

    2006-01-01

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

  4. A second-order class-D audio amplifier

    OpenAIRE

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

    2011-01-01

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

  5. Multimodal indexing of digital audio-visual documents: A case study for cultural heritage data

    NARCIS (Netherlands)

    Carmichael, J.; Larson, M.; Marlow, J.; Newman, E.; Clough, P.; Oomen, J.; Sav, S.

    2008-01-01

    This paper describes a multimedia multimodal information access sub-system (MIAS) for digital audio-visual documents, typically presented in streaming media format. The system is designed to provide both professional and general users with entry points into video documents that are relevant to their

  6. APPLICATION OF CONTROLLED SOURCE AUDIO MAGNETOTELLURIC (CSAMT AT GEOTHERMAL

    Directory of Open Access Journals (Sweden)

    Susilawati S.

    2017-04-01

    Full Text Available CSAMT or Controlled Source Audio-Magnetotelluric is one of the Geophysics methods to determine the resistivity of rock under earth surface. CSAMT method utilizes artificial stream and injected into the ground, the frequency of artificial sources ranging from 0.1 Hz to 10 kHz, CSAMT data source effect correction is inverted. From the inversion results showed that there is a layer having resistivity values ranged between 2.5 Ω.m – 15 Ω.m, which is interpreted that the layer is clay.

  7. Destabilizing effect of time-dependent oblique magnetic field on magnetic fluids streaming in porous media.

    Science.gov (United States)

    El-Dib, Yusry O; Ghaly, Ahmed Y

    2004-01-01

    The present work studies Kelvin-Helmholtz waves propagating between two magnetic fluids. The system is composed of two semi-infinite magnetic fluids streaming throughout porous media. The system is influenced by an oblique magnetic field. The solution of the linearized equations of motion under the boundary conditions leads to deriving the Mathieu equation governing the interfacial displacement and having complex coefficients. The stability criteria are discussed theoretically and numerically, from which stability diagrams are obtained. Regions of stability and instability are identified for the magnetic fields versus the wavenumber. It is found that the increase of the fluid density ratio, the fluid velocity ratio, the upper viscosity, and the lower porous permeability play a stabilizing role in the stability behavior in the presence of an oscillating vertical magnetic field or in the presence of an oscillating tangential magnetic field. The increase of the fluid viscosity plays a stabilizing role and can be used to retard the destabilizing influence for the vertical magnetic field. Dual roles are observed for the fluid velocity in the stability criteria. It is found that the field frequency plays against the constant part for the magnetic field.

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

    Science.gov (United States)

    Koroghlanian, Carol May

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

  9. Efficiency Optimization in Class-D Audio Amplifiers

    DEFF Research Database (Denmark)

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

    2015-01-01

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

  10. Prioritized Contact Transport Stream

    Science.gov (United States)

    Hunt, Walter Lee, Jr. (Inventor)

    2015-01-01

    A detection process, contact recognition process, classification process, and identification process are applied to raw sensor data to produce an identified contact record set containing one or more identified contact records. A prioritization process is applied to the identified contact record set to assign a contact priority to each contact record in the identified contact record set. Data are removed from the contact records in the identified contact record set based on the contact priorities assigned to those contact records. A first contact stream is produced from the resulting contact records. The first contact stream is streamed in a contact transport stream. The contact transport stream may include and stream additional contact streams. The contact transport stream may be varied dynamically over time based on parameters such as available bandwidth, contact priority, presence/absence of contacts, system state, and configuration parameters.

  11. 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.

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

    Science.gov (United States)

    Kim, Yong-Jin; Chang, Nam-Kee

    2001-01-01

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

  13. Presence and the utility of audio spatialization

    DEFF Research Database (Denmark)

    Bormann, Karsten

    2005-01-01

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

  14. Modified BTC Algorithm for Audio Signal Coding

    Directory of Open Access Journals (Sweden)

    TOMIC, S.

    2016-11-01

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

  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. 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.

  17. Making the Switch to Digital Audio

    Directory of Open Access Journals (Sweden)

    Shannon Gwin Mitchell

    2004-12-01

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

  18. New audio applications of beryllium metal

    International Nuclear Information System (INIS)

    Sato, M.

    1977-01-01

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

  19. Time-aware multi-viewpoint summarization of multilingual social text streams

    NARCIS (Netherlands)

    Ren, Zhaochun; Inel, Oana; Aroyo, Lora; De Rijke, Maarten

    2016-01-01

    A viewpoint is a triple consisting of an entity, a topic related to this entity and sentiment towards this topic. In time-aware multi-viewpoint summarization one monitors viewpoints for a running topic and selects a small set of informative documents. In this paper, we focus on time-aware

  20. Time-based Reconstruction of Free-streaming Data in CBM

    Science.gov (United States)

    Akishina, Valentina; Kisel, Ivan; Vassiliev, Iouri; Zyzak, Maksym

    2018-02-01

    Traditional latency-limited trigger architectures typical for conventional experiments are inapplicable for the CBM experiment. Instead, CBM will ship and collect time-stamped data into a readout buffer in a form of a time-slice of a certain length and deliver it to a large computer farm, where online event reconstruction and selection will be performed. Grouping measurements into physical collisions must be performed in software and requires reconstruction not only in space, but also in time, the so-called 4-dimensional track reconstruction and event building. The tracks, reconstructed with 4D Cellular Automaton track finder, are combined into event-corresponding clusters according to the estimated time in the target position and the errors, obtained with the Kalman Filter method. The reconstructed events are given as inputs to the KF Particle Finder package for short-lived particle reconstruction. The results of time-based reconstruction of simulated collisions in CBM are presented and discussed in details.

  1. Functional dissociation of transient and sustained fMRI BOLD components in human auditory cortex revealed with a streaming paradigm based on interaural time differences.

    Science.gov (United States)

    Schadwinkel, Stefan; Gutschalk, Alexander

    2010-12-01

    A number of physiological studies suggest that feature-selective adaptation is relevant to the pre-processing for auditory streaming, the perceptual separation of overlapping sound sources. Most of these studies are focused on spectral differences between streams, which are considered most important for streaming. However, spatial cues also support streaming, alone or in combination with spectral cues, but physiological studies of spatial cues for streaming remain scarce. Here, we investigate whether the tuning of selective adaptation for interaural time differences (ITD) coincides with the range where streaming perception is observed. FMRI activation that has been shown to adapt depending on the repetition rate was studied with a streaming paradigm where two tones were differently lateralized by ITD. Listeners were presented with five different ΔITD conditions (62.5, 125, 187.5, 343.75, or 687.5 μs) out of an active baseline with no ΔITD during fMRI. The results showed reduced adaptation for conditions with ΔITD ≥ 125 μs, reflected by enhanced sustained BOLD activity. The percentage of streaming perception for these stimuli increased from approximately 20% for ΔITD = 62.5 μs to > 60% for ΔITD = 125 μs. No further sustained BOLD enhancement was observed when the ΔITD was increased beyond ΔITD = 125 μs, whereas the streaming probability continued to increase up to 90% for ΔITD = 687.5 μs. Conversely, the transient BOLD response, at the transition from baseline to ΔITD blocks, increased most prominently as ΔITD was increased from 187.5 to 343.75 μs. These results demonstrate a clear dissociation of transient and sustained components of the BOLD activity in auditory cortex. © 2010 The Authors. European Journal of Neuroscience © 2010 Federation of European Neuroscience Societies and Blackwell Publishing Ltd.

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

    Directory of Open Access Journals (Sweden)

    Maoshen Jia

    2017-12-01

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

  3. ATLAS Live: Collaborative Information Streams

    Energy Technology Data Exchange (ETDEWEB)

    Goldfarb, Steven [Department of Physics, University of Michigan, Ann Arbor, MI 48109 (United States); Collaboration: ATLAS Collaboration

    2011-12-23

    I report on a pilot project launched in 2010 focusing on facilitating communication and information exchange within the ATLAS Collaboration, through the combination of digital signage software and webcasting. The project, called ATLAS Live, implements video streams of information, ranging from detailed detector and data status to educational and outreach material. The content, including text, images, video and audio, is collected, visualised and scheduled using digital signage software. The system is robust and flexible, utilizing scripts to input data from remote sources, such as the CERN Document Server, Indico, or any available URL, and to integrate these sources into professional-quality streams, including text scrolling, transition effects, inter and intra-screen divisibility. Information is published via the encoding and webcasting of standard video streams, viewable on all common platforms, using a web browser or other common video tool. Authorisation is enforced at the level of the streaming and at the web portals, using the CERN SSO system.

  4. ATLAS Live: Collaborative Information Streams

    International Nuclear Information System (INIS)

    Goldfarb, Steven

    2011-01-01

    I report on a pilot project launched in 2010 focusing on facilitating communication and information exchange within the ATLAS Collaboration, through the combination of digital signage software and webcasting. The project, called ATLAS Live, implements video streams of information, ranging from detailed detector and data status to educational and outreach material. The content, including text, images, video and audio, is collected, visualised and scheduled using digital signage software. The system is robust and flexible, utilizing scripts to input data from remote sources, such as the CERN Document Server, Indico, or any available URL, and to integrate these sources into professional-quality streams, including text scrolling, transition effects, inter and intra-screen divisibility. Information is published via the encoding and webcasting of standard video streams, viewable on all common platforms, using a web browser or other common video tool. Authorisation is enforced at the level of the streaming and at the web portals, using the CERN SSO system.

  5. ATLAS Live: Collaborative Information Streams

    CERN Document Server

    Goldfarb, S; The ATLAS collaboration

    2011-01-01

    I report on a pilot project launched in 2010 focusing on facilitating communication and information exchange within the ATLAS Collaboration, through the combination of digital signage software and webcasting. The project, called ATLAS Live, implements video streams of information, ranging from detailed detector and data status to educational and outreach material. The content, including text, images, video and audio, is collected, visualised and scheduled using digital signage software. The system is robust and flexible, utilizing scripts to input data from remote sources, such as the CERN Document Server, Indico, or any available URL, and to integrate these sources into professional-quality streams, including text scrolling, transition effects, inter and intra-screen divisibility. Information is published via the encoding and webcasting of standard video streams, viewable on all common platforms, using a web browser or other common video tool. Authorisation is enforced at the level of the streaming and at th...

  6. Deactivation of hydrophobic catalysts for a hydrogen isotope exchange: Application of the time-on-stream theory

    International Nuclear Information System (INIS)

    Choi, Heui-Joo; Lee, Han Soo; Ahn, Do-Hee; Kim, Jeong-Guk; Kim, Wi-soo; Sohn, SoonHwan

    2005-01-01

    A recycle reactor was built for the purpose of characterizing newly developed hydrophobic catalysts for a hydrogen isotope exchange. The catalytic rate constants of two types of hydrophobic catalysts were measured at a 100% relative humidity. The catalytic rate constants were measured at 60 deg C for 28 days and both the catalysts showed very high initial catalytic rate constants. The measured deactivation profile showed that the catalytic rate constants of both the catalysts were almost identical for 28 days. The deactivation of the catalysts was modelled based upon the time-on-stream theory. The deactivation profiles of the catalysts were estimated by using the model for a period of three years. The results showed that both the catalysts had a good exchange capacity for hydrogen isotopes and they could be applicable to a tritium removal facility that will be built at the Wolsong nuclear power plants in the near future

  7. Streams with Strahler Stream Order

    Data.gov (United States)

    Minnesota Department of Natural Resources — Stream segments with Strahler stream order values assigned. As of 01/08/08 the linework is from the DNR24K stream coverages and will not match the updated...

  8. Assessing Task Migration Impact on Embedded Soft Real-Time Streaming Multimedia Applications

    Directory of Open Access Journals (Sweden)

    Alimonda Andrea

    2008-01-01

    Full Text Available Abstract Multiprocessor systems on chips (MPSoCs are envisioned as the future of embedded platforms such as game-engines, smart-phones and palmtop computers. One of the main challenge preventing the widespread diffusion of these systems is the efficient mapping of multitask multimedia applications on processing elements. Dynamic solutions based on task migration has been recently explored to perform run-time reallocation of task to maximize performance and optimize energy consumption. Even if task migration can provide high flexibility, its overhead must be carefully evaluated when applied to soft real-time applications. In fact, these applications impose deadlines that may be missed during the migration process. In this paper we first present a middleware infrastructure supporting dynamic task allocation for NUMA architectures. Then we perform an extensive characterization of its impact on multimedia soft real-time applications using a software FM Radio benchmark.

  9. Assessing Task Migration Impact on Embedded Soft Real-Time Streaming Multimedia Applications

    Directory of Open Access Journals (Sweden)

    Andrea Acquaviva

    2008-01-01

    Full Text Available Multiprocessor systems on chips (MPSoCs are envisioned as the future of embedded platforms such as game-engines, smart-phones and palmtop computers. One of the main challenge preventing the widespread diffusion of these systems is the efficient mapping of multitask multimedia applications on processing elements. Dynamic solutions based on task migration has been recently explored to perform run-time reallocation of task to maximize performance and optimize energy consumption. Even if task migration can provide high flexibility, its overhead must be carefully evaluated when applied to soft real-time applications. In fact, these applications impose deadlines that may be missed during the migration process. In this paper we first present a middleware infrastructure supporting dynamic task allocation for NUMA architectures. Then we perform an extensive characterization of its impact on multimedia soft real-time applications using a software FM Radio benchmark.

  10. Pilot-Streaming: Design Considerations for a Stream Processing Framework for High-Performance Computing

    OpenAIRE

    Andre Luckow; Peter Kasson; Shantenu Jha

    2016-01-01

    This White Paper (submitted to STREAM 2016) identifies an approach to integrate streaming data with HPC resources. The paper outlines the design of Pilot-Streaming, which extends the concept of Pilot-abstraction to streaming real-time data.

  11. Low-cost guaranteed-throughput communication ring for real-time streaming MPSoCs

    NARCIS (Netherlands)

    Dekens, B.H.J.; Kurtin, Philip Sebastian; Bekooij, Marco Jan Gerrit; Smit, Gerardus Johannes Maria

    2013-01-01

    Connection-oriented guaranteed-throughput mesh-based networks on chip have been proposed as a replacement for buses in real-time embedded multiprocessor systems such as software defined radios. Even with attractive features like throughput and latency guarantees they are not always used because

  12. Image/Time Series Mining Algorithms: Applications to Developmental Biology, Document Processing and Data Streams

    Science.gov (United States)

    Tataw, Oben Moses

    2013-01-01

    Interdisciplinary research in computer science requires the development of computational techniques for practical application in different domains. This usually requires careful integration of different areas of technical expertise. This dissertation presents image and time series analysis algorithms, with practical interdisciplinary applications…

  13. Delivering Instruction via Streaming Media: A Higher Education Perspective.

    Science.gov (United States)

    Mortensen, Mark; Schlieve, Paul; Young, Jon

    2000-01-01

    Describes streaming media, an audio/video presentation that is delivered across a network so that it is viewed while being downloaded onto the user's computer, including a continuous stream of video that can be pre-recorded or live. Discusses its use for nontraditional students in higher education and reports on implementation experiences. (LRW)

  14. An accurate analysis for guaranteed performance of multiprocessor streaming applications

    NARCIS (Netherlands)

    Poplavko, P.

    2008-01-01

    Already for more than a decade, consumer electronic devices have been available for entertainment, educational, or telecommunication tasks based on multimedia streaming applications, i.e., applications that process streams of audio and video samples in digital form. Multimedia capabilities are

  15. 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

  16. 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...

  17. Fusion for Audio-Visual Laughter Detection

    NARCIS (Netherlands)

    Reuderink, B.

    2007-01-01

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

  18. Audio-Visual Classification of Sports Types

    DEFF Research Database (Denmark)

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

    2015-01-01

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

  19. Disentangling P̅ANDA 's time-based data stream

    International Nuclear Information System (INIS)

    Tiemens, M.

    2016-01-01

    To allow the online selection of events, the readout of the P̅ANDA detector will reconstruct particles online. Several algorithms to perform this task in one of the subsystems, the electromagnetic calorimeter, are being developed. The algorithms are discussed, and a simple test case shows that they appear to behave similarly in terms of the ability to reconstruct events, and the time it takes to do this. However, some peculiarities, like the fact that the algorithms show the best performance in what is expected to be the worst-case scenario, still require additional investigation. (paper)

  20. Deciphering relationships between in-stream travel times, nutrient concentrations, and uptake through analysis of hysteretic and non-hysteretic kinetic behavior

    Science.gov (United States)

    Covino, T. P.; Bowden, W. B.; Gooseff, M. N.; Wollheim, W. M.; McGlynn, B. L.; Whittinghill, K. A.; Wlostowski, A. N.; Herstand, M. R.

    2012-12-01

    Understanding the relationship between solute travel time, concentration, and nutrient uptake remains a central question in watershed hydrology and biogeochemistry. Theoretical understanding predicts that nutrient uptake should increase as in-stream solute travel time lengthens and/or as concentration increases; however, results from field-based studies have been contradictory. We used a newly developed approach, Tracer Additions for Spiraling Curve Characterization (TASCC), to investigate relationships between solute travel time, nutrient concentration, and nutrient uptake across a range of stream types. This approach allows us to quantify in-stream nutrient uptake across a range of travel times and nutrient concentrations using single instantaneous injections (slugs) of conservative and non-conservative tracers. In some systems we observed counter-clockwise hysteresis loops in the relationship between nutrient uptake and concentration. Greater nutrient uptake on the falling limb of tracer breakthrough curves indicates stronger uptake for a given concentration at longer travel times. However, in other systems we did not observe hysteresis in these relationships. Lack of hysteresis indicates that nutrient uptake kinetics were not influenced by travel time travel time. Here we investigate the potential roles of travel time and in-stream flowpaths that could be responsible for hysteretic behavior.

  1. Timing of revenue streams from newly recruited faculty: implications for faculty retention.

    Science.gov (United States)

    Joiner, Keith A; Hiteman, Sarah; Wormsley, Steven; St Germain, Patricia

    2007-12-01

    To determine the timing and magnitude of revenues generated by newly recruited faculty, to facilitate configuration of recruitment packages appropriately matched to expected financial returns. The aggregate of all positive cash flows to central college of medicine administration -- from research, clinical care, tuition, philanthropy, and royalties and patents, from all faculty newly recruited to the University of Arizona College of Medicine between 1998 and 2004 -- was quantified using the net present value (npv) methodology, which incorporates the time value of money. Tenure-track faculty and, in particular, those with laboratory research programs, generated the highest positive central cash flows. The npv for positive cash flows (npv[+]) during 6 and 10 years for newly recruited assistant professors with laboratory research programs were $118,600 and $255,400, respectively, and, for professors with laboratory research programs, $172,600 and $298,000, respectively (associate professors were not analyzed because of limited numbers). Faculty whose appointments at the University of Arizona College of Medicine exceeded 15 years in duration were the most productive in central revenue generation, far in excess of their numbers proportionate to the total. The results emphasize the critical importance of faculty retention, because even those newly recruited faculty who are most successful in central revenue generation (tenure track with laboratory research programs) must be retained for periods well in excess of 10 years to recoup the initial central investment required for their recruitment.

  2. Audio Satellites: Overhearing Everyday Life

    DEFF Research Database (Denmark)

    Kirkegaard, Jonas Rasmussen; Breinbjerg, M.; Højlund, M. K.

    2017-01-01

    around or displaced arbitrarily in a given landscape. In the web browser, the different sound streams from the individual satellites can be mixed together to form a cooperative soundscape. The project thus allows people to tune into and explore the overheard soundscape of everyday life in a collaborative...

  3. 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.

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

    Energy Technology Data Exchange (ETDEWEB)

    NONE

    2000-03-01

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

  5. Real-time video streaming of sonographic clips using domestic internet networks and free videoconferencing software.

    Science.gov (United States)

    Liteplo, Andrew S; Noble, Vicki E; Attwood, Ben H C

    2011-11-01

    As the use of point-of-care sonography spreads, so too does the need for remote expert over-reading via telesonogrpahy. We sought to assess the feasibility of using familiar, widespread, and cost-effective existent technology to allow remote over-reading of sonograms in real time and to compare 4 different methods of transmission and communication for both the feasibility of transmission and image quality. Sonographic video clips were transmitted using 2 different connections (WiFi and 3G) and via 2 different videoconferencing modalities (iChat [Apple Inc, Cupertino, CA] and Skype [Skype Software Sàrl, Luxembourg]), for a total of 4 different permutations. The clips were received at a remote location and recorded and then scored by expert reviewers for image quality, resolution, and detail. Wireless transmission of sonographic clips was feasible in all cases when WiFi was used and when Skype was used over a 3G connection. Images transmitted via a WiFi connection were statistically superior to those transmitted via 3G in all parameters of quality (average P = .031), and those sent by iChat were superior to those sent by Skype but not statistically so (average P = .057). Wireless transmission of sonographic video clips using inexpensive hardware, free videoconferencing software, and domestic Internet networks is feasible with retention of image quality sufficient for interpretation. WiFi transmission results in greater image quality than transmission by a 3G network.

  6. Fusion of product and process data: Batch-mode and real-time streaming

    Energy Technology Data Exchange (ETDEWEB)

    Vincent De Sapio; Spike Leonard

    1999-12-01

    In today's DP product realization enterprise it is imperative to reduce the design-to-fabrication cycle time and cost while improving the quality of DP parts (reducing defects). Much of this challenge resides in the inherent gap between the product and process worlds. The lack of seamless, bi-directional flow of information prevents true concurrency in the product realization world. This report addresses a framework for product-process data fusion to help achieve next generation product realization. A fundamental objective is to create an open environment for multichannel observation of process date, and subsequent mapping of that data onto product geometry. In addition to the sensor-based observation of manufacturing processes, model-based process data provides an important complement to empirically acquired data. Two basic groups of manufacturing models are process physics, and machine kinematics and dynamics. Process physics addresses analytical models that describe the physical phenomena of the process itself. Machine kinematic and dynamic models address the mechanical behavior of the processing equipment. As a secondary objective, an attempt has been made in this report to address part of the model-based realm through the development of an open object-oriented library and toolkit for machine kinematics and dynamics. Ultimately, it is desirable to integrate design definition, with all types of process data; both sensor-based and model-based. Collectively, the goal is to allow all disciplines within the product realization enterprise to have a centralized medium for the fusion of product and process data.

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

    International Nuclear Information System (INIS)

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

    2009-01-01

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

  8. Reducing Door-to-Needle Times using Toyota’s Lean Manufacturing Principles and Value Stream Analysis

    Science.gov (United States)

    Ford, Andria L.; Williams, Jennifer A.; Spencer, Mary; McCammon, Craig; Khoury, Naim; Sampson, Tomoko; Panagos, Peter; Lee, Jin-Moo

    2012-01-01

    Background Earlier tPA treatment for acute ischemic stroke increases efficacy, prompting national efforts to reduce door-to-needle times (DNTs). We utilized lean process improvement methodology to develop a streamlined IV tPA protocol. Methods In early 2011, a multi-disciplinary team analyzed the steps required to treat acute ischemic stroke patients with IV tPA, utilizing value stream analysis (VSA). We directly compared the tPA-treated patients in the “pre-VSA” epoch to the “post-VSA” epoch with regard to baseline characteristics, protocol metrics, and clinical outcomes. Results The VSA revealed several tPA protocol inefficiencies: routing of patients to room, then to CT, then back to room; serial processing of work flow; and delays in waiting for lab results. On 3/1/2011, a new protocol incorporated changes to minimize delays: routing patients directly to head CT prior to patient room, utilizing parallel process work-flow, and implementing point-of-care labs. In the pre-and post-VSA epochs, 132 and 87 patients were treated with IV tPA, respectively. Compared to pre-VSA, DNTs and percent of patients treated ≤60 minutes from hospital arrival were improved in the post-VSA epoch: 60 min vs. 39 min (pLean process improvement methodology can expedite time-dependent stroke care, without compromising safety. PMID:23138440

  9. Investigation of Relationship Between Hydrologic Processes of Precipitation, Evaporation and Stream Flow Using Linear Time Series Models (Case study: Western Basins of Lake Urmia

    Directory of Open Access Journals (Sweden)

    M. Moravej

    2016-02-01

    Full Text Available Introduction: Studying the hydrological cycle, especially in large scales such as water catchments, is difficult and complicated despite the fact that the numbers of hydrological components are limited. This complexity rises from complex interactions between hydrological components and environment. Recognition, determination and modeling of all interactive processes are needed to address this issue, but it's not feasible for dealing with practical engineering problems. So, it is more convenient to consider hydrological components as stochastic phenomenon, and use stochastic models for modeling them. Stochastic simulation of time series models related to water resources, particularly hydrologic time series, have been widely used in recent decades in order to solve issues pertaining planning and management of water resource systems. In this study time series models fitted to the precipitation, evaporation and stream flow series separately and the relationships between stream flow and precipitation processes are investigated. In fact, the three mentioned processes should be modeled in parallel to each other in order to acquire a comprehensive vision of hydrological conditions in the region. Moreover, the relationship between the hydrologic processes has been mostly studied with respect to their trends. It is desirable to investigate the relationship between trends of hydrological processes and climate change, while the relationship of the models has not been taken into consideration. The main objective of this study is to investigate the relationship between hydrological processes and their effects on each other and the selected models. Material and Method: In the current study, the four sub-basins of Lake Urmia Basin namely Zolachay (A, Nazloochay (B, Shahrchay (C and Barandoozchay (D were considered. Precipitation, evaporation and stream flow time series were modeled by linear time series. Fundamental assumptions of time series analysis namely

  10. High-Fidelity Piezoelectric Audio Device

    Science.gov (United States)

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

    2003-01-01

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

  11. 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

  12. Musical Audio Synthesis Using Autoencoding Neural Nets

    OpenAIRE

    Sarroff, Andy; Casey, Michael A.

    2014-01-01

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

  13. The Ocean Observatories Initiative: Unprecedented access to real-time data streaming from the Cabled Array through OOI Cyberinfrastructure

    Science.gov (United States)

    Knuth, F.; Vardaro, M.; Belabbassi, L.; Smith, M. J.; Garzio, L. M.; Crowley, M. F.; Kerfoot, J.; Kawka, O. E.

    2016-02-01

    The National Science Foundation's Ocean Observatories Initiative (OOI), is a broad-scale, multidisciplinary facility that will transform oceanographic research by providing users with unprecedented access to long-term datasets from a variety of deployed physical, chemical, biological, and geological sensors. The Cabled Array component of the OOI, installed and operated by the University of Washington, is located on the Juan de Fuca tectonic plate off the coast of Oregon. It is a unique network of >100 cabled instruments and instrumented moorings transmitting data to shore in real-time via fiber optic technology. Instruments now installed include HD video and digital still cameras, mass spectrometers, a resistivity-temperature probe inside the orifice of a high-temperature hydrothermal vent, upward-looking ADCP's, pH and pC02 sensors, Horizontal Electrometer Pressure Inverted Echosounders and many others. Here, we present the technical aspects of data streaming from the Cabled Array through the OOI Cyberinfrastructure. We illustrate the types of instruments and data products available, data volume and density, processing levels and algorithms used, data delivery methods, file formats and access methods through the graphical user interface. Our goal is to facilitate the use and access to these unprecedented, co-registered oceanographic datasets. We encourage researchers to collaborate through the use of these simultaneous, interdisciplinary measurements, in the exploration of short-lived events (tectonic, volcanic, biological, severe storms), as well as long-term trends in ocean systems (circulation patterns, climate change, ocean acidity, ecosystem shifts).

  14. Modelling mean transit time of stream base flow during tropical cyclone rainstorm in a steep relief forested catchment

    Science.gov (United States)

    Lee, Jun-Yi; Huang, -Chuan, Jr.

    2017-04-01

    Mean transit time (MTT) is one of the of fundamental catchment descriptors to advance understanding on hydrological, ecological, and biogeochemical processes and improve water resources management. However, there were few documented the base flow partitioning (BFP) and mean transit time within a mountainous catchment in typhoon alley. We used a unique data set of 18O isotope and conductivity composition of rainfall (136 mm to 778 mm) and streamflow water samples collected for 14 tropical cyclone events (during 2011 to 2015) in a steep relief forested catchment (Pinglin, in northern Taiwan). A lumped hydrological model, HBV, considering dispersion model transit time distribution was used to estimate total flow, base flow, and MTT of stream base flow. Linear regression between MTT and hydrometric (precipitation intensity and antecedent precipitation index) variables were used to explore controls on MTT variation. Results revealed that both the simulation performance of total flow and base flow were satisfactory, and the Nash-Sutcliffe model efficiency coefficient of total flow and base flow was 0.848 and 0.732, respectively. The event magnitude increased with the decrease of estimated MTTs. Meanwhile, the estimated MTTs varied 4-21 days with the increase of BFP between 63-92%. The negative correlation between event magnitude and MTT and BFP showed the forcing controls the MTT and BFP. Besides, a negative relationship between MTT and the antecedent precipitation index was also found. In other words, wetter antecedent moisture content more rapidly active the fast flow paths. This approach is well suited for constraining process-based modeling in a range of high precipitation intensity and steep relief forested environments.

  15. BigSR: an empirical study of real-time expressive RDF stream reasoning on modern Big Data platforms

    OpenAIRE

    Ren, Xiangnan; Curé, Olivier; Naacke, Hubert; Xiao, Guohui

    2018-01-01

    The trade-off between language expressiveness and system scalability (E&S) is a well-known problem in RDF stream reasoning. Higher expressiveness supports more complex reasoning logic, however, it may also hinder system scalability. Current research mainly focuses on logical frameworks suitable for stream reasoning as well as the implementation and the evaluation of prototype systems. These systems are normally developed in a centralized setting which suffer from inherent limited scalability,...

  16. Interaction of on-site and near real time measured turbidity and enzyme activity in stream water.

    Science.gov (United States)

    Stadler, Philipp; Farnleitner, Andreas H.; Zessner, Matthias

    2013-04-01

    On-site and on-line systems that provide an integrated surveillance of physicochemical and microbiological parameters gain significance in water quality monitoring. Particular relating to diffuse pollution from agricultural areas and use-orientated protection of waters the detection of faecal pollution is a fundamental part. For the near real time and on-site detection of microbiological faecal pollution of water, the beta-D- Glucuronidase (GLUC) enzymatic activity has been suggested as a surrogate parameter. Due to possible short measure intervals of three hours, this method has high potential as a water quality monitoring tool. While cultivation based standard determination takes more than one working day (Cabral 2010) the potential advantage of detecting the GLUC activity is the high temporal measuring resolution. Yet, there is still a big gap of knowledge on the sensitivity and specificity concerning the faecal indication capacity of GLUC in relation to standard assays (Cabral 2010). Interference effects of physicochemical parameters on the enzymatic activity respectively fluorescence have been discussed (Molina-Munoz et al. 2007; Tryland and Fiksdal 1998, Biswal et al. 2003). Results from a monitoring of a rivulet in an agricultural catchment in Lower Austria (HOAL - Hydrological Open Air Laboratory) are presented here. The HOAL offers technical resources that allow measurements at high temporal and spatial resolution and to apply various hydrological methods in one catchment. Two automated enzymatic measuring devices (Coliguard, mbOnline, Austria) and physicochemical in-stream measurements are used, as well as in-stream spectroscopy (spectrolyser, s::can, Austria). Accuracy of both enzymatic measuring devices is compared through diverse hydrological and seasonal conditions. Reference analyses by cultivation based determination were performed. Data from Coliguard devices is combined with physicochemical and spectroscopy data to gain information about the

  17. Analysis of musical expression in audio signals

    Science.gov (United States)

    Dixon, Simon

    2003-01-01

    In western art music, composers communicate their work to performers via a standard notation which specificies the musical pitches and relative timings of notes. This notation may also include some higher level information such as variations in the dynamics, tempo and timing. Famous performers are characterised by their expressive interpretation, the ability to convey structural and emotive information within the given framework. The majority of work on audio content analysis focusses on retrieving score-level information; this paper reports on the extraction of parameters describing the performance, a task which requires a much higher degree of accuracy. Two systems are presented: BeatRoot, an off-line beat tracking system which finds the times of musical beats and tracks changes in tempo throughout a performance, and the Performance Worm, a system which provides a real-time visualisation of the two most important expressive dimensions, tempo and dynamics. Both of these systems are being used to process data for a large-scale study of musical expression in classical and romantic piano performance, which uses artificial intelligence (machine learning) techniques to discover fundamental patterns or principles governing expressive performance.

  18. Stream Crossings

    Data.gov (United States)

    Vermont Center for Geographic Information — Physical measurements and attributes of stream crossing structures and adjacent stream reaches which are used to provide a relative rating of aquatic organism...

  19. Audio-Visual Speaker Diarization Based on Spatiotemporal Bayesian Fusion.

    Science.gov (United States)

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

    2018-05-01

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

  20. Stream/Bounce Event Perception Reveals a Temporal Limit of Motion Correspondence Based on Surface Feature over Space and Time

    Directory of Open Access Journals (Sweden)

    Yousuke Kawachi

    2011-06-01

    Full Text Available We examined how stream/bounce event perception is affected by motion correspondence based on the surface features of moving objects passing behind an occlusion. In the stream/bounce display two identical objects moving across each other in a two-dimensional display can be perceived as either streaming through or bouncing off each other at coincidence. Here, surface features such as colour (Experiments 1 and 2 or luminance (Experiment 3 were switched between the two objects at coincidence. The moment of coincidence was invisible to observers due to an occluder. Additionally, the presentation of the moving objects was manipulated in duration after the feature switch at coincidence. The results revealed that a postcoincidence duration of approximately 200 ms was required for the visual system to stabilize judgments of stream/bounce events by determining motion correspondence between the objects across the occlusion on the basis of the surface feature. The critical duration was similar across motion speeds of objects and types of surface features. Moreover, controls (Experiments 4a–4c showed that cognitive bias based on feature (colour/luminance congruency across the occlusion could not fully account for the effects of surface features on the stream/bounce judgments. We discuss the roles of motion correspondence, visual feature processing, and attentive tracking in the stream/bounce judgments.

  1. ATLAS Live: Collaborative Information Streams

    CERN Document Server

    Goldfarb, S; The ATLAS collaboration

    2010-01-01

    I report on a pilot project launched in 2010 focusing on facilitating communication and information exchange within the ATLAS Collaboration, through the combination of digital signage software and webcasting. The project, called ATLAS Live, implements video streams of information, ranging from detailed detector and data status to educational and outreach material. The content, including text, images, video and audio, is collected, visualised and scheduled using the SCALA digital signage software system. The system is robust and flexible, allowing for the usage of scripts to input data from remote sources, such as the CERN Document Server, Indico, or any available URL, and to integrate these sources into professional-quality streams, including text scrolling, transition effects, inter and intrascreen divisibility. The video is made available to the collaboration or public through the encoding and webcasting of standard video streams, viewable on all common platforms, using a web browser or other common video t...

  2. Routing Optimization of AVB Streams in TSN Networks

    DEFF Research Database (Denmark)

    Laursen, Sune Mølgaard; Pop, Paul; Steiner, Wilfried

    2016-01-01

    In this paper we are interested in safety-critical real-time applications implemented on distributed architectures using the Time-Sensitive Networking (TSN) standard. The ongoing standardization of TSN is an IEEE effort to bring deterministic real-time capabilities into the IEEE 802.1 Ethernet...... standard supporting safety-critical systems and guaranteed Quality-of-Service. TSN will support Time-Triggered (TT) communication based on schedule tables, Audio-Video-Bridging (AVB) streams with bounded end-to-end latency as well as Best-Effort messages. We consider that we know the topology...... Procedure (GRASP)-based heuristic for this routing optimization problem. The proposed approaches has been evaluated using several test cases....

  3. Audio-Visual Tibetan Speech Recognition Based on a Deep Dynamic Bayesian Network for Natural Human Robot Interaction

    Directory of Open Access Journals (Sweden)

    Yue Zhao

    2012-12-01

    Full Text Available Audio-visual speech recognition is a natural and robust approach to improving human-robot interaction in noisy environments. Although multi-stream Dynamic Bayesian Network and coupled HMM are widely used for audio-visual speech recognition, they fail to learn the shared features between modalities and ignore the dependency of features among the frames within each discrete state. In this paper, we propose a Deep Dynamic Bayesian Network (DDBN to perform unsupervised extraction of spatial-temporal multimodal features from Tibetan audio-visual speech data and build an accurate audio-visual speech recognition model under a no frame-independency assumption. The experiment results on Tibetan speech data from some real-world environments showed the proposed DDBN outperforms the state-of-art methods in word recognition accuracy.

  4. Akamai Streaming

    OpenAIRE

    ECT Team, Purdue

    2007-01-01

    Akamai offers world-class streaming media services that enable Internet content providers and enterprises to succeed in today's Web-centric marketplace. They deliver live event Webcasts (complete with video production, encoding, and signal acquisition services), streaming media on demand, 24/7 Webcasts and a variety of streaming application services based upon their EdgeAdvantage.

  5. Off-Stream Watering Systems and Partial Barriers as a Strategy to Maximize Cattle Production and Minimize Time Spent in the Riparian Area.

    Science.gov (United States)

    Rawluk, Ashley A; Crow, Gary; Legesse, Getahun; Veira, Douglas M; Bullock, Paul R; González, Luciano A; Dubois, Melanie; Ominski, Kim H

    2014-10-29

    A study was conducted in 2009 at two locations in Manitoba (Killarney and Souris), Canada to determine the impact of off-stream waterers (OSW) with or without natural barriers on (i) amount of time cattle spent in the 10 m buffer created within the riparian area, referred to as the riparian polygon (RP), (ii) watering location (OSW or stream), and (iii) animal performance measured as weight gain. This study was divided into three 28-day periods over the grazing season. At each location, the pasture-which ranged from 21.0 ha to 39.2 ha in size-was divided into three treatments: no OSW nor barriers (1CONT), OSW with barriers along the stream bank to deter cattle from watering at the stream (2BARR), and OSW without barriers (3NOBARR). Cattle in 2BARR spent less time in the RP in Periods 1 (p = 0.0002), 2 (p = 0.1116), and 3 (p natural barriers on deterring cattle from the riparian area between periods and locations may be partly attributable to the environmental conditions present during this field trial as well as difference in pasture size and the ability of the established barriers to deter cattle from using the stream as a water source. Treatment had no significant effect (p > 0.05) on cow and calf weights averaged over the summer period. These results indicate that the presence of an OSW does not create significant differences in animal performance when used in extensive pasture scenarios such as those studied within the present study. Whereas the barriers did not consistently discourage watering at the stream, the results provide some indication of the efficacy of the OSW as well as the natural barriers on deterring cattle from the riparian area.

  6. Evaluation of Stream Mining Classifiers for Real-Time Clinical Decision Support System: A Case Study of Blood Glucose Prediction in Diabetes Therapy

    Directory of Open Access Journals (Sweden)

    Simon Fong

    2013-01-01

    Full Text Available Earlier on, a conceptual design on the real-time clinical decision support system (rt-CDSS with data stream mining was proposed and published. The new system is introduced that can analyze medical data streams and can make real-time prediction. This system is based on a stream mining algorithm called VFDT. The VFDT is extended with the capability of using pointers to allow the decision tree to remember the mapping relationship between leaf nodes and the history records. In this paper, which is a sequel to the rt-CDSS design, several popular machine learning algorithms are investigated for their suitability to be a candidate in the implementation of classifier at the rt-CDSS. A classifier essentially needs to accurately map the events inputted to the system into one of the several predefined classes of assessments, such that the rt-CDSS can follow up with the prescribed remedies being recommended to the clinicians. For a real-time system like rt-CDSS, the major technological challenges lie in the capability of the classifier to process, analyze and classify the dynamic input data, quickly and upmost reliably. An experimental comparison is conducted. This paper contributes to the insight of choosing and embedding a stream mining classifier into rt-CDSS with a case study of diabetes therapy.

  7. Real-time dissemination of air quality information using data streams and Web technologies: linking air quality to health risks in urban areas.

    Science.gov (United States)

    Davila, Silvije; Ilić, Jadranka Pečar; Bešlić, Ivan

    2015-06-01

    This article presents a new, original application of modern information and communication technology to provide effective real-time dissemination of air quality information and related health risks to the general public. Our on-line subsystem for urban real-time air quality monitoring is a crucial component of a more comprehensive integrated information system, which has been developed by the Institute for Medical Research and Occupational Health. It relies on a StreamInsight data stream management system and service-oriented architecture to process data streamed from seven monitoring stations across Zagreb. Parameters that are monitored include gases (NO, NO2, CO, O3, H2S, SO2, benzene, NH3), particulate matter (PM10 and PM2.5), and meteorological data (wind speed and direction, temperature and pressure). Streamed data are processed in real-time using complex continuous queries. They first go through automated validation, then hourly air quality index is calculated for every station, and a report sent to the Croatian Environment Agency. If the parameter values exceed the corresponding regulation limits for three consecutive hours, the web service generates an alert for population groups at risk. Coupled with the Common Air Quality Index model, our web application brings air pollution information closer to the general population and raises awareness about environmental and health issues. Soon we intend to expand the service to a mobile application that is being developed.

  8. 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.

  9. 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.

  10. EVALUASI KEPUASAN PENGGUNA TERHADAP APLIKASI AUDIO BOOKS

    Directory of Open Access Journals (Sweden)

    Raditya Maulana Anuraga

    2017-02-01

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

  11. 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.

  12. Audio production principles practical studio applications

    CERN Document Server

    Elmosnino, Stephane

    2018-01-01

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

  13. Audio Technology and Mobile Human Computer Interaction

    DEFF Research Database (Denmark)

    Chamberlain, Alan; Bødker, Mads; Hazzard, Adrian

    2017-01-01

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

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

    DEFF Research Database (Denmark)

    Turchet, Luca; Serafin, Stefania; Dimitrov, Smilen

    2010-01-01

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

  15. 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; ·     �...

  16. Audio pacemaker : Walking, talking indigenous knowledge

    CSIR Research Space (South Africa)

    Bidwell, NJ

    2012-10-01

    Full Text Available stream_source_info Bidwell1_2012_ABSTRACT ONLY.pdf.txt stream_content_type text/plain stream_size 1422 Content-Encoding ISO-8859-1 stream_name Bidwell1_2012_ABSTRACT ONLY.pdf.txt Content-Type text/plain; charset=ISO-8859-1...

  17. Off-Stream Watering Systems and Partial Barriers as a Strategy to Maximize Cattle Production and Minimize Time Spent in the Riparian Area

    Directory of Open Access Journals (Sweden)

    Ashley A. Rawluk

    2014-10-01

    Full Text Available A study was conducted in 2009 at two locations in Manitoba (Killarney and Souris, Canada to determine the impact of off-stream waterers (OSW with or without natural barriers on (i amount of time cattle spent in the 10 m buffer created within the riparian area, referred to as the riparian polygon (RP, (ii watering location (OSW or stream, and (iii animal performance measured as weight gain. This study was divided into three 28-day periods over the grazing season. At each location, the pasture—which ranged from 21.0 ha to 39.2 ha in size—was divided into three treatments: no OSW nor barriers (1CONT, OSW with barriers along the stream bank to deter cattle from watering at the stream (2BARR, and OSW without barriers (3NOBARR. Cattle in 2BARR spent less time in the RP in Periods 1 (p = 0.0002, 2 (p = 0.1116, and 3 (p < 0.0001 at the Killarney site compared to cattle in 3NOBARR at the same site. Cattle in 2BARR at the Souris site spent more time in the RP in Period 1 (p < 0.0001 and less time in Period 2 (p = 0.0002 compared to cattle in 3NOBARR. Cattle did use the OSW, but not exclusively, as watering at the stream was still observed. The observed inconsistency in the effectiveness of the natural barriers on deterring cattle from the riparian area between periods and locations may be partly attributable to the environmental conditions present during this field trial as well as difference in pasture size and the ability of the established barriers to deter cattle from using the stream as a water source. Treatment had no significant effect (p > 0.05 on cow and calf weights averaged over the summer period. These results indicate that the presence of an OSW does not create significant differences in animal performance when used in extensive pasture scenarios such as those studied within the present study. Whereas the barriers did not consistently discourage watering at the stream, the results provide some indication of the efficacy of the OSW as well

  18. Convergent Time-Varying Regression Models for Data Streams: Tracking Concept Drift by the Recursive Parzen-Based Generalized Regression Neural Networks.

    Science.gov (United States)

    Duda, Piotr; Jaworski, Maciej; Rutkowski, Leszek

    2018-03-01

    One of the greatest challenges in data mining is related to processing and analysis of massive data streams. Contrary to traditional static data mining problems, data streams require that each element is processed only once, the amount of allocated memory is constant and the models incorporate changes of investigated streams. A vast majority of available methods have been developed for data stream classification and only a few of them attempted to solve regression problems, using various heuristic approaches. In this paper, we develop mathematically justified regression models working in a time-varying environment. More specifically, we study incremental versions of generalized regression neural networks, called IGRNNs, and we prove their tracking properties - weak (in probability) and strong (with probability one) convergence assuming various concept drift scenarios. First, we present the IGRNNs, based on the Parzen kernels, for modeling stationary systems under nonstationary noise. Next, we extend our approach to modeling time-varying systems under nonstationary noise. We present several types of concept drifts to be handled by our approach in such a way that weak and strong convergence holds under certain conditions. Finally, in the series of simulations, we compare our method with commonly used heuristic approaches, based on forgetting mechanism or sliding windows, to deal with concept drift. Finally, we apply our concept in a real life scenario solving the problem of currency exchange rates prediction.

  19. The Effect of Beaver Activity on the Ammonium Uptake and Water Residence Time Characteristics of a Third-Order Stream Reach

    Science.gov (United States)

    Briggs, M.; Gooseff, M. N.; Wollheim, W. M.; Peterson, B. J.; Morkeski, K.

    2009-12-01

    Increasing beaver populations within low gradient basins in the northeastern United States are fundamentally changing the way water and dissolved nutrients are exported through these stream networks to the coast. Beaver dams can increase water residence time and contact with organic material, promote anoxic conditions and enhance both surface and hyporheic transient storage; all of these may have an impact on biogeochemical reactivity and nutrient retention. To quantitatively assess some of these effects we co-injected NaCl and NH4+ into the same 3rd-order stream reach in Massachusetts, USA under pre- and post-dam conditions. These experiments were done at similar discharge rates to isolate the impacts of a large natural beaver dam (7 m X 1.3 m) on the low-gradient (0.002) system where variable discharge also imparts a strong control on residence time. During the post-dam experiment there was an estimated 2300 m3 of water impounded behind the structure, which influenced more than 300 m of the 650 m stream reach. Our results showed that median transport time through the reach increased by 160% after dam construction. Additionally the tracer tailing time normalized to the corresponding median transport time increased from 1.08 to 1.51, indicating a pronounced tailing of the tracer signal in the post-dam condition. Data collected within the beaver pond just upstream of the dam indicated poor mixing and the presence of preferential flow paths through the generally stagnant zone. The uptake length (Sw) for NH4+ was 1250 m under the pre-dam condition, and may have changed for the post-dam reach in part because of the observed changes in residence time. As beaver population growth continues within these basins the consequences may be a smoothing of the outlet hydrograph and increased nutrient and organic matter removal and storage along the stream network.

  20. The Success of Free to Play Games and Possibilities of Audio Monetization

    OpenAIRE

    Hahl, Kalle

    2014-01-01

    Video games are a huge business – nearly four times greater than film and music business combined. Free to play is the fastest growing category in video gaming. Game audio is part of the development of every game having a direct correlation between the growth of gaming industry and the growth of gaming audio industry. Games have inherently different goals for the players and the developers. Players are consumers seeking for entertainment. Developers are content producers trying to moneti...

  1. Estimated fecal coliform bacteria concentrations using near real-time continuous water-quality and streamflow data from five stream sites in Chester County, Pennsylvania, 2007–16

    Science.gov (United States)

    Senior, Lisa A.

    2017-09-15

    Several streams used for recreational activities, such as fishing, swimming, and boating, in Chester County, Pennsylvania, are known to have periodic elevated concentrations of fecal coliform bacteria, a type of bacteria used to indicate the potential presence of fecally related pathogens that may pose health risks to humans exposed through water contact. The availability of near real-time continuous stream discharge, turbidity, and other water-quality data for some streams in the county presents an opportunity to use surrogates to estimate near real-time concentrations of fecal coliform (FC) bacteria and thus provide some information about associated potential health risks during recreational use of streams.The U.S. Geological Survey (USGS), in cooperation with the Chester County Health Department (CCHD) and the Chester County Water Resources Authority (CCWRA), has collected discrete stream samples for analysis of FC concentrations during March–October annually at or near five gaging stations where near real-time continuous data on stream discharge, turbidity, and water temperature have been collected since 2007 (or since 2012 at 2 of the 5 stations). In 2014, the USGS, in cooperation with the CCWRA and CCHD, began to develop regression equations to estimate FC concentrations using available near real-time continuous data. Regression equations included possible explanatory variables of stream discharge, turbidity, water temperature, and seasonal factors calculated using Julian Day with base-10 logarithmic (log) transformations of selected variables.The regression equations were developed using the data from 2007 to 2015 (101–106 discrete bacteria samples per site) for three gaging stations on Brandywine Creek (West Branch Brandywine Creek at Modena, East Branch Brandywine Creek below Downingtown, and Brandywine Creek at Chadds Ford) and from 2012 to 2015 (37–38 discrete bacteria samples per site) for one station each on French Creek near Phoenixville and

  2. Analysis and Implementation of Gossip-Based P2P Streaming with Distributed Incentive Mechanisms for Peer Cooperation

    Directory of Open Access Journals (Sweden)

    Sachin Agarwal

    2007-10-01

    Full Text Available Peer-to-peer (P2P systems are becoming a popular means of streaming audio and video content but they are prone to bandwidth starvation if selfish peers do not contribute bandwidth to other peers. We prove that an incentive mechanism can be created for a live streaming P2P protocol while preserving the asymptotic properties of randomized gossip-based streaming. In order to show the utility of our result, we adapt a distributed incentive scheme from P2P file storage literature to the live streaming scenario. We provide simulation results that confirm the ability to achieve a constant download rate (in time, per peer that is needed for streaming applications on peers. The incentive scheme fairly differentiates peers' download rates according to the amount of useful bandwidth they contribute back to the P2P system, thus creating a powerful quality-of-service incentive for peers to contribute bandwidth to other peers. We propose a functional architecture and protocol format for a gossip-based streaming system with incentive mechanisms, and present evaluation data from a real implementation of a P2P streaming application.

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

    DEFF Research Database (Denmark)

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

    2009-01-01

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

  4. Fall Detection Using Smartphone Audio Features.

    Science.gov (United States)

    Cheffena, Michael

    2016-07-01

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

  5. Audio Description as a Pedagogical Tool

    Directory of Open Access Journals (Sweden)

    Georgina Kleege

    2015-05-01

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

  6. Stream systems.

    Science.gov (United States)

    Jack E. Williams; Gordon H. Reeves

    2006-01-01

    Restored, high-quality streams provide innumerable benefits to society. In the Pacific Northwest, high-quality stream habitat often is associated with an abundance of salmonid fishes such as chinook salmon (Oncorhynchus tshawytscha), coho salmon (O. kisutch), and steelhead (O. mykiss). Many other native...

  7. The time-course of activation in the dorsal and ventral visual streams during landmark cueing and perceptual discrimination tasks.

    Science.gov (United States)

    Lambert, Anthony J; Wootton, Adrienne

    2017-08-01

    Different patterns of high density EEG activity were elicited by the same peripheral stimuli, in the context of Landmark Cueing and Perceptual Discrimination tasks. The C1 component of the visual event-related potential (ERP) at parietal - occipital electrode sites was larger in the Landmark Cueing task, and source localisation suggested greater activation in the superior parietal lobule (SPL) in this task, compared to the Perceptual Discrimination task, indicating stronger early recruitment of the dorsal visual stream. In the Perceptual Discrimination task, source localisation suggested widespread activation of the inferior temporal gyrus (ITG) and fusiform gyrus (FFG), structures associated with the ventral visual stream, during the early phase of the P1 ERP component. Moreover, during a later epoch (171-270ms after stimulus onset) increased temporal-occipital negativity, and stronger recruitment of ITG and FFG were observed in the Perceptual Discrimination task. These findings illuminate the contrasting functions of the dorsal and ventral visual streams, to support rapid shifts of attention in response to contextual landmarks, and conscious discrimination, respectively. Copyright © 2017 Elsevier Ltd. All rights reserved.

  8. Implications of 36Cl exposure ages from Skye, northwest Scotland for the timing of ice stream deglaciation and deglacial ice dynamics

    Science.gov (United States)

    Small, David; Rinterknecht, Vincent; Austin, William E. N.; Bates, Richard; Benn, Douglas I.; Scourse, James D.; Bourlès, Didier L.; Hibbert, Fiona D.

    2016-10-01

    Geochronological constraints on the deglaciation of former marine based ice streams provide information on the rates and modes by which marine based ice sheets have responded to external forcing factors such as climate change. This paper presents new 36Cl cosmic ray exposure dating from boulders located on two moraines (Glen Brittle and Loch Scavaig) in southern Skye, northwest Scotland. Ages from the Glen Brittle moraines constrain deglaciation of a major marine terminating ice stream, the Barra-Donegal Ice Stream that drained the former British-Irish Ice Sheet, depending on choice of production method and scaling model this occurred 19.9 ± 1.5-17.6 ± 1.3 ka ago. We compare this timing of deglaciation to existing geochronological data and changes in a variety of potential forcing factors constrained through proxy records and numerical models to determine what deglaciation age is most consistent with existing evidence. Another small section of moraine, the Scavaig moraine, is traced offshore through multibeam swath-bathymetry and interpreted as delimiting a later stillstand/readvance stage following ice stream deglaciation. Additional cosmic ray exposure dating from the onshore portion of this moraine indicate that it was deposited 16.3 ± 1.3-15.2 ± 0.9 ka ago. When calculated using the most up-to-date scaling scheme this time of deposition is, within uncertainty, the same as the timing of a widely identified readvance, the Wester Ross Readvance, observed elsewhere in northwest Scotland. This extends the area over which this readvance has potentially occurred, reinforcing the view that it was climatically forced.

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

    Science.gov (United States)

    Ikumi, Nara; Soto-Faraco, Salvador

    2014-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Nara Ikumi

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

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

    International Nuclear Information System (INIS)

    Kubo, H. Dale; Wang Lili

    2002-01-01

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

  12. 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.

  13. Audio Networking in the Music Industry

    Directory of Open Access Journals (Sweden)

    Glebs Kuzmics

    2018-01-01

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

  14. Personalized Audio Systems - a Bayesian Approach

    DEFF Research Database (Denmark)

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

    2013-01-01

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

  15. STEGANOGRAPHY USAGE TO CONTROL MULTIMEDIA STREAM

    Directory of Open Access Journals (Sweden)

    Grzegorz Koziel

    2014-03-01

    Full Text Available In the paper, a proposal of new application for steganography is presented. It is possible to use steganographic techniques to control multimedia stream playback. Special control markers can be included in the sound signal and the player can detect markers and modify the playback parameters according to the hidden instructions. This solution allows for remembering user preferences within the audio track as well as allowing for preparation of various versions of the same content at the production level.

  16. 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.

  17. Nonspeech audio in user interfaces for TV

    NARCIS (Netherlands)

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

    1997-01-01

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

  18. 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.

  19. 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...

  20. Study of audio speakers containing ferrofluid

    Energy Technology Data Exchange (ETDEWEB)

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

    2008-05-21

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

  1. 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…

  2. 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.

  3. 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.

  4. Endocrine disruptors in freshwater streams of Hesse, Germany: changes in concentration levels in the time span from 2003 to 2005.

    Science.gov (United States)

    Quednow, Kristin; Püttmann, Wilhelm

    2008-03-01

    Four small freshwater streams in the region known as Hessisches Ried in Germany were investigated with respect to the temporal and spatial concentration variations of the endocrine disruptors bisphenol A (BPA), 4-tert-octylphenol (4-tert-OP), and the technical isomer mixture of 4-nonylphenol (tech.-4-NP). Measured concentrations of the target compounds in the river water samples ranged from marketing and use of nonylphenols. Results from the analysis of additionally collected water samples from sewage treatment plant (STP) effluents indicate that the STPs cannot be the only sources for tech.-4-NP found in the river water.

  5. Stream Evaluation

    Data.gov (United States)

    Kansas Data Access and Support Center — Digital representation of the map accompanying the "Kansas stream and river fishery resource evaluation" (R.E. Moss and K. Brunson, 1981.U.S. Fish and Wildlife...

  6. 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...

  7. Consequence of audio visual collection in school libraries

    OpenAIRE

    Kuri, Ramesh

    2016-01-01

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

  8. Optimal bus and buffer allocation for a set of leaky-bucket-controlled streams

    NARCIS (Netherlands)

    Boef, den E.; Korst, J.H.M.; Verhaegh, W.F.J.; De Souza, J.N.; Dini, P.; Lorenz, P.

    2004-01-01

    In an in-home digital network (IHDN) it may be expected that several variable-bit-rate streams (audio, video) run simultaneously over a shared communication device, e.g. a bus. The data supply and demand of most of these streams will not be exactly known in advance, but only a coarse traffic

  9. 47 CFR 10.520 - Common audio attention signal.

    Science.gov (United States)

    2010-10-01

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

  10. Debugging of Class-D Audio Power Amplifiers

    DEFF Research Database (Denmark)

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

    2012-01-01

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

  11. Fusion of audio and visual cues for laughter detection

    NARCIS (Netherlands)

    Petridis, Stavros; Pantic, Maja

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

  12. 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

  13. Using Audio-Derived Affective Offset to Enhance TV Recommendation

    DEFF Research Database (Denmark)

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

    2014-01-01

    . First a user's mood profile is determined using 12-class audio-based emotion classifications . An initial TV content item is then displayed to the user based on the extracted mood profile. The user has the option to either accept the recommendation, or to critique the item once or several times......, by navigating the emotion space to request an alternative match. The final match is then compared to the initial match, in terms of the difference in the items' affective parameterization . This offset is then utilized in future recommendation sessions. The system was evaluated by eliciting three different...

  14. Multiple Time-Scale Monitoring to Address Dynamic Seasonality and Storm Pulses of Stream Water Quality in Mountainous Watersheds

    Directory of Open Access Journals (Sweden)

    Hyun-Ju Lee

    2015-11-01

    Full Text Available Rainfall variability and extreme events can amplify the seasonality and storm pulses of stream water chemistry in mountainous watersheds under monsoon climates. To establish a monitoring program optimized for identifying potential risks to stream water quality arising from rainfall variability and extremes, we examined water chemistry data collected on different timescales. At a small forested watershed, bi-weekly sampling lasted over two years, in comparison to three other biweekly sampling sites. In addition, high-frequency continuous measurements of pH, electrical conductivity, and turbidity were conducted in tandem with automatic water sampling at 2 h intervals during eight rainfall events. Biweekly monitoring showed that during the summer monsoon period, electrical conductivity (EC, dissolved oxygen (DO, and dissolved ion concentrations generally decreased, but total suspended solids (TSS slightly increased. A noticeable variation from the usual seasonal pattern was that DO levels substantially decreased during an extended drought. Bi-hourly storm event samplings exhibited large changes in the concentrations of TSS and particulate and dissolved organic carbon (POC; DOC during intense rainfall events. However, extreme fluctuations in sediment export during discharge peaks could be detected only by turbidity measurements at 5 min intervals. Concomitant measurements during rainfall events established empirical relationships between turbidity and TSS or POC. These results suggest that routine monitoring based on weekly to monthly sampling is valid only in addressing general seasonal patterns or long-lasting phenomena such as drought effects. We propose an “adaptive” monitoring scheme that combines routine monitoring for general seasonal patterns and high-frequency instrumental measurements of water quality components exhibiting rapid responses pulsing during intense rainfall events.

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

    OpenAIRE

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

    2016-01-01

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

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

    Science.gov (United States)

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

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

  17. A compact electroencephalogram recording device with integrated audio stimulation system

    Science.gov (United States)

    Paukkunen, Antti K. O.; Kurttio, Anttu A.; Leminen, Miika M.; Sepponen, Raimo E.

    2010-06-01

    A compact (96×128×32 mm3, 374 g), battery-powered, eight-channel electroencephalogram recording device with an integrated audio stimulation system and a wireless interface is presented. The recording device is capable of producing high-quality data, while the operating time is also reasonable for evoked potential studies. The effective measurement resolution is about 4 nV at 200 Hz sample rate, typical noise level is below 0.7 μVrms at 0.16-70 Hz, and the estimated operating time is 1.5 h. An embedded audio decoder circuit reads and plays wave sound files stored on a memory card. The activities are controlled by an 8 bit main control unit which allows accurate timing of the stimuli. The interstimulus interval jitter measured is less than 1 ms. Wireless communication is made through bluetooth and the data recorded are transmitted to an external personal computer (PC) interface in real time. The PC interface is implemented with LABVIEW® and in addition to data acquisition it also allows online signal processing, data storage, and control of measurement activities such as contact impedance measurement, for example. The practical application of the device is demonstrated in mismatch negativity experiment with three test subjects.

  18. Design and development of a highly sensitive, field portable plasma source instrument for on-line liquid stream monitoring and real-time sample analysis

    International Nuclear Information System (INIS)

    Duan, Yixiang; Su, Yongxuan; Jin, Zhe; Abeln, Stephen P.

    2000-01-01

    The development of a highly sensitive, field portable, low-powered instrument for on-site, real-time liquid waste stream monitoring is described in this article. A series of factors such as system sensitivity and portability, plasma source, sample introduction, desolvation system, power supply, and the instrument configuration, were carefully considered in the design of the portable instrument. A newly designed, miniature, modified microwave plasma source was selected as the emission source for spectroscopy measurement, and an integrated small spectrometer with a charge-coupled device detector was installed for signal processing and detection. An innovative beam collection system with optical fibers was designed and used for emission signal collection. Microwave plasma can be sustained with various gases at relatively low power, and it possesses high detection capabilities for both metal and nonmetal pollutants, making it desirable to use for on-site, real-time, liquid waste stream monitoring. An effective in situ sampling system was coupled with a high efficiency desolvation device for direct-sampling liquid samples into the plasma. A portable computer control system is used for data processing. The new, integrated instrument can be easily used for on-site, real-time monitoring in the field. The system possesses a series of advantages, including high sensitivity for metal and nonmetal elements; in situ sampling; compact structure; low cost; and ease of operation and handling. These advantages will significantly overcome the limitations of previous monitoring techniques and make great contributions to environmental restoration and monitoring. (c)

  19. Audio feature extraction using probability distribution function

    Science.gov (United States)

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

    2015-05-01

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

  20. New musical organology : the audio-games

    OpenAIRE

    Zénouda , Hervé

    2012-01-01

    International audience; This article aims to shed light on a new and emerging creative field: " Audio Games, " a crossroad between video games and computer music. Today, a plethora of tiny applications, which propose entertaining audiovisual experiences with a preponderant sound dimension, are available for game consoles, computers, and mobile phones. These experiences represent a new universe where the gameplay of video games is applied to musical composition, hence creating new links betwee...

  1. Audio Networking in the Music Industry

    OpenAIRE

    Glebs Kuzmics; Maaruf Ali

    2018-01-01

    This paper surveys the rôle of computer networking technologies in the music industry. A comparison of their relevant technologies, their defining advantages and disadvantages; analyses and discussion of the situation in the market of network enabled audio products followed by a discussion of different devices are presented. The idea of replacing a proprietary solution with open-source and freeware software programs has been chosen as the fundamental concept of this research. The technologies...

  2. 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.

  3. AudioMUD: a multiuser virtual environment for blind people.

    Science.gov (United States)

    Sánchez, Jaime; Hassler, Tiago

    2007-03-01

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

  4. Decentralized Cloud Method For Multicasting Media Stream

    Directory of Open Access Journals (Sweden)

    D M B N Bandara

    2015-08-01

    Full Text Available With the advancement of Information technology the concept of idea sharing has advanced. Mostly on presentations personal computer and projector have become essentials. But on most occasions for connecting these equipment cables and physical devices are used. This is inefficient and time consuming. If a problem occurs someone with technical knowledge is necessary to solve the situation. The objective of this research is to use the wireless technology to reduce the manual configuration and build up a platform where one can easily share files a visuals media and feedback. A system has been developed to detect all the devices over a network and upon granted permission will share video audio and access controls. Final outcome of the research was a collaborative software bundle which work together on a network. One part of the system is a Desktop Network Software. And other is a Mobile Application. Desktop application can detect all other devices in the network which provides the same facility and if required can allocate a group and share its screen files and have a message stream to each device using multicasting. Mobile application can act as a mobile remote to the host computer of the group which can detect any input from user and pass it to the system.

  5. Endocrine disruptors in freshwater streams of Hesse, Germany: Changes in concentration levels in the time span from 2003 to 2005

    Energy Technology Data Exchange (ETDEWEB)

    Quednow, Kristin [J. W. Goethe University Frankfurt am Main, Institute of Atmospheric and Environmental Sciences, Department of Environmental Analytical Chemistry, Georg-Voigt-Strasse 14, 60054 Frankfurt (Germany)], E-mail: quednow@kristall.uni-frankfurt.de; Puettmann, Wilhelm [J. W. Goethe University Frankfurt am Main, Institute of Atmospheric and Environmental Sciences, Department of Environmental Analytical Chemistry, Georg-Voigt-Strasse 14, 60054 Frankfurt (Germany)

    2008-03-15

    Four small freshwater streams in the region known as Hessisches Ried in Germany were investigated with respect to the temporal and spatial concentration variations of the endocrine disruptors bisphenol A (BPA), 4-tert-octylphenol (4-tert-OP), and the technical isomer mixture of 4-nonylphenol (tech.-4-NP). Measured concentrations of the target compounds in the river water samples ranged from <20 ng/l to 1927 ng/l, <10 ng/l to 770 ng/l, and <10 ng/l to 420 ng/l for BPA, 4-tert-OP and tech.-4-NP, respectively. BPA levels were, with the exception of two samples, below the predicted no-effect concentration (PNEC) for water organisms. Tech.-4-NP concentrations showed a significant tendency of decreasing concentrations during the sampling period. This is mainly attributed to the implementation of the European Directive 2003/53/EG, which restricts both the marketing and use of nonylphenols. Results from the analysis of additionally collected water samples from sewage treatment plant (STP) effluents indicate that the STPs cannot be the only sources for tech.-4-NP found in the river water. - Concentrations of 4-nonylphenols in rivers of Hessisches Ried in Germany decreased in the sampling period from September 2003 to September 2005.

  6. Endocrine disruptors in freshwater streams of Hesse, Germany: Changes in concentration levels in the time span from 2003 to 2005

    International Nuclear Information System (INIS)

    Quednow, Kristin; Puettmann, Wilhelm

    2008-01-01

    Four small freshwater streams in the region known as Hessisches Ried in Germany were investigated with respect to the temporal and spatial concentration variations of the endocrine disruptors bisphenol A (BPA), 4-tert-octylphenol (4-tert-OP), and the technical isomer mixture of 4-nonylphenol (tech.-4-NP). Measured concentrations of the target compounds in the river water samples ranged from <20 ng/l to 1927 ng/l, <10 ng/l to 770 ng/l, and <10 ng/l to 420 ng/l for BPA, 4-tert-OP and tech.-4-NP, respectively. BPA levels were, with the exception of two samples, below the predicted no-effect concentration (PNEC) for water organisms. Tech.-4-NP concentrations showed a significant tendency of decreasing concentrations during the sampling period. This is mainly attributed to the implementation of the European Directive 2003/53/EG, which restricts both the marketing and use of nonylphenols. Results from the analysis of additionally collected water samples from sewage treatment plant (STP) effluents indicate that the STPs cannot be the only sources for tech.-4-NP found in the river water. - Concentrations of 4-nonylphenols in rivers of Hessisches Ried in Germany decreased in the sampling period from September 2003 to September 2005

  7. Guidelines and Procedures for Computing Time-Series Suspended-Sediment Concentrations and Loads from In-Stream Turbidity-Sensor and Streamflow Data

    Science.gov (United States)

    Rasmussen, Patrick P.; Gray, John R.; Glysson, G. Douglas; Ziegler, Andrew C.

    2009-01-01

    In-stream continuous turbidity and streamflow data, calibrated with measured suspended-sediment concentration data, can be used to compute a time series of suspended-sediment concentration and load at a stream site. Development of a simple linear (ordinary least squares) regression model for computing suspended-sediment concentrations from instantaneous turbidity data is the first step in the computation process. If the model standard percentage error (MSPE) of the simple linear regression model meets a minimum criterion, this model should be used to compute a time series of suspended-sediment concentrations. Otherwise, a multiple linear regression model using paired instantaneous turbidity and streamflow data is developed and compared to the simple regression model. If the inclusion of the streamflow variable proves to be statistically significant and the uncertainty associated with the multiple regression model results in an improvement over that for the simple linear model, the turbidity-streamflow multiple linear regression model should be used to compute a suspended-sediment concentration time series. The computed concentration time series is subsequently used with its paired streamflow time series to compute suspended-sediment loads by standard U.S. Geological Survey techniques. Once an acceptable regression model is developed, it can be used to compute suspended-sediment concentration beyond the period of record used in model development with proper ongoing collection and analysis of calibration samples. Regression models to compute suspended-sediment concentrations are generally site specific and should never be considered static, but they represent a set period in a continually dynamic system in which additional data will help verify any change in sediment load, type, and source.

  8. Real time high frequency monitoring of water quality in river streams using a UV-visible spectrometer: interest, limits and consequences for monitoring strategies

    Science.gov (United States)

    Faucheux, Mikaël; Fovet, Ophélie; Gruau, Gérard; Jaffrézic, Anne; Petitjean, Patrice; Gascuel-Odoux, Chantal; Ruiz, Laurent

    2013-04-01

    Stream water chemistry is highly variable in space and time, therefore high frequency water quality measurement methods are likely to lead to conceptual advances in the hydrological sciences. Sub-daily data on water quality improve the characterization of pollutant sources and pathways during flood events as well as during long-term periods [1]. However, real time, high frequency monitoring devices needs to be properly calibrated and validated in real streams. This study analyses data from in situ monitoring of a stream water quality. During two hydrological years (2010-11, 2011-12), a submersible UV-visible spectrometer (Scan Spectrolyser) was used for surface water quality measurement at the outlet of a headwater catchment located at Kervidy-Naizin, Western France (AgrHys long-term hydrological observatory, http://www.inra.fr/ore_agrhys/). The spectrometer is reagentless and equipped with an auto-cleaning system. It allows real time, in situ and high frequency (20 min) measurements and uses a multiwavelengt spectral (200-750 nm) for simultaneous measurement of nitrate, dissolved organic carbon (DOC) and total suspended solids (TSS). A global calibration based on a PLS (Partial Least Squares) regression is provided by the manufacturer as default configuration of the UV-visible spectrometer. We carried out a local calibration of the spectrometer based on nitrates and DOC concentrations analysed in the laboratory from daily manual sampling and sub-daily automatic sampling of flood events. TSS results are compared with 15 min turbidity records from a continuous turdidimeter (Ponsel). The results show a good correlation between laboratory data and spectrometer data both during basis flows periods and flood events. However, the local calibration gives better results than the global one. Nutrient fluxes estimates based on high and different low frequency time series (daily to monthly) are compared to discuss the implication for environmental monitoring strategies. Such

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

    Energy Technology Data Exchange (ETDEWEB)

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

    1993-06-01

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

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

    Directory of Open Access Journals (Sweden)

    Mansoor Hyder

    2013-07-01

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

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

    International Nuclear Information System (INIS)

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

    2013-01-01

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

  12. Do cortical midline variability and low frequency fluctuations mediate William James' "Stream of Consciousness"? "Neurophenomenal Balance Hypothesis" of "Inner Time Consciousness".

    Science.gov (United States)

    Northoff, Georg

    2014-11-01

    William James famously characterized consciousness by 'stream of consciousness' which describes the temporal continuity and flow of the contents of consciousness in our 'inner time consciousness'. More specifically he distinguished between "substantive parts", the contents of consciousness, and "transitive parts", the linkages between different contents. While much research has recently focused on the substantive parts, the neural mechanisms underlying the transitive parts and their characterization by the balance between 'sensible continuity' and 'continuous change' remain unclear. The aim of this paper is to develop so-called neuro-phenomenal hypothesis about specifically the transitive parts and their two phenomenal hallmark features, sensible continuity and continuous change in 'inner time consciousness'. Based on recent findings, I hypothesize that the cortical midline structures and their high degree of variability and strong low frequency fluctuations play an essential role in mediating the phenomenal balance between sensible continuity and continuous change. Copyright © 2014 Elsevier Inc. All rights reserved.

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

    DEFF Research Database (Denmark)

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

    2014-01-01

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

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

    OpenAIRE

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

    2016-01-01

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

  15. Nitrogen saturation in stream ecosystems.

    Science.gov (United States)

    Earl, Stevan R; Valett, H Maurice; Webster, Jackson R

    2006-12-01

    The concept of nitrogen (N) saturation has organized the assessment of N loading in terrestrial ecosystems. Here we extend the concept to lotic ecosystems by coupling Michaelis-Menten kinetics and nutrient spiraling. We propose a series of saturation response types, which may be used to characterize the proximity of streams to N saturation. We conducted a series of short-term N releases using a tracer (15NO3-N) to measure uptake. Experiments were conducted in streams spanning a gradient of background N concentration. Uptake increased in four of six streams as NO3-N was incrementally elevated, indicating that these streams were not saturated. Uptake generally corresponded to Michaelis-Menten kinetics but deviated from the model in two streams where some other growth-critical factor may have been limiting. Proximity to saturation was correlated to background N concentration but was better predicted by the ratio of dissolved inorganic N (DIN) to soluble reactive phosphorus (SRP), suggesting phosphorus limitation in several high-N streams. Uptake velocity, a reflection of uptake efficiency, declined nonlinearly with increasing N amendment in all streams. At the same time, uptake velocity was highest in the low-N streams. Our conceptual model of N transport, uptake, and uptake efficiency suggests that, while streams may be active sites of N uptake on the landscape, N saturation contributes to nonlinear changes in stream N dynamics that correspond to decreased uptake efficiency.

  16. Estudi i implementació del protocol de streaming http live streaming per un client i-phone

    OpenAIRE

    Núñez Vera, Jordi

    2013-01-01

    [ANGLÈS] The aim of this project is, on the one hand, the analysis of Apple's HTTP Live Streaming protocol, which is an adaptative video and audio streaming protocol able to change the streams' bit rate according to the capacity of the media through which it is being transmitted. On the other hand, the project shows a client development of this protocol for the iPhone mobile device describing this platform from scratch. I trace here the necessary steps for developing applications on iOS and I...

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

    Science.gov (United States)

    Kolekar, M. H.; Sengupta, S.

    2005-07-01

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

  18. Convolution-based classification of audio and symbolic representations of music

    DEFF Research Database (Denmark)

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

    2018-01-01

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

  19. Audio frequency in vivo optical coherence elastography

    Science.gov (United States)

    Adie, Steven G.; Kennedy, Brendan F.; Armstrong, Julian J.; Alexandrov, Sergey A.; Sampson, David D.

    2009-05-01

    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.

  20. Audio frequency in vivo optical coherence elastography

    International Nuclear Information System (INIS)

    Adie, Steven G; Kennedy, Brendan F; Armstrong, Julian J; Alexandrov, Sergey A; Sampson, David D

    2009-01-01

    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.

  1. Predistortion of a Bidirectional Cuk Audio Amplifier

    DEFF Research Database (Denmark)

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

    2014-01-01

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

  2. 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

  3. Calibration of an audio frequency noise generator

    DEFF Research Database (Denmark)

    Diamond, Joseph M.

    1966-01-01

    a noise bandwidth Bn = π/2 × (3dB bandwidth). To apply this method to low audio frequencies, the noise bandwidth of the low Q parallel resonant circuit has been found, including the effects of both series and parallel damping. The method has been used to calibrate a General Radio 1390-B noise generator...... it is used for measurement purposes. The spectral density of a noise source may be found by measuring its rms output over a known noise bandwidth. Such a bandwidth may be provided by a passive filter using accurately known elements. For example, the parallel resonant circuit with purely parallel damping has...

  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. Quality models for audiovisual streaming

    Science.gov (United States)

    Thang, Truong Cong; Kim, Young Suk; Kim, Cheon Seog; Ro, Yong Man

    2006-01-01

    Quality is an essential factor in multimedia communication, especially in compression and adaptation. Quality metrics can be divided into three categories: within-modality quality, cross-modality quality, and multi-modality quality. Most research has so far focused on within-modality quality. Moreover, quality is normally just considered from the perceptual perspective. In practice, content may be drastically adapted, even converted to another modality. In this case, we should consider the quality from semantic perspective as well. In this work, we investigate the multi-modality quality from the semantic perspective. To model the semantic quality, we apply the concept of "conceptual graph", which consists of semantic nodes and relations between the nodes. As an typical of multi-modality example, we focus on audiovisual streaming service. Specifically, we evaluate the amount of information conveyed by a audiovisual content where both video and audio channels may be strongly degraded, even audio are converted to text. In the experiments, we also consider the perceptual quality model of audiovisual content, so as to see the difference with semantic quality model.

  6. A survey on Big Data Stream Mining

    African Journals Online (AJOL)

    pc

    2018-03-05

    Mar 5, 2018 ... huge amount of stream like telecommunication systems. So, there ... streams have many challenges for data mining algorithm design like using of ..... A. Bifet and R. Gavalda, "Learning from Time-Changing Data with. Adaptive ...

  7. Industrial-Strength Streaming Video.

    Science.gov (United States)

    Avgerakis, George; Waring, Becky

    1997-01-01

    Corporate training, financial services, entertainment, and education are among the top applications for streaming video servers, which send video to the desktop without downloading the whole file to the hard disk, saving time and eliminating copyrights questions. Examines streaming video technology, lists ten tips for better net video, and ranks…

  8. Dataflow formalisation of real-time streaming applications on a composable and predictable multi-processor SOC

    NARCIS (Netherlands)

    Nelson, A.T.; Goossens, K.G.W.; Akesson, K.B.

    2015-01-01

    Embedded systems often contain multiple applications, some of which have real-time requirements and whose performance must be guaranteed. To efficiently execute applications, modern embedded systems contain Globally Asynchronous Locally Synchronous (GALS) processors, network on chip, DRAM and SRAM

  9. Academic streaming in Europe

    DEFF Research Database (Denmark)

    Falaschi, Alessandro; Mønster, Dan; Doležal, Ivan

    2004-01-01

    The TF-NETCAST task force was active from March 2003 to March 2004, and during this time the mem- bers worked on various aspects of streaming media related to the ultimate goal of setting up common services and infrastructures to enable netcasting of high quality content to the academic community...

  10. Use of Effective Audio in E-learning Courseware

    OpenAIRE

    Ray, Kisor

    2015-01-01

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

  11. Imagination and Modern Audio Visual Form

    Directory of Open Access Journals (Sweden)

    Ana Đurković

    2017-09-01

    Full Text Available Through three episodes Archetype of modern fairy tales, the mysterious world of fantasy and reality,tell as a serious story about archetypes, symbols, knowledge of good and evil. Rts editor: Natasa Neskovic Written and directed by: Suncica Jergovic Editing: Ana Djurkovic How to illuminate concept of phantasy and affective factors in our imagination a priori something so imaginary, by their genetic provenance, such as a movie scene, or digital picture and sound. You can not always avoid the association to a valid phrase of arnhajm’s truth: mass age -massage: the medium is the message. In elementary and tersely definition of „the shot“ from Plaževsky film language there is term for „le cadre“, however these are selected bits of reality, immanent frame that contains the individual act of images divided of the continent’s view of reality, handling the specific code of semantic value, when its’s imaginative, of course, by aesthetic categories and evaluations. In this type of positive simulacrum, it can not be better segment for the current thinking about the limits of imagination and truth in contemporary media, and contemporary global environment, than the original audio-visual forms through whose prism we search throught a fairy tale in a same time myth and imagination as well as exploring its overall impact on the personality. Everything can be a fairy tale, even false, amoral platitudes politicized by political lobbies in a contemporary existing power sistems, but this is no fairy tale authenticity in it, or creative act, nor humanity and artificial and historical entity of a man that is always present in the ethical effort of a true artist. So, we are investigating the conditions of creative images, modalities of audiovisual media in film language,and it is the archetype of the fairy tale, which, with its psychodynamics still exists and which is removed when the modern man is tired of lies and simulations during his global

  12. A Tree Based Broadcast Scheme for (m, k)-firm Real-Time Stream in Wireless Sensor Networks.

    Science.gov (United States)

    Park, HoSung; Kim, Beom-Su; Kim, Kyong Hoon; Shah, Babar; Kim, Ki-Il

    2017-11-09

    Recently, various unicast routing protocols have been proposed to deliver measured data from the sensor node to the sink node within the predetermined deadline in wireless sensor networks. In parallel with their approaches, some applications demand the specific service, which is based on broadcast to all nodes within the deadline, the feasible real-time traffic model and improvements in energy efficiency. However, current protocols based on either flooding or one-to-one unicast cannot meet the above requirements entirely. Moreover, as far as the authors know, there is no study for the real-time broadcast protocol to support the application-specific traffic model in WSN yet. Based on the above analysis, in this paper, we propose a new ( m , k )-firm-based Real-time Broadcast Protocol (FRBP) by constructing a broadcast tree to satisfy the ( m , k )-firm, which is applicable to the real-time model in resource-constrained WSNs. The broadcast tree in FRBP is constructed by the distance-based priority scheme, whereas energy efficiency is improved by selecting as few as nodes on a tree possible. To overcome the unstable network environment, the recovery scheme invokes rapid partial tree reconstruction in order to designate another node as the parent on a tree according to the measured ( m , k )-firm real-time condition and local states monitoring. Finally, simulation results are given to demonstrate the superiority of FRBP compared to the existing schemes in terms of average deadline missing ratio, average throughput and energy consumption.

  13. Cortical Integration of Audio-Visual Information

    Science.gov (United States)

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

    2013-01-01

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

  14. 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.

  15. 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.

  16. Audio scene segmentation for video with generic content

    Science.gov (United States)

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

    2008-01-01

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

  17. "Are You Listening Please?" The Advantages of Electronic Audio Feedback Compared to Written Feedback

    Science.gov (United States)

    Lunt, Tom; Curran, John

    2010-01-01

    Feedback on students' work is, probably, one of the most important aspects of learning, yet students' report, according to the National Union of Students (NUS) Survey of 2008, unhappiness with the feedback process. Students were unhappy with the quality, detail and timing of feedback. This paper examines the benefits of using audio, as opposed to…

  18. Linking Audio and Visual Information while Navigating in a Virtual Reality Kiosk Display

    Science.gov (United States)

    Sullivan, Briana; Ware, Colin; Plumlee, Matthew

    2006-01-01

    3D interactive virtual reality museum exhibits should be easy to use, entertaining, and informative. If the interface is intuitive, it will allow the user more time to learn the educational content of the exhibit. This research deals with interface issues concerning activating audio descriptions of images in such exhibits while the user is…

  19. The use of ambient audio to increase safety and immersion in location-based games

    Science.gov (United States)

    Kurczak, John Jason

    The purpose of this thesis is to propose an alternative type of interface for mobile software being used while walking or running. Our work addresses the problem of visual user interfaces for mobile software be- ing potentially unsafe for pedestrians, and not being very immersive when used for location-based games. In addition, location-based games and applications can be dif- ficult to develop when directly interfacing with the sensors used to track the user's location. These problems need to be addressed because portable computing devices are be- coming a popular tool for navigation, playing games, and accessing the internet while walking. This poses a safety problem for mobile users, who may be paying too much attention to their device to notice and react to hazards in their environment. The difficulty of developing location-based games and other location-aware applications may significantly hinder the prevalence of applications that explore new interaction techniques for ubiquitous computing. We created the TREC toolkit to address the issues with tracking sensors while developing location-based games and applications. We have developed functional location-based applications with TREC to demonstrate the amount of work that can be saved by using this toolkit. In order to have a safer and more immersive alternative to visual interfaces, we have developed ambient audio interfaces for use with mobile applications. Ambient audio uses continuous streams of sound over headphones to present information to mobile users without distracting them from walking safely. In order to test the effectiveness of ambient audio, we ran a study to compare ambient audio with handheld visual interfaces in a location-based game. We compared players' ability to safely navigate the environment, their sense of immersion in the game, and their performance at the in-game tasks. We found that ambient audio was able to significantly increase players' safety and sense of immersion compared to a

  20. Quantifying the role of vegetation in controlling the time-variant age of evapotranspiration, soil water and stream flow

    Science.gov (United States)

    Smith, A.; Tetzlaff, D.; Soulsby, C.

    2017-12-01

    Identifying the sources of water which sustain plant water uptake is an essential prerequisite to understanding the interactions of vegetation and water within the critical zone. Estimating the sources of root-water uptake is complicated by ecohydrological separation, or the notion of "two-water worlds" which distinguishes more mobile and immobile water sources which respectively sustain streamflow and evapotranspiration. Water mobility within the soil determines both the transit time/residence time of water through/in soils and the subsequent age of root-uptake and xylem water. We used time-variant StorAge Selection (SAS) functions to conceptualise the transit/residence times in the critical zone using a dual-storage soil column differentiating gravity (mobile) and tension dependent (immobile) water, calibrated to measured stable isotope signatures of soil water. Storage-discharge relationships [Brutsaert and Nieber, 1977] were used to identify gravity and tension dependent storages. A temporally variable distribution for root water uptake was identified using simulated stable isotopes in xylem and soil water. Composition of δ2H and δ18O was measured in soil water at 4 depths (5, 10, 15, and 20 cm) on 10 occasions, and 5 times for xylem water within the dominant heather (Calluna sp. and Erica sp.) vegetation in a Scottish Highland catchment over a two-year period. Within a 50 cm soil column, we found that more than 53% of the total stored water was water that was present before the start of the simulation. Mean residence times of the mobile water in the upper 20 cm of the soil were 16, 25, 36, and 44 days, respectively. Mean evaporation transit time varied between 9 and 40 days, driven by seasonal changes and precipitation events. Lastly, mean transit times of xylem water ranged between 95-205 days, driven by changes in soil moisture. During low soil moisture (i.e. lower than mean soil moisture), root-uptake was from lower depths, while higher than mean soil

  1. Real-time estimation of TP load in a Mississippi Delta Stream using a dynamic data driven application system

    Science.gov (United States)

    Ying Ouyang; Theodor D. Leininger; Jeff Hatten

    2013-01-01

    Elevated phosphorus (P) in surface waters can cause eutrophication of aquatic ecosystems and can impair water for drinking, industry, agriculture, and recreation. Currently, no effort has been devoted to estimating real-time variation and load of total P (TP) in surface waters due to the lack of suitable and/or cost-effective wireless sensors. However, when considering...

  2. Computing time-series suspended-sediment concentrations and loads from in-stream turbidity-sensor and streamflow data

    Science.gov (United States)

    Rasmussen, Patrick P.; Gray, John R.; Glysson, G. Doug; Ziegler, Andrew C.

    2010-01-01

    Over the last decade, use of a method for computing suspended-sediment concentration and loads using turbidity sensors—primarily nephelometry, but also optical backscatter—has proliferated. Because an in- itu turbidity sensor is capa le of measuring turbidity instantaneously, a turbidity time series can be recorded and related directly to time-varying suspended-sediment concentrations. Depending on the suspended-sediment characteristics of the measurement site, this method can be more reliable and, in many cases, a more accurate means for computing suspended-sediment concentrations and loads than traditional U.S. Geological Survey computational methods. Guidelines and procedures for estimating time s ries of suspended-sediment concentration and loading as a function of turbidity and streamflow data have been published in a U.S. Geological Survey Techniques and Methods Report, Book 3, Chapter C4. This paper is a summary of these guidelines and discusses some of the concepts, s atistical procedures, and techniques used to maintain a multiyear suspended sediment time series.

  3. Trends and sensitivities of low streamflow extremes to discharge timing and magnitude in Pacific Northwest mountain streams

    Science.gov (United States)

    Patrick R. Kormos; Charlie Luce; Seth J. Wenger; Wouter R. Berghuijs

    2016-01-01

    Path analyses of historical streamflow data from the Pacific Northwest indicate that the precipitation amount has been the dominant control on the magnitude of low streamflow extremes compared to the air temperature-affected timing of snowmelt runoff. The relative sensitivities of low streamflow to precipitation and temperature changes have important...

  4. A Tree Based Broadcast Scheme for (m, k-firm Real-Time Stream in Wireless Sensor Networks

    Directory of Open Access Journals (Sweden)

    HoSung Park

    2017-11-01

    Full Text Available Recently, various unicast routing protocols have been proposed to deliver measured data from the sensor node to the sink node within the predetermined deadline in wireless sensor networks. In parallel with their approaches, some applications demand the specific service, which is based on broadcast to all nodes within the deadline, the feasible real-time traffic model and improvements in energy efficiency. However, current protocols based on either flooding or one-to-one unicast cannot meet the above requirements entirely. Moreover, as far as the authors know, there is no study for the real-time broadcast protocol to support the application-specific traffic model in WSN yet. Based on the above analysis, in this paper, we propose a new (m, k-firm-based Real-time Broadcast Protocol (FRBP by constructing a broadcast tree to satisfy the (m, k-firm, which is applicable to the real-time model in resource-constrained WSNs. The broadcast tree in FRBP is constructed by the distance-based priority scheme, whereas energy efficiency is improved by selecting as few as nodes on a tree possible. To overcome the unstable network environment, the recovery scheme invokes rapid partial tree reconstruction in order to designate another node as the parent on a tree according to the measured (m, k-firm real-time condition and local states monitoring. Finally, simulation results are given to demonstrate the superiority of FRBP compared to the existing schemes in terms of average deadline missing ratio, average throughput and energy consumption.

  5. Automatic Detection and Classification of Audio Events for Road Surveillance Applications

    Directory of Open Access Journals (Sweden)

    Noor Almaadeed

    2018-06-01

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

  6. Sound stream segregation: a neuromorphic approach to solve the “cocktail party problem” in real-time

    OpenAIRE

    Thakur, Chetan Singh; Wang, Runchun M.; Afshar, Saeed; Hamilton, Tara J.; Tapson, Jonathan C.; Shamma, Shihab A.; van Schaik, André

    2015-01-01

    The human auditory system has the ability to segregate complex auditory scenes into a foreground component and a background, allowing us to listen to specific speech sounds from a mixture of sounds. Selective attention plays a crucial role in this process, colloquially known as the “cocktail party effect.” It has not been possible to build a machine that can emulate this human ability in real-time. Here, we have developed a framework for the implementation of a neuromorphic sound segregation ...

  7. A Novel Audio Cryptosystem Using Chaotic Maps and DNA Encoding

    Directory of Open Access Journals (Sweden)

    S. J. Sheela

    2017-01-01

    Full Text Available Chaotic maps have good potential in security applications due to their inherent characteristics relevant to cryptography. This paper introduces a new audio cryptosystem based on chaotic maps, hybrid chaotic shift transform (HCST, and deoxyribonucleic acid (DNA encoding rules. The scheme uses chaotic maps such as two-dimensional modified Henon map (2D-MHM and standard map. The 2D-MHM which has sophisticated chaotic behavior for an extensive range of control parameters is used to perform HCST. DNA encoding technology is used as an auxiliary tool which enhances the security of the cryptosystem. The performance of the algorithm is evaluated for various speech signals using different encryption/decryption quality metrics. The simulation and comparison results show that the algorithm can achieve good encryption results and is able to resist several cryptographic attacks. The various types of analysis revealed that the algorithm is suitable for narrow band radio communication and real-time speech encryption applications.

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

    DEFF Research Database (Denmark)

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

    2015-01-01

    We present the concept of adversarial audio in the context of deep neural networks (DNNs) for music content analysis. An adversary is an algorithm that makes minor perturbations to an input that cause major repercussions to the system response. In particular, we design an adversary for a DNN...... that takes as input short-time spectral magnitudes of recorded music and outputs a high-level music descriptor. We demonstrate how this adversary can make the DNN behave in any way with only extremely minor changes to the music recording signal. We show that the adversary cannot be neutralised by a simple...... filtering of the input. Finally, we discuss adversaries in the broader context of the evaluation of music content analysis systems....

  9. Visualising the environmental appearance of audio products

    Energy Technology Data Exchange (ETDEWEB)

    Stilma, M. [Univ. of Twente, Enschede (Netherlands); Stevels, A. [Delft Univ. of Technology, Delft (Netherlands)]|[Philips Consumer Electronics, Eindhoven (Netherlands); Christiaans, H.; Kandachar, P. [Delft Univ. of Technology, Delft (Netherlands)

    2004-07-01

    Can environmental friendliness be communicated by the design style and appearance of products? (such as form, colour, style or material)? Consumers are interested in buying environmental products and design styles might be used as communicative tools. However, current 'green' products show something else. Environmental aspects are chiefly promoted by marketing programs based on technical items like the use of materials, hazardous substances, energy consumption, etc. By a qualitative and exploratory research the environmental design styles according to consumers' opinions were analysed with larger audio products as case study. Visible distinctive differences can be identified between the most and the least environmental rated products. A 'Green flagship', which claims to be environmentally orientated, wasn't recognised as such by consumers. And women and men perceive environmental friendliness in another way. From this research can be concluded that more attention is needed to visualise the good technical environmental performance of products. (orig.)

  10. Audio visual information materials for risk communication

    International Nuclear Information System (INIS)

    Gunji, Ikuko; Tabata, Rimiko; Ohuchi, Naomi

    2005-07-01

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

  11. 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

  12. 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...

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

    DEFF Research Database (Denmark)

    Ljusev, Petar; Andersen, Michael Andreas E.

    2005-01-01

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

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

    OpenAIRE

    Ljusev, Petar; Andersen, Michael Andreas E.

    2005-01-01

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

  15. Unsupervised topic modelling on South African parliament audio data

    CSIR Research Space (South Africa)

    Kleynhans, N

    2014-11-01

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

  16. The Effect of Audio and Animation in Multimedia Instruction

    Science.gov (United States)

    Koroghlanian, Carol; Klein, James D.

    2004-01-01

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

  17. The Use of Audio and Animation in Computer Based Instruction.

    Science.gov (United States)

    Koroghlanian, Carol; Klein, James D.

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

  18. Multi Carrier Modulation Audio Power Amplifier with Programmable Logic

    DEFF Research Database (Denmark)

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

    2009-01-01

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

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

    Science.gov (United States)

    Tucker, Judith Cook

    1992-01-01

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

  20. Efficiency in audio processing : filter banks and transcoding

    NARCIS (Netherlands)

    Lee, Jun Wei

    2007-01-01

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

  1. 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

  2. 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

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

    NARCIS (Netherlands)

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

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

  4. Classifying laughter and speech using audio-visual feature prediction

    NARCIS (Netherlands)

    Petridis, Stavros; Asghar, Ali; Pantic, Maja

    2010-01-01

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

  5. Haptic and Audio-visual Stimuli: Enhancing Experiences and Interaction

    NARCIS (Netherlands)

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

    2010-01-01

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

  6. Automated Speech and Audio Analysis for Semantic Access to Multimedia

    NARCIS (Netherlands)

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

    2006-01-01

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

  7. Automated speech and audio analysis for semantic access to multimedia

    NARCIS (Netherlands)

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

    2006-01-01

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

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

    DEFF Research Database (Denmark)

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

    2014-01-01

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

  9. 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...

  10. Audio Teleconferencing: Low Cost Technology for External Studies Networking.

    Science.gov (United States)

    Robertson, Bill

    1987-01-01

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

  11. Content Discovery from Composite Audio : An unsupervised approach

    NARCIS (Netherlands)

    Lu, L.

    2009-01-01

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

  12. Removable Watermarking Sebagai Pengendalian Terhadap Cyber Crime Pada Audio Digital

    Directory of Open Access Journals (Sweden)

    Reyhani Lian Putri

    2017-08-01

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

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

    Science.gov (United States)

    Johnston, William L.

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

  14. Noise-Canceling Helmet Audio System

    Science.gov (United States)

    Seibert, Marc A.; Culotta, Anthony J.

    2007-01-01

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

  15. 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.

  16. Technical note: Stage and water width measurement of a mountain stream using a simple time-lapse camera

    Directory of Open Access Journals (Sweden)

    P. Leduc

    2018-01-01

    Full Text Available Remote sensing applied to river monitoring adds complementary information useful for understanding the system behaviour. In this paper, we present a method for visual stage gauging and water surface width measurement using a ground-based time-lapse camera and a fully automatic image analysis algorithm for flow monitoring at a river cross section of a steep, bouldery channel. The remote stage measurement was coupled with a water level logger (pressure transducer on site and shows that the image-based method gives a reliable estimate of the water height variation and daily flow record when validated against the pressure transducer (R = 0.91. From the remotely sensed pictures, we also extracted the water width and show that it is possible to correlate water surface width and stage. The images also provide valuable ancillary information for interpreting and understanding flow hydraulics and site weather conditions. This image-based gauging method is a reliable, informative and inexpensive alternative or adjunct to conventional stage measurement especially for remote sites.

  17. 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.

  18. Audio Satellites – Overhearing Everyday Life

    DEFF Research Database (Denmark)

    Breinbjerg, Morten; Højlund, Marie Koldkjær; Riis, Morten S.

    2016-01-01

    around or displaced arbitrarily in a given landscape. In the web interface, the different sound streams from the individual satellites can be mixed together to form a cooperative soundscape. The project thus allows people to tune into and explore the overheard soundscape of everyday life...

  19. High-Order Sparse Linear Predictors for Audio Processing

    DEFF Research Database (Denmark)

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

    2010-01-01

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

  20. Audio Mining with emphasis on Music Genre Classification

    DEFF Research Database (Denmark)

    Meng, Anders

    2004-01-01

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

  1. Stream Clustering of Growing Objects

    Science.gov (United States)

    Siddiqui, Zaigham Faraz; Spiliopoulou, Myra

    We study incremental clustering of objects that grow and accumulate over time. The objects come from a multi-table stream e.g. streams of Customer and Transaction. As the Transactions stream accumulates, the Customers’ profiles grow. First, we use an incremental propositionalisation to convert the multi-table stream into a single-table stream upon which we apply clustering. For this purpose, we develop an online version of K-Means algorithm that can handle these swelling objects and any new objects that arrive. The algorithm also monitors the quality of the model and performs re-clustering when it deteriorates. We evaluate our method on the PKDD Challenge 1999 dataset.

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

    Directory of Open Access Journals (Sweden)

    K. Ulovec

    2018-04-01

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

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

    International Nuclear Information System (INIS)

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

    2006-01-01

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

  4. Correlations of cave levels, stream terraces and planation surfaces along the River Mur-Timing of landscape evolution along the eastern margin of the Alps.

    Science.gov (United States)

    Wagner, Thomas; Fritz, Harald; Stüwe, Kurt; Nestroy, Othmar; Rodnight, Helena; Hellstrom, John; Benischke, Ralf

    2011-11-01

    The transition zone of the Eastern Alps to the Pannonian Basin provides one of the best sources of information on landscape evolution of the Eastern Alpine mountain range. The region was non-glaciated during the entire Pleistocene. Thus, direct influence of glacial carving as a landscape forming process can be excluded and relics of landforms are preserved that date back to at least the Late Neogene. In this study, we provide a correlation between various planation surfaces across the orogen-basin transition. In particular, we use stream terraces, planation surfaces and cave levels that cover a vertical spread of some 700 m. Our correlation is used to show that both sides of the transition zone uplifted together starting at least about 5 Ma ago. For our correlation we use recently published terrestrial cosmogenic nuclide (TCN) burial ages from cave sediments, new optically stimulated luminescence (OSL) ages of a stream terrace and U-Th ages from speleothems. Minimum age constraints of cave levels from burial ages of cave sediments covering the last ~ 4 Ma are used to place age constraints on surface features by parallelizing cave levels with planation surfaces. The OSL results for the top section of the type locality of the Helfbrunn terrace suggest an Early Würm development (80.5 ± 3.7 to 68.7 ± 4.0 ka). The terrace origin as a penultimate gravel deposit (in classical Alpine terminology Riss) is therefore questioned. U-series speleothem ages from caves nearby indicate formation during Marine Isotope Stages (MIS) 5c and 5a which are both interstadial warm periods. As OSL ages from the terrace also show a time of deposition during MIS 5a ending at the MIS 5/4 transition, this supports the idea of temperate climatic conditions at the time of deposition. In general, tectonic activity is interpreted to be the main driving force for the formation and evolution of these landforms, whilst climate change is suggested to be of minor importance. Obvious hiatuses

  5. Fast algorithm for automatically computing Strahler stream order

    Science.gov (United States)

    Lanfear, Kenneth J.

    1990-01-01

    An efficient algorithm was developed to determine Strahler stream order for segments of stream networks represented in a Geographic Information System (GIS). The algorithm correctly assigns Strahler stream order in topologically complex situations such as braided streams and multiple drainage outlets. Execution time varies nearly linearly with the number of stream segments in the network. This technique is expected to be particularly useful for studying the topology of dense stream networks derived from digital elevation model data.

  6. 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…

  7. 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

  8. Cytoplasmic Streaming in the Drosophila Oocyte.

    Science.gov (United States)

    Quinlan, Margot E

    2016-10-06

    Objects are commonly moved within the cell by either passive diffusion or active directed transport. A third possibility is advection, in which objects within the cytoplasm are moved with the flow of the cytoplasm. Bulk movement of the cytoplasm, or streaming, as required for advection, is more common in large cells than in small cells. For example, streaming is observed in elongated plant cells and the oocytes of several species. In the Drosophila oocyte, two stages of streaming are observed: relatively slow streaming during mid-oogenesis and streaming that is approximately ten times faster during late oogenesis. These flows are implicated in two processes: polarity establishment and mixing. In this review, I discuss the underlying mechanism of streaming, how slow and fast streaming are differentiated, and what we know about the physiological roles of the two types of streaming.

  9. Low-cost synchronization of high-speed audio and video recordings in bio-acoustic experiments.

    Science.gov (United States)

    Laurijssen, Dennis; Verreycken, Erik; Geipel, Inga; Daems, Walter; Peremans, Herbert; Steckel, Jan

    2018-02-27

    In this paper, we present a method for synchronizing high-speed audio and video recordings of bio-acoustic experiments. By embedding a random signal into the recorded video and audio data, robust synchronization of a diverse set of sensor streams can be performed without the need to keep detailed records. The synchronization can be performed using recording devices without dedicated synchronization inputs. We demonstrate the efficacy of the approach in two sets of experiments: behavioral experiments on different species of echolocating bats and the recordings of field crickets. We present the general operating principle of the synchronization method, discuss its synchronization strength and provide insights into how to construct such a device using off-the-shelf components. © 2018. Published by The Company of Biologists Ltd.

  10. Dynamical modeling of tidal streams

    International Nuclear Information System (INIS)

    Bovy, Jo

    2014-01-01

    I present a new framework for modeling the dynamics of tidal streams. The framework consists of simple models for the initial action-angle distribution of tidal debris, which can be straightforwardly evolved forward in time. Taking advantage of the essentially one-dimensional nature of tidal streams, the transformation to position-velocity coordinates can be linearized and interpolated near a small number of points along the stream, thus allowing for efficient computations of a stream's properties in observable quantities. I illustrate how to calculate the stream's average location (its 'track') in different coordinate systems, how to quickly estimate the dispersion around its track, and how to draw mock stream data. As a generative model, this framework allows one to compute the full probability distribution function and marginalize over or condition it on certain phase-space dimensions as well as convolve it with observational uncertainties. This will be instrumental in proper data analysis of stream data. In addition to providing a computationally efficient practical tool for modeling the dynamics of tidal streams, the action-angle nature of the framework helps elucidate how the observed width of the stream relates to the velocity dispersion or mass of the progenitor, and how the progenitors of 'orphan' streams could be located. The practical usefulness of the proposed framework crucially depends on the ability to calculate action-angle variables for any orbit in any gravitational potential. A novel method for calculating actions, frequencies, and angles in any static potential using a single orbit integration is described in the Appendix.

  11. Music Genre Classification Using MIDI and Audio Features

    Science.gov (United States)

    Cataltepe, Zehra; Yaslan, Yusuf; Sonmez, Abdullah

    2007-12-01

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

  12. Music Genre Classification Using MIDI and Audio Features

    Directory of Open Access Journals (Sweden)

    Abdullah Sonmez

    2007-01-01

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

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

    International Nuclear Information System (INIS)

    Wood, Paul; Sinton, David

    2010-01-01

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

  14. Understanding the Effect of Audio Communication Delay on Distributed Team Interaction

    Science.gov (United States)

    2013-06-01

    means for members to socialize and learn about each other, engenders development cooperative relationships, and lays a foundation for future interaction...length will result in increases in task completion time and mental workload. 3. Audiovisual technology will moderate the effect of communication...than audio alone. 4. Audiovisual technology will moderate the effect of communication delays such that task completion time and mental workload will

  15. Knowledge discovery from data streams

    CERN Document Server

    Gama, Joao

    2010-01-01

    Since the beginning of the Internet age and the increased use of ubiquitous computing devices, the large volume and continuous flow of distributed data have imposed new constraints on the design of learning algorithms. Exploring how to extract knowledge structures from evolving and time-changing data, Knowledge Discovery from Data Streams presents a coherent overview of state-of-the-art research in learning from data streams.The book covers the fundamentals that are imperative to understanding data streams and describes important applications, such as TCP/IP traffic, GPS data, sensor networks,

  16. Real-time loudspeaker distance estimation with stereo audio

    DEFF Research Database (Denmark)

    2017-01-01

    , using a first microphone arranged adjacent to the first loudspeaker, and acquiring a second recorded signal vector x2 from a second microphone arranged adjacent to the second loudspeaker, wherein x1 and x2 are N-dimensional vectors, setting the distance equal to ηv/f, where v is the speed of sound, f...

  17. Current-Driven Switch-Mode Audio Power Amplifiers

    DEFF Research Database (Denmark)

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

    2012-01-01

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

  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 estimate the direction-of-arrival (DOA) of both the direct path of the audio source and the early reflections. In these methods, the contribution of the early reflections is essentially subtracted from the signal observations before localization of the direct path component, which may reduce the estimation...

  19. 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....

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

    DEFF Research Database (Denmark)

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

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

  1. A review of lossless audio compression standards and algorithms

    Science.gov (United States)

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

    2017-09-01

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

  2. Robustness evaluation of transactional audio watermarking systems

    Science.gov (United States)

    Neubauer, Christian; Steinebach, Martin; Siebenhaar, Frank; Pickel, Joerg

    2003-06-01

    Distribution via Internet is of increasing importance. Easy access, transmission and consumption of digitally represented music is very attractive to the consumer but led also directly to an increasing problem of illegal copying. To cope with this problem watermarking is a promising concept since it provides a useful mechanism to track illicit copies by persistently attaching property rights information to the material. Especially for online music distribution the use of so-called transaction watermarking, also denoted with the term bitstream watermarking, is beneficial since it offers the opportunity to embed watermarks directly into perceptually encoded material without the need of full decompression/compression. Besides the concept of bitstream watermarking, former publications presented the complexity, the audio quality and the detection performance. These results are now extended by an assessment of the robustness of such schemes. The detection performance before and after applying selected attacks is presented for MPEG-1/2 Layer 3 (MP3) and MPEG-2/4 AAC bitstream watermarking, contrasted to the performance of PCM spread spectrum watermarking.

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

    Science.gov (United States)

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

    2015-10-01

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

  4. Stream-processing pipelines: processing of streams on multiprocessor architecture

    NARCIS (Netherlands)

    Kavaldjiev, N.K.; Smit, Gerardus Johannes Maria; Jansen, P.G.

    In this paper we study the timing aspects of the operation of stream-processing applications that run on a multiprocessor architecture. Dependencies are derived for the processing and communication times of the processors in such a system. Three cases of real-time constrained operation and four

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

    DEFF Research Database (Denmark)

    Dimitrov, Smilen; Serafin, Stefania

    2011-01-01

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

  6. A survey of systems for massive stream analytics

    OpenAIRE

    Singh, Maninder Pal; Hoque, Mohammad A.; Tarkoma, Sasu

    2016-01-01

    The immense growth of data demands switching from traditional data processing solutions to systems, which can process a continuous stream of real time data. Various applications employ stream processing systems to provide solutions to emerging Big Data problems. Open-source solutions such as Storm, Spark Streaming, and S4 are the attempts to answer key stream processing questions. The recent introduction of real time stream processing commercial solutions such as Amazon Kinesis, IBM Infospher...

  7. Concurrent audio-visual feedback for supporting drivers at intersections: A study using two linked driving simulators.

    Science.gov (United States)

    Houtenbos, M; de Winter, J C F; Hale, A R; Wieringa, P A; Hagenzieker, M P

    2017-04-01

    A large portion of road traffic crashes occur at intersections for the reason that drivers lack necessary visual information. This research examined the effects of an audio-visual display that provides real-time sonification and visualization of the speed and direction of another car approaching the crossroads on an intersecting road. The location of red blinking lights (left vs. right on the speedometer) and the lateral input direction of beeps (left vs. right ear in headphones) corresponded to the direction from where the other car approached, and the blink and beep rates were a function of the approaching car's speed. Two driving simulators were linked so that the participant and the experimenter drove in the same virtual world. Participants (N = 25) completed four sessions (two with the audio-visual display on, two with the audio-visual display off), each session consisting of 22 intersections at which the experimenter approached from the left or right and either maintained speed or slowed down. Compared to driving with the display off, the audio-visual display resulted in enhanced traffic efficiency (i.e., greater mean speed, less coasting) while not compromising safety (i.e., the time gap between the two vehicles was equivalent). A post-experiment questionnaire showed that the beeps were regarded as more useful than the lights. It is argued that the audio-visual display is a promising means of supporting drivers until fully automated driving is technically feasible. Copyright © 2016. Published by Elsevier Ltd.

  8. 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...

  9. Class D audio amplifiers for high voltage capacitive transducers

    DEFF Research Database (Denmark)

    Nielsen, Dennis

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

  10. 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)

  11. A numerical analysis of shipboard and coastal zone color scanner time series of new production within Gulf Stream cyclonic eddies in the South Atlantic Bight

    Science.gov (United States)

    Pribble, J. Raymond; Walsh, John J.; Dieterle, Dwight A.; Mueller-Karger, Frank E.

    1994-01-01

    Eddy-induced upwelling occurs along the western edge of the Gulf Stream between Cape Canaveral, Florida, and Cape Hatteras, North Carolina, in the South Atlantic Bight (SAB). Coastal zone color scanner images of 1-km resolution spanning the period April 13-21, 1979, were processed to examine these eddy features in relation to concurrent shipboard and current/temperature measurements at moored arrays. A quasi-one-dimensional (z), time dependent biological model, using only nitrate as a nutrient source, has been combined with a three-dimensional physical model in an attempt to replicate the observed phytoplankton field at the northward edge of an eddy. The model is applicable only to the SAB south of the Charleston Bump, at approximately 31.5 deg N, since no feature analogous to the bump exists in the model bathymetry. The modeled chlorophyll, nitrate, and primary production fields of the euphotic zone are very similar to those obtained from the satellite and shipboard data at the leading edges of the observed eddies south of the Charleston Bump. The horizontal and vertical simulated fluxes of nitrate and chlorophyll show that only approximately 10% of the upwelled nitrate is utilized by the phytoplankton of the modeled grid box on the northern edge of the cyclone, while approximately 75% is lost horizontally, with the remainder still in the euphotic zone after the 10-day period of the model. Loss of chlorophyll due to sinking is very small in this strong upwelling region of the cyclone. The model is relatively insensitive to variations in the sinking parameterization and the external nitrate and chlorophyll fields but is very sensitive to a reduction of the maximum potential growth rate to half that measured. Given the success of this model in simulating the new production of the selcted upwelling region, other upwelling regions for which measurements or successful models of physical and biological quantities and rates exist could be modeled similarly.

  12. Design guidelines for audio presentation of graphs and tables

    OpenAIRE

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

    2003-01-01

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

  13. El Digital Audio Tape Recorder. Contra autores y creadores

    Directory of Open Access Journals (Sweden)

    Jun Ono

    2015-01-01

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

  14. 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.

  15. Digital signal processing methods and algorithms for audio conferencing systems

    OpenAIRE

    Lindström, Fredric

    2007-01-01

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

  16. Video equipment of tele dosimetry and audio

    International Nuclear Information System (INIS)

    Ojeda R, M.A.; Padilla C, I.

    2007-01-01

    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)

  17. Relationship Between Satellite-Derived Snow Cover and Snowmelt-Runoff Timing and Stream Power in the Wind River Range, Wyoming

    Science.gov (United States)

    Hall, Dorothy K.; Foster, James L.; DiGirolamo, Nicolo E.; Riggs, George A.

    2010-01-01

    Earlier onset of springtime weather including earlier snowmelt has been documented in the western United States over at least the last 50 years. Because the majority (>70%) of the water supply in the western U.S. comes from snowmelt, analysis of the declining spring snowpack (and shrinking glaciers) has important implications for streamflow management. The amount of water in a snowpack influences stream discharge which can also influence erosion and sediment transport by changing stream power, or the rate at which a stream can do work such as move sediment and erode the stream bed. The focus of this work is the Wind River Range (WRR) in west-central Wyoming. Ten years of Moderate-Resolution Imaging Spectroradiometer (MODIS) snow-cover, cloud- gap-filled (CGF) map products and 30 years of discharge and meteorological station data are studied. Streamflow data from six streams in the WRR drainage basins show lower annual discharge and earlier snowmelt in the decade of the 2000s than in the previous three decades, though no trend of either lower streamflow or earlier snowmelt was observed using MODIS snow-cover maps within the decade of the 2000s. Results show a statistically-significant trend at the 95% confidence level (or higher) of increasing weekly maximum air temperature (for three out of the five meteorological stations studied) in the decade of the 1970s, and also for the 40-year study period. MODIS-derived snow cover (percent of basin covered) measured on 30 April explains over 89% of the variance in discharge for maximum monthly streamflow in the decade of the 2000s using Spearman rank correlation analysis. We also investigated stream power for Bull Lake Creek Above Bull Lake from 1970 to 2009; a statistically-significant end toward reduced stream power was found (significant at the 90% confidence level). Observed changes in streamflow and stream power may be related to increasing weekly maximum air temperature measured during the 40-year study period. The

  18. Streaming Media

    Science.gov (United States)

    Pulley, John

    2009-01-01

    At a time when the evolutionary pace of new media resembles the real-time mutation of certain microorganisms, the age-old question of how best to connect with constituents can seem impossibly complex--even for an elite institution plugged into the motherboard of Silicon Valley. Identifying the most effective vehicle for reaching a particular…

  19. Transmisión de audio usando redes Zigbee

    Directory of Open Access Journals (Sweden)

    David Delgado León

    2011-03-01

    Full Text Available Normal 0 21 false false false MicrosoftInternetExplorer4 /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Tabla normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-parent:""; mso-padding-alt:0cm 5.4pt 0cm 5.4pt; mso-para-margin:0cm; mso-para-margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:10.0pt; font-family:"Times New Roman"; mso-ansi-language:#0400; mso-fareast-language:#0400; mso-bidi-language:#0400;} Normal 0 21 false false false MicrosoftInternetExplorer4 /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Tabla normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-parent:""; mso-padding-alt:0cm 5.4pt 0cm 5.4pt; mso-para-margin:0cm; mso-para-margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:10.0pt; font-family:"Times New Roman"; mso-ansi-language:#0400; mso-fareast-language:#0400; mso-bidi-language:#0400;} Zigbee es un protocolo de comunicaciones basado en el estándar para redes inalámbricas IEEE_802.15.4. Concebido para el control y la monitorización de redes de sensores tanto en entornos industriales, médicos, como domóticos, ha existido un creciente interés por evaluarlo en aplicaciones de multimedia. Aun sin garantizar QoS (Quality of service por su limitado ancho de banda existen un conjunto de aplicaciones para vigilancia, grupos de rescate y salvamento, seguridad en entornos domóticos, grupos desplegados en un área limitada con necesidad de comunicación donde un sistema de audio y video en tiempo real de bajo costo basado en tecnología Zigbee es una idea sumamente atractiva.   Se presenta el diseño de un sistema que permita la comunicación de un grupo de usuarios desplegadas en un área limitada. Utiliza Microcontroladores RISC y tecnología Zigbee. Se investiga la factibilidad de usar la tecnología Zigbee para la transmisión de audio, se analizan

  20. Applying Spatial Audio to Human Interfaces: 25 Years of NASA Experience

    Science.gov (United States)

    Begault, Durand R.; Wenzel, Elizabeth M.; Godfrey, Martine; Miller, Joel D.; Anderson, Mark R.

    2010-01-01

    From the perspective of human factors engineering, the inclusion of spatial audio within a human-machine interface is advantageous from several perspectives. Demonstrated benefits include the ability to monitor multiple streams of speech and non-speech warning tones using a cocktail party advantage, and for aurally-guided visual search. Other potential benefits include the spatial coordination and interaction of multimodal events, and evaluation of new communication technologies and alerting systems using virtual simulation. Many of these technologies were developed at NASA Ames Research Center, beginning in 1985. This paper reviews examples and describes the advantages of spatial sound in NASA-related technologies, including space operations, aeronautics, and search and rescue. The work has involved hardware and software development as well as basic and applied research.

  1. Audio interfaces should be designed based on data visualisation first principles

    OpenAIRE

    Dewey, Christopher; Wakefield, Jonathan P.

    2016-01-01

    Audio mixing interfaces (AMIs) commonly conform to a small number of paradigms. These paradigms have\\ud significant shortcomings. Data visualisation first principles should be employed to consider alternatives. Existing AMI\\ud paradigms are discussed and concepts of image theory and elementary perceptual elements outlined. AMIs should be evaluated by usability experiments however performing these properly is time-consuming. There are many data visualisation options and combinations. Collabora...

  2. Autonomous Byte Stream Randomizer

    Science.gov (United States)

    Paloulian, George K.; Woo, Simon S.; Chow, Edward T.

    2013-01-01

    Net-centric networking environments are often faced with limited resources and must utilize bandwidth as efficiently as possible. In networking environments that span wide areas, the data transmission has to be efficient without any redundant or exuberant metadata. The Autonomous Byte Stream Randomizer software provides an extra level of security on top of existing data encryption methods. Randomizing the data s byte stream adds an extra layer to existing data protection methods, thus making it harder for an attacker to decrypt protected data. Based on a generated crypto-graphically secure random seed, a random sequence of numbers is used to intelligently and efficiently swap the organization of bytes in data using the unbiased and memory-efficient in-place Fisher-Yates shuffle method. Swapping bytes and reorganizing the crucial structure of the byte data renders the data file unreadable and leaves the data in a deconstructed state. This deconstruction adds an extra level of security requiring the byte stream to be reconstructed with the random seed in order to be readable. Once the data byte stream has been randomized, the software enables the data to be distributed to N nodes in an environment. Each piece of the data in randomized and distributed form is a separate entity unreadable on its own right, but when combined with all N pieces, is able to be reconstructed back to one. Reconstruction requires possession of the key used for randomizing the bytes, leading to the generation of the same cryptographically secure random sequence of numbers used to randomize the data. This software is a cornerstone capability possessing the ability to generate the same cryptographically secure sequence on different machines and time intervals, thus allowing this software to be used more heavily in net-centric environments where data transfer bandwidth is limited.

  3. Evaluation of a wireless audio streaming accessory to improve mobile telephone performance of cochlear implant users.

    Science.gov (United States)

    Wolfe, Jace; Morais Duke, Mila; Schafer, Erin; Cire, George; Menapace, Christine; O'Neill, Lori

    2016-01-01

    The objective of this study was to evaluate the potential improvement in word recognition in quiet and in noise obtained with use of a Bluetooth-compatible wireless hearing assistance technology (HAT) relative to the acoustic mobile telephone condition (e.g. the mobile telephone receiver held to the microphone of the sound processor). A two-way repeated measures design was used to evaluate differences in telephone word recognition obtained in quiet and in competing noise in the acoustic mobile telephone condition compared to performance obtained with use of the CI sound processor and a telephone HAT. Sixteen adult users of Nucleus cochlear implants and the Nucleus 6 sound processor were included in this study. Word recognition over the mobile telephone in quiet and in noise was significantly better with use of the wireless HAT compared to performance in the acoustic mobile telephone condition. Word recognition over the mobile telephone was better in quiet when compared to performance in noise. The results of this study indicate that use of a wireless HAT improves word recognition over the mobile telephone in quiet and in noise relative to performance in the acoustic mobile telephone condition for a group of adult cochlear implant recipients.

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

    Directory of Open Access Journals (Sweden)

    P. Dhanalakshmi

    2010-12-01

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

  5. Migrating Home Computer Audio Waveforms to Digital Objects: A Case Study on Digital Archaeology

    Directory of Open Access Journals (Sweden)

    Mark Guttenbrunner

    2011-03-01

    Full Text Available Rescuing data from inaccessible or damaged storage media for the purpose of preserving the digital data for the long term is one of the dimensions of digital archaeology. With the current pace of technological development, any system can become obsolete in a matter of years and hence the data stored in a specific storage media might not be accessible anymore due to the unavailability of the system to access the media. In order to preserve digital records residing in such storage media, it is necessary to extract the data stored in those media by some means.One early storage medium for home computers in the 1980s was audio tape. The first home computer systems allowed the use of standard cassette players to record and replay data. Audio cassettes are more durable than old home computers when properly stored. Devices playing this medium (i.e. tape recorders can be found in working condition or can be repaired, as they are usually made out of standard components. By re-engineering the format of the waveform and the file formats, the data on such media can then be extracted from a digitised audio stream and migrated to a non-obsolete format.In this paper we present a case study on extracting the data stored on an audio tape by an early home computer system, namely the Philips Videopac+ G7400. The original data formats were re-engineered and an application was written to support the migration of the data stored on tapes without using the original system. This eliminates the necessity of keeping an obsolete system alive for enabling access to the data on the storage media meant for this system. Two different methods to interpret the data and eliminate possible errors in the tape were implemented and evaluated on original tapes, which were recorded 20 years ago. Results show that with some error correction methods, parts of the tapes are still readable even without the original system. It also implies that it is easier to build solutions while original

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

    Directory of Open Access Journals (Sweden)

    Xing He

    2008-01-01

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

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

    DEFF Research Database (Denmark)

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

    2013-01-01

    Recommending TV content to groups of viewers is best carried out when relevant information such as the demographics of the group is available. However, it can be difficult and time consuming to extract information for every user in the group. This paper shows how an audio analysis of the age...... and gender of a group of users watching the TV can be used for recommending a sequence of N short TV content items for the group. First, a state of the art audio-based classifier determines the age and gender of each user in an M-user group and creates a group profile. A genetic recommender algorithm...... profile, thus ensuring that items are proportionally allocated to users with respect to their demographic categorization. The proposed system is compared to an ideal system where the group demographics are provided explicitly. Results using real speaker utterances show that, in spite of the inaccuracies...

  8. An Efficient Method for Image and Audio Steganography using Least Significant Bit (LSB) Substitution

    Science.gov (United States)

    Chadha, Ankit; Satam, Neha; Sood, Rakshak; Bade, Dattatray

    2013-09-01

    In order to improve the data hiding in all types of multimedia data formats such as image and audio and to make hidden message imperceptible, a novel method for steganography is introduced in this paper. It is based on Least Significant Bit (LSB) manipulation and inclusion of redundant noise as secret key in the message. This method is applied to data hiding in images. For data hiding in audio, Discrete Cosine Transform (DCT) and Discrete Wavelet Transform (DWT) both are used. All the results displayed prove to be time-efficient and effective. Also the algorithm is tested for various numbers of bits. For those values of bits, Mean Square Error (MSE) and Peak-Signal-to-Noise-Ratio (PSNR) are calculated and plotted. Experimental results show that the stego-image is visually indistinguishable from the original cover-image when nsteganography process does not reveal presence of any hidden message, thus qualifying the criteria of imperceptible message.

  9. Neuromorphic Audio-Visual Sensor Fusion on a Sound-Localising Robot

    Directory of Open Access Journals (Sweden)

    Vincent Yue-Sek Chan

    2012-02-01

    Full Text Available This paper presents the first robotic system featuring audio-visual sensor fusion with neuromorphic sensors. We combine a pair of silicon cochleae and a silicon retina on a robotic platform to allow the robot to learn sound localisation through self-motion and visual feedback, using an adaptive ITD-based sound localisation algorithm. After training, the robot can localise sound sources (white or pink noise in a reverberant environment with an RMS error of 4 to 5 degrees in azimuth. In the second part of the paper, we investigate the source binding problem. An experiment is conducted to test the effectiveness of matching an audio event with a corresponding visual event based on their onset time. The results show that this technique can be quite effective, despite its simplicity.

  10. Embodied accounts of HIV and hope: using audio diaries with interviews.

    Science.gov (United States)

    Bernays, Sarah; Rhodes, Tim; Jankovic Terzic, Katarina

    2014-05-01

    Capturing the complexity of the experience of chronic illness over time presents significant methodological and ethical challenges. In this article, we present methodological and substantive insights from a longitudinal qualitative study with 20 people living with HIV in Serbia. We used both repeated in-depth interviews and audio diaries to explore the role of hope in coping with and managing HIV. Using thematic longitudinal analysis, we found that the audio diaries produced distinctive, embodied accounts that straddled the public/private divide and engaged with alternative social scripts of illness experience. We suggest that this enabled less socially anticipated accounts of coping, hoping, and distress to be spoken and shared. We argue that examining the influence of different methods on accounting not only illustrates the value of qualitative mixed-method study designs but also provides crucial insights to better understand the lived experience of chronic illness.

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

    Science.gov (United States)

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

    2016-05-06

    Use of audio visual tools to aid in medical education is ever on a rise. Our study intends to find the efficacy of a video prepared on "fatty liver," a topic that is often a challenge for pre-clinical teachers, in enhancing cognitive processing and ultimately learning. We prepared a video presentation of 11:36 min, incorporating various concepts of the topic, while keeping in view Mayer's and Ellaway guidelines for multimedia presentation. A pre-post test study on subject knowledge was conducted for 100 students with the video shown as intervention. A retrospective pre study was conducted as a survey which inquired about students understanding of the key concepts of the topic and a feedback on our video was taken. Students performed significantly better in the post test (mean score 8.52 vs. 5.45 in pre-test), positively responded in the retrospective pre-test and gave a positive feedback for our video presentation. Well-designed multimedia tools can aid in cognitive processing and enhance working memory capacity as shown in our study. In times when "smart" device penetration is high, information and communication tools in medical education, which can act as essential aid and not as replacement for traditional curriculums, can be beneficial to the students. © 2015 by The International Union of Biochemistry and Molecular Biology, 44:241-245, 2016. © 2015 The International Union of Biochemistry and Molecular Biology.

  12. Implementation and Analysis Audio Steganography Used Parity Coding for Symmetric Cryptography Key Delivery

    Directory of Open Access Journals (Sweden)

    Afany Zeinata Firdaus

    2013-12-01

    Full Text Available In today's era of communication, online data transactions is increasing. Various information even more accessible, both upload and download. Because it takes a capable security system. Blowfish cryptographic equipped with Audio Steganography is one way to secure the data so that the data can not be accessed by unauthorized parties. In this study Audio Steganography technique is implemented using parity coding method that is used to send the key cryptography blowfish in e-commerce applications based on Android. The results obtained for the average computation time on stage insertion (embedding the secret message is shorter than the average computation time making phase (extracting the secret message. From the test results can also be seen that the more the number of characters pasted the greater the noise received, where the highest SNR is obtained when a character is inserted as many as 506 characters is equal to 11.9905 dB, while the lowest SNR obtained when a character is inserted as many as 2006 characters at 5,6897 dB . Keywords: audio steganograph, parity coding, embedding, extractin, cryptography blowfih.

  13. On Modeling Affect in Audio with Non-Linear Symbolic Dynamics

    Directory of Open Access Journals (Sweden)

    Pauline Mouawad

    2017-09-01

    Full Text Available The discovery of semantic information from complex signals is a task concerned with connecting humans’ perceptions and/or intentions with the signals content. In the case of audio signals, complex perceptions are appraised in a listener’s mind, that trigger affective responses that may be relevant for well-being and survival. In this paper we are interested in the broader question of relations between uncertainty in data as measured using various information criteria and emotions, and we propose a novel method that combines nonlinear dynamics analysis with a method of adaptive time series symbolization that finds the meaningful audio structure in terms of symbolized recurrence properties. In a first phase we obtain symbolic recurrence quantification measures from symbolic recurrence plots, without the need to reconstruct the phase space with embedding. Then we estimate symbolic dynamical invariants from symbolized time series, after embedding. The invariants are: correlation dimension, correlation entropy and Lyapunov exponent. Through their application for the logistic map, we show that our measures are in agreement with known methods from literature. We further show that one symbolic recurrence measure, namely the symbolic Shannon entropy, correlates positively with the positive Lyapunov exponents. Finally we evaluate the performance of our measures in emotion recognition through the implementation of classification tasks for different types of audio signals, and show that in some cases, they perform better than state-of-the-art methods that rely on low-level acoustic features.

  14. The Build-Up Course of Visuo-Motor and Audio-Motor Temporal Recalibration

    Directory of Open Access Journals (Sweden)

    Yoshimori Sugano

    2011-10-01

    Full Text Available The sensorimotor timing is recalibrated after a brief exposure to a delayed feedback of voluntary actions (temporal recalibration effect: TRE (Heron et al., 2009; Stetson et al., 2006; Sugano et al., 2010. We introduce a new paradigm, namely ‘synchronous tapping’ (ST which allows us to investigate how the TRE builds up during adaptation. In each experimental trial, participants were repeatedly exposed to a constant lag (∼150 ms between their voluntary action (pressing a mouse and a feedback stimulus (a visual flash / an auditory click 10 times. Immediately after that, they performed a ST task with the same stimulus as a pace signal (7 flashes / clicks. A subjective ‘no-delay condition’ (∼50 ms served as control. The TRE manifested itself as a change in the tap-stimulus asynchrony that compensated the exposed lag (eg, after lag adaptation, the tap preceded the stimulus more than in control and built up quickly (∼3–6 trials, ∼23–45 sec in both the visuo- and audio-motor domain. The audio-motor TRE was bigger and built-up faster than the visuo-motor one. To conclude, the TRE is comparable between visuo- and audio-motor domain, though they are slightly different in size and build-up rate.

  15. Training on Movement Figure-Ground Discrimination Remediates Low-Level Visual Timing Deficits in the Dorsal Stream, Improving High-Level Cognitive Functioning, Including Attention, Reading Fluency, and Working Memory

    OpenAIRE

    Lawton, Teri; Shelley-Tremblay, John

    2017-01-01

    The purpose of this study was to determine whether neurotraining to discriminate a moving test pattern relative to a stationary background, figure-ground discrimination, improves vision and cognitive functioning in dyslexics, as well as typically-developing normal students. We predict that improving the speed and sensitivity of figure-ground movement discrimination (PATH to Reading neurotraining) acts to remediate visual timing deficits in the dorsal stream, thereby improving processing speed...

  16. Characterization of Flow Paths, Residence Time and Media Chemistry in Complex Landscapes to Integrate Surface, Groundwater and Stream Processes and Inform Models of Hydrologic and Water Quality Response to Land Use Activities; Savannah River Site

    Energy Technology Data Exchange (ETDEWEB)

    Bitew, Menberu [Univ. of Georgia Research Foundation, Inc., Athens, GA (United States); Jackson, Rhett [University of Georgia Research Foundation, Inc.

    2015-02-01

    The objective of this report is to document the methodology used to calculate the three hydro-geomorphic indices: C Index, Nhot spot, and Interflow Contributing Area (IFC Area). These indices were applied in the Upper Four Mile Creek Watershed in order to better understand the potential mechanisms controlling retention time, path lengths, and potential for nutrient and solute metabolism and exchange associated with the geomorphic configurations of the upland contributing areas, groundwater, the riparian zone, and stream channels.

  17. A Content-Adaptive Analysis and Representation Framework for Audio Event Discovery from "Unscripted" Multimedia

    Science.gov (United States)

    Radhakrishnan, Regunathan; Divakaran, Ajay; Xiong, Ziyou; Otsuka, Isao

    2006-12-01

    We propose a content-adaptive analysis and representation framework to discover events using audio features from "unscripted" multimedia such as sports and surveillance for summarization. The proposed analysis framework performs an inlier/outlier-based temporal segmentation of the content. It is motivated by the observation that "interesting" events in unscripted multimedia occur sparsely in a background of usual or "uninteresting" events. We treat the sequence of low/mid-level features extracted from the audio as a time series and identify subsequences that are outliers. The outlier detection is based on eigenvector analysis of the affinity matrix constructed from statistical models estimated from the subsequences of the time series. We define the confidence measure on each of the detected outliers as the probability that it is an outlier. Then, we establish a relationship between the parameters of the proposed framework and the confidence measure. Furthermore, we use the confidence measure to rank the detected outliers in terms of their departures from the background process. Our experimental results with sequences of low- and mid-level audio features extracted from sports video show that "highlight" events can be extracted effectively as outliers from a background process using the proposed framework. We proceed to show the effectiveness of the proposed framework in bringing out suspicious events from surveillance videos without any a priori knowledge. We show that such temporal segmentation into background and outliers, along with the ranking based on the departure from the background, can be used to generate content summaries of any desired length. Finally, we also show that the proposed framework can be used to systematically select "key audio classes" that are indicative of events of interest in the chosen domain.

  18. Streaming Compression of Hexahedral Meshes

    Energy Technology Data Exchange (ETDEWEB)

    Isenburg, M; Courbet, C

    2010-02-03

    We describe a method for streaming compression of hexahedral meshes. Given an interleaved stream of vertices and hexahedral our coder incrementally compresses the mesh in the presented order. Our coder is extremely memory efficient when the input stream documents when vertices are referenced for the last time (i.e. when it contains topological finalization tags). Our coder then continuously releases and reuses data structures that no longer contribute to compressing the remainder of the stream. This means in practice that our coder has only a small fraction of the whole mesh in memory at any time. We can therefore compress very large meshes - even meshes that do not file in memory. Compared to traditional, non-streaming approaches that load the entire mesh and globally reorder it during compression, our algorithm trades a less compact compressed representation for significant gains in speed, memory, and I/O efficiency. For example, on the 456k hexahedra 'blade' mesh, our coder is twice as fast and uses 88 times less memory (only 3.1 MB) with the compressed file increasing about 3% in size. We also present the first scheme for predictive compression of properties associated with hexahedral cells.

  19. 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. 

  20. On-stream chemical element monitor

    International Nuclear Information System (INIS)

    Averitt, O.R.; Dorsch, R.R.

    1979-01-01

    An apparatus and method for on-stream chemical element monitoring are described wherein a multiplicity of sample streams are flowed continuously through individual analytical cells and fluorescence analyses are performed on the sample streams in sequence, together with a method of controlling the time duration of each analysis as a function of the concomitant radiation exposure of a preselected perforate reference material interposed in the sample-radiation source path