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)

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.

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.

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)

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

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.

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.

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...

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.

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....

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%.

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.

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.

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…

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…

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.

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...

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.

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 ...

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.

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...

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

Science.gov (United States)

Romero-Fresco, Pablo; Fryer, Louise

2013-01-01

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

A 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....

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.

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.

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

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.

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...

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.

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.

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...

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

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.

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.

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.

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 ...

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.

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.

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.

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

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...

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.

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)

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.

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.

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.

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....

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.

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

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.

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.

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...

CERN automatic audio-conference service

CERN Multimedia

Sierra Moral, R

2009-01-01

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

CERN automatic audio-conference service

CERN Document Server

Sierra Moral, R

2010-01-01

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

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…

Audio Twister

DEFF Research Database (Denmark)

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

2015-01-01

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

Back to basics audio

CERN Document Server

Nathan, Julian

1998-01-01

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

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

Aurally Aided Visual Search Performance Comparing Virtual Audio Systems

DEFF Research Database (Denmark)

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

2014-01-01

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

Aurally Aided Visual Search Performance Comparing Virtual Audio Systems

DEFF Research Database (Denmark)

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

2014-01-01

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

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; ·     �...

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.

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...

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

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

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...

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.

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...

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.

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...

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...

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.”

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. 

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.

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

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...

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.

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.

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

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.

CERN automatic audio-conference service

International Nuclear Information System (INIS)

Sierra Moral, Rodrigo

2010-01-01

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

CERN automatic audio-conference service

Energy Technology Data Exchange (ETDEWEB)

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

2010-04-01

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

CERN automatic audio-conference service

Science.gov (United States)

Sierra Moral, Rodrigo

2010-04-01

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

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.

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.

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)

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.

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....

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.

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

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…

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.

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

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...

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.

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...

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.

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.

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.

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,

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.

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

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...

Advances in audio source seperation and multisource audio content retrieval

Science.gov (United States)

Vincent, Emmanuel

2012-06-01

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

Roundtable Audio Discussion

Directory of Open Access Journals (Sweden)

Chris Bigum

2007-01-01

Full Text Available RoundTable on Technology, Teaching and Tools. This is a roundtable audio interview conducted by James Farmer, founder of Edublogs, with Anne Bartlett-Bragg (University of Technology Sydney and Chris Bigum (Deakin University. Skype was used to make and record the audio conference and the resulting sound file was edited by Andrew McLauchlan.

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.

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

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.

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

Science.gov (United States)

Ikumi, Nara; Soto-Faraco, Salvador

2014-01-01

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

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

Directory of Open Access Journals (Sweden)

Nara Ikumi

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

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.

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,

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.

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

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.

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...

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

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.

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...

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.

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

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

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.

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

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...

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.

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...

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.

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.

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....

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

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....

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...

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…

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 ...

Presence and the utility of audio spatialization

DEFF Research Database (Denmark)

Bormann, Karsten

2005-01-01

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

Modified BTC Algorithm for Audio Signal Coding

Directory of Open Access Journals (Sweden)

TOMIC, S.

2016-11-01

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

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.

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)

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.

Audio scene segmentation for video with generic content

Science.gov (United States)

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

2008-01-01

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

Digital audio watermarking fundamentals, techniques and challenges

CERN Document Server

Xiang, Yong; Yan, Bin

2017-01-01

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

Distortion Estimation in Compressed Music Using Only Audio Fingerprints

NARCIS (Netherlands)

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

2008-01-01

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

Detecting double compression of audio signal

Science.gov (United States)

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

2010-01-01

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

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.

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...

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...

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...

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

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.

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.

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.

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...

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.

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

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.

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

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

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

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

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...

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)

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)

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.

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.

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

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.

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.

Fusion for Audio-Visual Laughter Detection

NARCIS (Netherlands)

Reuderink, B.

2007-01-01

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

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

OpenAIRE

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

2016-01-01

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

Tourism research and audio methods

DEFF Research Database (Denmark)

Jensen, Martin Trandberg

2016-01-01

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

Newnes audio and Hi-Fi engineer's pocket book

CERN Document Server

Capel, Vivian

2013-01-01

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

DAFX Digital Audio Effects

CERN Document Server

Zö

2011-01-01

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

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.

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.

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...

47 CFR 10.520 - Common audio attention signal.

Science.gov (United States)

2010-10-01

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

Audio Journal in an ELT Context

Directory of Open Access Journals (Sweden)

Neşe Aysin Siyli

2012-09-01

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

Virtual Microphones for Multichannel Audio Resynthesis

Directory of Open Access Journals (Sweden)

Athanasios Mouchtaris

2003-09-01

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

Music Genre Classification Using MIDI and Audio Features

Science.gov (United States)

Cataltepe, Zehra; Yaslan, Yusuf; Sonmez, Abdullah

2007-12-01

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

Realtime Audio with Garbage Collection

OpenAIRE

Matheussen, Kjetil Svalastog

2010-01-01

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

Audio-Visual Classification of Sports Types

DEFF Research Database (Denmark)

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

2015-01-01

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

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...

Spatial audio reproduction with primary ambient extraction

CERN Document Server

He, JianJun

2017-01-01

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

Music Genre Classification Using MIDI and Audio Features

Directory of Open Access Journals (Sweden)

Abdullah Sonmez

2007-01-01

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

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 %....

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.

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

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.

High-Order Sparse Linear Predictors for Audio Processing

DEFF Research Database (Denmark)

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

2010-01-01

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

Audio Mining with emphasis on Music Genre Classification

DEFF Research Database (Denmark)

Meng, Anders

2004-01-01

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

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

NARCIS (Netherlands)

Pocta, P.; Beerends, J.G.

2015-01-01

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

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

Directory of Open Access Journals (Sweden)

Xiang Shijun

2011-01-01

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

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.

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

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...

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…

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.

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.

Audio localization for mobile robots

OpenAIRE

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

2009-01-01

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

EVALUASI KEPUASAN PENGGUNA TERHADAP APLIKASI AUDIO BOOKS

Directory of Open Access Journals (Sweden)

Raditya Maulana Anuraga

2017-02-01

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

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.

Removable Watermarking Sebagai Pengendalian Terhadap Cyber Crime Pada Audio Digital

Directory of Open Access Journals (Sweden)

Reyhani Lian Putri

2017-08-01

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

A listening test system for automotive audio

DEFF Research Database (Denmark)

Christensen, Flemming; Geoff, Martin; Minnaar, Pauli

2005-01-01

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

Voice activity detection using audio-visual information

DEFF Research Database (Denmark)

Petsatodis, Theodore; Pnevmatikakis, Aristodemos; Boukis, Christos

2009-01-01

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

Audio 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

MP3 audio-editing software for the department of radiology

International Nuclear Information System (INIS)

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

2006-01-01

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

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...

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…

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.

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.

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.

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

Energy Technology Data Exchange (ETDEWEB)

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

1993-06-01

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

Portable audio electronics for impedance-based measurements in microfluidics

International Nuclear Information System (INIS)

Wood, Paul; Sinton, David

2010-01-01

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

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.

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

Directory of Open Access Journals (Sweden)

John Mourjopoulos

2007-01-01

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

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

Directory of Open Access Journals (Sweden)

Mourjopoulos John

2007-01-01

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

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.

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.

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. 

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...

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.

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.

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.

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

Science.gov (United States)

Fenner, Peter; Andrews, Ted F.

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

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.

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.

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

Detection and characterization of lightning-based sources using continuous wavelet transform: application to audio-magnetotellurics

Science.gov (United States)

Larnier, H.; Sailhac, P.; Chambodut, A.

2018-01-01

Atmospheric electromagnetic waves created by global lightning activity contain information about electrical processes of the inner and the outer Earth. Large signal-to-noise ratio events are particularly interesting because they convey information about electromagnetic properties along their path. We introduce a new methodology to automatically detect and characterize lightning-based waves using a time-frequency decomposition obtained through the application of continuous wavelet transform. We focus specifically on three types of sources, namely, atmospherics, slow tails and whistlers, that cover the frequency range 10 Hz to 10 kHz. Each wave has distinguishable characteristics in the time-frequency domain due to source shape and dispersion processes. Our methodology allows automatic detection of each type of event in the time-frequency decomposition thanks to their specific signature. Horizontal polarization attributes are also recovered in the time-frequency domain. This procedure is first applied to synthetic extremely low frequency time-series with different signal-to-noise ratios to test for robustness. We then apply it on real data: three stations of audio-magnetotelluric data acquired in Guadeloupe, oversea French territories. Most of analysed atmospherics and slow tails display linear polarization, whereas analysed whistlers are elliptically polarized. The diversity of lightning activity is finally analysed in an audio-magnetotelluric data processing framework, as used in subsurface prospecting, through estimation of the impedance response functions. We show that audio-magnetotelluric processing results depend mainly on the frequency content of electromagnetic waves observed in processed time-series, with an emphasis on the difference between morning and afternoon acquisition. Our new methodology based on the time-frequency signature of lightning-induced electromagnetic waves allows automatic detection and characterization of events in audio

RECURSOS TECNOLÓGICOS AUDIOVISUALES DE FORMACIÓN EN RED: SISTEMAS STREAMING MEDIA Y TELEINMERSIVOS TECHNOLOGICAL RESOURCES FOR ON-LINE INSTRUCTION: STREAMING MEDIA AND TELEIMMERSION SYSTEMS

Directory of Open Access Journals (Sweden)

Juan Antonio Juanes Méndez

2010-06-01

Full Text Available Presentamos dos modalidades tecnológicas de enseñanza en red: la consolidada y ampliamente utilizada tecnología video-streaming, y el futuro de la comunicación a distancia, la teleinmersión. La primera permite la transmisión de audio/vídeo por la red para que puede ser vista por el usuario en su ordenador personal, desde cualquier lugar que disponga de una conexión a red. La información será recibida y decodificada por el usuario final utilizando cualquier reproductor de los que existen en el mercado. La teleinmersión, por su parte, permite crear espacios virtuales de colaboración entre profesionales, ofreciendo entornos muy cercanos a la realidad. Esta tecnología revolucionará, sin duda, nuestros sistemas de enseñanza en los próximos años, facilitando la interacción profesor-alumno. Es evidente que la formación e-learning aporta a los alumnos y a los docentes grandes ventajas como: menores tiempos de aprendizaje, flexibilidad de horarios y de ubicación geográfica, entre otras. We describe two technological modes of on-line teaching: the consolidated and widely used video-streaming mode and teleimmersion, the future of distance communications. The former mode allows the transmission of audio/video through the network so that it can be seen by the user on a PC from anywhere harbouring a network connection. The information is received and decoded by the final user using any reproducer available on the market. Teleimmersion allows the creation of virtual spaces for collaboration among professionals, offering venues that are very similar to reality. This technology will undoubtedly revolutionize our teaching systems in the near future, facilitating instructor-student interaction. It is clear that e-learning- instruction offers both students and instructors huge advantages, such as shorter learning times and schedule and geographic flexibility, among others.

Audio Description as a Pedagogical Tool

Directory of Open Access Journals (Sweden)

Georgina Kleege

2015-05-01

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

Extracting meaning from audio signals - a machine learning approach

DEFF Research Database (Denmark)

Larsen, Jan

2007-01-01

* Machine learning framework for sound search * Genre classification * Music and audio separation * Wind noise suppression......* Machine learning framework for sound search * Genre classification * Music and audio separation * Wind noise suppression...

Improvements of ModalMax High-Fidelity Piezoelectric Audio Device

Science.gov (United States)

Woodard, Stanley E.

2005-01-01

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

Efficiency in audio processing : filter banks and transcoding

NARCIS (Netherlands)

Lee, Jun Wei

2007-01-01

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

Decision-level fusion for audio-visual laughter detection

NARCIS (Netherlands)

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

2008-01-01

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

Decision-Level Fusion for Audio-Visual Laughter Detection

NARCIS (Netherlands)

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

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

Content Discovery from Composite Audio : An unsupervised approach

NARCIS (Netherlands)

Lu, L.

2009-01-01

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

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

Science.gov (United States)

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

2015-01-01

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

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.

Stream.cz a jeho originální seriálová tvorba

OpenAIRE

Vašíčková, Dorota

2017-01-01

In general, the rise of the internet has brought many changes of production and distribution into the audio-visual industry. These changes triggered the development of specific internet portals that offer video content. The thesis focuses on the current development of internet televisions and especially on a particular platform and enriches the current theoretical list of forms with a specific example of a Czech internet television. The case study of the Czech internet television Stream.cz de...

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

Science.gov (United States)

Gallagher, Michael

2014-01-01

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

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

DEFF Research Database (Denmark)

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

2012-01-01

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

Personalized Audio Systems - a Bayesian Approach

DEFF Research Database (Denmark)

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

2013-01-01

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

IELTS speaking instruction through audio/voice conferencing

Directory of Open Access Journals (Sweden)

Hamed Ghaemi

2012-02-01

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

A high efficiency PWM CMOS class-D audio power amplifier

Energy Technology Data Exchange (ETDEWEB)

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

2009-02-15

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

A high efficiency PWM CMOS class-D audio power amplifier

International Nuclear Information System (INIS)

Zhu Zhangming; Liu Lianxi; Yang Yintang; Lei Han

2009-01-01

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

A high efficiency PWM CMOS class-D audio power amplifier

Science.gov (United States)

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

2009-02-01

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

Audio-Visual Fusion for Sound Source Localization and Improved Attention

Energy Technology Data Exchange (ETDEWEB)

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

2011-07-15

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

Audio-Visual Fusion for Sound Source Localization and Improved Attention

International Nuclear Information System (INIS)

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

2011-01-01

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

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

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

DEFF Research Database (Denmark)

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

2016-01-01

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

A review of lossless audio compression standards and algorithms

Science.gov (United States)

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

2017-09-01

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

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

Directory of Open Access Journals (Sweden)

Mansoor Hyder

2013-07-01

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

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

International Nuclear Information System (INIS)

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

2013-01-01

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

WebGL and web audio software lightweight components for multimedia education

Science.gov (United States)

Chang, Xin; Yuksel, Kivanc; Skarbek, Władysław

2017-08-01

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

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.

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.

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.

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

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.

Class D audio amplifiers for high voltage capacitive transducers

DEFF Research Database (Denmark)

Nielsen, Dennis

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

Can audio recording improve patients' recall of outpatient consultations?

DEFF Research Database (Denmark)

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

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

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.

StirMark Benchmark: audio watermarking attacks based on lossy compression

Science.gov (United States)

Steinebach, Martin; Lang, Andreas; Dittmann, Jana

2002-04-01

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

AKTIVITAS SEKUNDER AUDIO UNTUK MENJAGA KEWASPADAAN PENGEMUDI MOBIL INDONESIA

Directory of Open Access Journals (Sweden)

Iftikar Zahedi Sutalaksana

2013-03-01

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

An Interactive Concert Program Based on Infrared Watermark and Audio Synthesis

Science.gov (United States)

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

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

Audio Networking in the Music Industry

Directory of Open Access Journals (Sweden)

Glebs Kuzmics

2018-01-01

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

Frequency Hopping Method for Audio Watermarking

Directory of Open Access Journals (Sweden)

A. Anastasijević

2012-11-01

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

Four-quadrant flyback converter for direct audio power amplification

DEFF Research Database (Denmark)

Ljusev, Petar; Andersen, Michael Andreas E.

2005-01-01

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

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.

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....

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)

Unsupervised topic modelling on South African parliament audio data

CSIR Research Space (South Africa)

Kleynhans, N

2014-11-01

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

Classifying laughter and speech using audio-visual feature prediction

NARCIS (Netherlands)

Petridis, Stavros; Asghar, Ali; Pantic, Maja

2010-01-01

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

Streaming video - præsentation af trailere over internettet

DEFF Research Database (Denmark)

Jensen, Ole Riis; Forchhammer, Søren

1998-01-01

interaktiv tjeneste baseret på realtidsfremvisning af videotrailere, såkaldt streaming video og audio, og hvor brugeren kan få information om en film der tænkes udbudt på en pay-per-view kanal. Der i arbejdets forløb blevet opbygget en demo, der implementerer en sådan tjeneste.......Som et led i Tele Danmark Kabel TV's markedsføring ønsker man at kunne præsentere filmtrailere og andet materiale i form af levende billeder til kunderne via Wold Wide Web (WWW). Dette projekts hovedformål er at undersøge eksisterende metoder og udvikle redskaber til at præsentere en tilsvarende...

Sistema de adquisición y procesamiento de audio

OpenAIRE

Pérez Segurado, Rubén

2015-01-01

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

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

Directory of Open Access Journals (Sweden)

Pachet François

2009-01-01

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

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...

Remotely supported prehospital ultrasound: A feasibility study of real-time image transmission and expert guidance to aid diagnosis in remote and rural communities.

Science.gov (United States)

Eadie, Leila; Mulhern, John; Regan, Luke; Mort, Alasdair; Shannon, Helen; Macaden, Ashish; Wilson, Philip

2017-01-01

Introduction Our aim is to expedite prehospital assessment of remote and rural patients using remotely-supported ultrasound and satellite/cellular communications. In this paradigm, paramedics are remotely-supported ultrasound operators, guided by hospital-based specialists, to record images before receiving diagnostic advice. Technology can support users in areas with little access to medical imaging and suboptimal communications coverage by connecting to multiple cellular networks and/or satellites to stream live ultrasound and audio-video. Methods An ambulance-based demonstrator system captured standard trauma and novel transcranial ultrasound scans from 10 healthy volunteers at 16 locations across the Scottish Highlands. Volunteers underwent brief scanning training before receiving expert guidance via the communications link. Ultrasound images were streamed with an audio/video feed to reviewers for interpretation. Two sessions were transmitted via satellite and 21 used cellular networks. Reviewers rated image and communication quality, and their utility for diagnosis. Transmission latency and bandwidth were recorded, and effects of scanner and reviewer experience were assessed. Results Appropriate views were provided in 94% of the simulated trauma scans. The mean upload rate was 835/150 kbps and mean latency was 114/2072 ms for cellular and satellite networks, respectively. Scanning experience had a significant impact on time to achieve a diagnostic image, and review of offline scans required significantly less time than live-streamed scans. Discussion This prehospital ultrasound system could facilitate early diagnosis and streamlining of treatment pathways for remote emergency patients, being particularly applicable in rural areas worldwide with poor communications infrastructure and extensive transport times.

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

Science.gov (United States)

Wood, Paul; Sinton, David

2010-08-01

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

Design of batch audio/video conversion platform based on JavaEE

Science.gov (United States)

Cui, Yansong; Jiang, Lianpin

2018-03-01

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

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

OpenAIRE

Jorge Caldera Serrano; Felipe Alonso

2007-01-01

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

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...

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.

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

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...

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

CERN Document Server

Lerch, Alexander

2012-01-01

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

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

OpenAIRE

Blai Meléndez-Catalan, Emilio Molina, Emilia Gómez

2017-01-01

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

On the Use of Memory Models in Audio Features

DEFF Research Database (Denmark)

Jensen, Karl Kristoffer

2011-01-01

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

Audio Teleconferencing: Low Cost Technology for External Studies Networking.

Science.gov (United States)

Robertson, Bill

1987-01-01

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

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.

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

OpenAIRE

Cavaco, Sofia; Magalhaes, Joao; Mordido, Gonçalo

2018-01-01

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

Real-time 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.

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.

Haptic and Audio-visual Stimuli: Enhancing Experiences and Interaction

NARCIS (Netherlands)

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

2010-01-01

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

The Effect of Audio and Animation in Multimedia Instruction

Science.gov (United States)

Koroghlanian, Carol; Klein, James D.

2004-01-01

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

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

Science.gov (United States)

Johnston, William L.

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

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

Directory of Open Access Journals (Sweden)

Warrick eRoseboom

2013-04-01

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

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...

Four-quadrant flyback converter for direct audio power amplification

OpenAIRE

Ljusev, Petar; Andersen, Michael Andreas E.

2005-01-01

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

Semantic Analysis of Multimedial Information Usign Both Audio and Visual Clues

Directory of Open Access Journals (Sweden)

Andrej Lukac

2008-01-01

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

Self-oscillating modulators for direct energy conversion audio power amplifiers

Energy Technology Data Exchange (ETDEWEB)

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

2005-07-01

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

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

DEFF Research Database (Denmark)

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

2015-01-01

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

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

Directory of Open Access Journals (Sweden)

Jussi Kuutti

2014-02-01

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

Balancing Audio

DEFF Research Database (Denmark)

Walther-Hansen, Mads

2016-01-01

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

Four-quadrant flyback converter for direct audio power amplification

Energy Technology Data Exchange (ETDEWEB)

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

2005-07-01

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

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.

Parametric Audio Based Decoder and Music Synthesizer for Mobile Applications

NARCIS (Netherlands)

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

2011-01-01

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

Automated Speech and Audio Analysis for Semantic Access to Multimedia

NARCIS (Netherlands)

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

2006-01-01

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

Automated speech and audio analysis for semantic access to multimedia

NARCIS (Netherlands)

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

2006-01-01

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

Audio Feedback -- Better Feedback?

Science.gov (United States)

Voelkel, Susanne; Mello, Luciane V.

2014-01-01

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

StreamMap: Smooth Dynamic Visualization of High-Density Streaming Points.

Science.gov (United States)

Li, Chenhui; Baciu, George; Han, Yu

2018-03-01

Interactive visualization of streaming points for real-time scatterplots and linear blending of correlation patterns is increasingly becoming the dominant mode of visual analytics for both big data and streaming data from active sensors and broadcasting media. To better visualize and interact with inter-stream patterns, it is generally necessary to smooth out gaps or distortions in the streaming data. Previous approaches either animate the points directly or present a sampled static heat-map. We propose a new approach, called StreamMap, to smoothly blend high-density streaming points and create a visual flow that emphasizes the density pattern distributions. In essence, we present three new contributions for the visualization of high-density streaming points. The first contribution is a density-based method called super kernel density estimation that aggregates streaming points using an adaptive kernel to solve the overlapping problem. The second contribution is a robust density morphing algorithm that generates several smooth intermediate frames for a given pair of frames. The third contribution is a trend representation design that can help convey the flow directions of the streaming points. The experimental results on three datasets demonstrate the effectiveness of StreamMap when dynamic visualization and visual analysis of trend patterns on streaming points are required.

Harmonic Enhancement in Low Bitrate Audio Coding Using an Efficient Long-Term Predictor

Directory of Open Access Journals (Sweden)

Song Jeongook

2010-01-01

Full Text Available This paper proposes audio coding using an efficient long-term prediction method to enhance the perceptual quality of audio codecs to speech input signals at low bit-rates. The MPEG-4 AAC-LTP exploited a similar concept, but its improvement was not significant because of small prediction gain due to long prediction lags and aliased components caused by the transformation with a time-domain aliasing cancelation (TDAC technique. The proposed algorithm increases the prediction gain by employing a deharmonizing predictor and a long-term compensation filter. The look-back memory elements are first constructed by applying the de-harmonizing predictor to the input signal, then the prediction residual is encoded and decoded by transform audio coding. Finally, the long-term compensation filter is applied to the updated look-back memory of the decoded prediction residual to obtain synthesized signals. Experimental results show that the proposed algorithm has much lower spectral distortion and higher perceptual quality than conventional approaches especially for harmonic signals, such as voiced speech.

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

Science.gov (United States)

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

2007-11-01

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

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.

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.

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

Self-oscillating modulators for direct energy conversion audio power amplifiers

DEFF Research Database (Denmark)

Ljusev, Petar; Andersen, Michael Andreas E.

2005-01-01

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

An Analysis of Quality of Service (QoS In Live Video Streaming Using Evolved HSPA Network Media

Directory of Open Access Journals (Sweden)

Achmad Zakaria Azhar

2016-10-01

Full Text Available Evolved High Speed Packet Access (HSPA+ is a mobile telecommunication system technology and the evolution of HSPA technology. This technology has a packet data based service with downlink speeds up to 21.1 Mbps and uplink speed up to 11.5 Mbps on the bandwidth 5MHz. This technology is expected to fulfill and support the needs for information that involves all aspects of multimedia such as video and audio, especially live video streaming. By utilizing this technology it will facilitate communicating the information, for example to monitoring the situation of the house, the news coverage at some certain area, and other events in real time. This thesis aims to identify and test the Quality of Service (QoS performance on the network that is used for live video streaming with the parameters of throughput, delay, jitter and packet loss. The software used for monitoring the data traffic of the live video streaming network is wireshark network analyzer. From the test results it is obtained that the average throughput of provider B is 5,295 Kbps bigger than the provider A, the average delay of provider B is 0.618 ms smaller than the provider A, the average jitter of provider B is 0.420 ms smaller than the provider A and the average packet loss of provider B is 0.451% smaller than the provider A.

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…

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.

StreamQRE: Modular Specification and Efficient Evaluation of Quantitative Queries over Streaming Data.

Science.gov (United States)

Mamouras, Konstantinos; Raghothaman, Mukund; Alur, Rajeev; Ives, Zachary G; Khanna, Sanjeev

2017-06-01

Real-time decision making in emerging IoT applications typically relies on computing quantitative summaries of large data streams in an efficient and incremental manner. To simplify the task of programming the desired logic, we propose StreamQRE, which provides natural and high-level constructs for processing streaming data. Our language has a novel integration of linguistic constructs from two distinct programming paradigms: streaming extensions of relational query languages and quantitative extensions of regular expressions. The former allows the programmer to employ relational constructs to partition the input data by keys and to integrate data streams from different sources, while the latter can be used to exploit the logical hierarchy in the input stream for modular specifications. We first present the core language with a small set of combinators, formal semantics, and a decidable type system. We then show how to express a number of common patterns with illustrative examples. Our compilation algorithm translates the high-level query into a streaming algorithm with precise complexity bounds on per-item processing time and total memory footprint. We also show how to integrate approximation algorithms into our framework. We report on an implementation in Java, and evaluate it with respect to existing high-performance engines for processing streaming data. Our experimental evaluation shows that (1) StreamQRE allows more natural and succinct specification of queries compared to existing frameworks, (2) the throughput of our implementation is higher than comparable systems (for example, two-to-four times greater than RxJava), and (3) the approximation algorithms supported by our implementation can lead to substantial memory savings.

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.

Multilevel inverter based class D audio amplifier for capacitive transducers

DEFF Research Database (Denmark)

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

2014-01-01

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

The Use of Audio and Animation in Computer Based Instruction.

Science.gov (United States)

Koroghlanian, Carol; Klein, James D.

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

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?

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...

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.

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

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

Science.gov (United States)

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

2013-01-01

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

A conceptual framework for audio-visual museum media

DEFF Research Database (Denmark)

Kirkedahl Lysholm Nielsen, Mikkel

2017-01-01

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

The Single- and Multichannel Audio Recordings Database (SMARD)

DEFF Research Database (Denmark)

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

2014-01-01

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

Engaging Students with Audio Feedback

Science.gov (United States)

Cann, Alan

2014-01-01

Students express widespread dissatisfaction with academic feedback. Teaching staff perceive a frequent lack of student engagement with written feedback, much of which goes uncollected or unread. Published evidence shows that audio feedback is highly acceptable to students but is underused. This paper explores methods to produce and deliver audio…

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

OpenAIRE

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

2013-01-01

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

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.

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

Science.gov (United States)

Hage, Steffen R; Nieder, Andreas

2015-05-06

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

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.

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.

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

Science.gov (United States)

2016-03-01

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

Audio-Visual Perception System for a Humanoid Robotic Head

Directory of Open Access Journals (Sweden)

Raquel Viciana-Abad

2014-05-01

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

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.

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.

Music Identification System Using MPEG-7 Audio Signature Descriptors

Science.gov (United States)

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

2013-01-01

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

Music Identification System Using MPEG-7 Audio Signature Descriptors

Directory of Open Access Journals (Sweden)

Shingchern D. You

2013-01-01

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

Technical Evaluation Report 31: Internet Audio Products (3/ 3

Directory of Open Access Journals (Sweden)

Jim Rudolph

2004-08-01

Full Text Available Two contrasting additions to the online audio market are reviewed: iVocalize, a browser-based audio-conferencing software, and Skype, a PC-to-PC Internet telephone tool. These products are selected for review on the basis of their success in gaining rapid popular attention and usage during 2003-04. The iVocalize review emphasizes the product’s role in the development of a series of successful online audio communities – notably several serving visually impaired users. The Skype review stresses the ease with which the product may be used for simultaneous PC-to-PC communication among up to five users. Editor’s Note: This paper serves as an introduction to reports about online community building, and reviews of online products for disabled persons, in the next ten reports in this series. JPB, Series Ed.

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.

A Perceptual Model for Sinusoidal Audio Coding Based on Spectral Integration

NARCIS (Netherlands)

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

2005-01-01

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

A perceptual model for sinusoidal audio coding based on spectral integration

NARCIS (Netherlands)

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

2005-01-01

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

Audio-Tactile Integration in Congenitally and Late Deaf Cochlear Implant Users

Science.gov (United States)

Nava, Elena; Bottari, Davide; Villwock, Agnes; Fengler, Ineke; Büchner, Andreas; Lenarz, Thomas; Röder, Brigitte

2014-01-01

Several studies conducted in mammals and humans have shown that multisensory processing may be impaired following congenital sensory loss and in particular if no experience is achieved within specific early developmental time windows known as sensitive periods. In this study we investigated whether basic multisensory abilities are impaired in hearing-restored individuals with deafness acquired at different stages of development. To this aim, we tested congenitally and late deaf cochlear implant (CI) recipients, age-matched with two groups of hearing controls, on an audio-tactile redundancy paradigm, in which reaction times to unimodal and crossmodal redundant signals were measured. Our results showed that both congenitally and late deaf CI recipients were able to integrate audio-tactile stimuli, suggesting that congenital and acquired deafness does not prevent the development and recovery of basic multisensory processing. However, we found that congenitally deaf CI recipients had a lower multisensory gain compared to their matched controls, which may be explained by their faster responses to tactile stimuli. We discuss this finding in the context of reorganisation of the sensory systems following sensory loss and the possibility that these changes cannot be “rewired” through auditory reafferentation. PMID:24918766

Audio-tactile integration in congenitally and late deaf cochlear implant users.

Directory of Open Access Journals (Sweden)

Elena Nava

Full Text Available Several studies conducted in mammals and humans have shown that multisensory processing may be impaired following congenital sensory loss and in particular if no experience is achieved within specific early developmental time windows known as sensitive periods. In this study we investigated whether basic multisensory abilities are impaired in hearing-restored individuals with deafness acquired at different stages of development. To this aim, we tested congenitally and late deaf cochlear implant (CI recipients, age-matched with two groups of hearing controls, on an audio-tactile redundancy paradigm, in which reaction times to unimodal and crossmodal redundant signals were measured. Our results showed that both congenitally and late deaf CI recipients were able to integrate audio-tactile stimuli, suggesting that congenital and acquired deafness does not prevent the development and recovery of basic multisensory processing. However, we found that congenitally deaf CI recipients had a lower multisensory gain compared to their matched controls, which may be explained by their faster responses to tactile stimuli. We discuss this finding in the context of reorganisation of the sensory systems following sensory loss and the possibility that these changes cannot be "rewired" through auditory reafferentation.

A prospective, randomised, controlled study examining binaural beat audio and pre-operative anxiety in patients undergoing general anaesthesia for day case surgery.

Science.gov (United States)

Padmanabhan, R; Hildreth, A J; Laws, D

2005-09-01

Pre-operative anxiety is common and often significant. Ambulatory surgery challenges our pre-operative goal of an anxiety-free patient by requiring people to be 'street ready' within a brief period of time after surgery. Recently, it has been demonstrated that music can be used successfully to relieve patient anxiety before operations, and that audio embedded with tones that create binaural beats within the brain of the listener decreases subjective levels of anxiety in patients with chronic anxiety states. We measured anxiety with the State-Trait Anxiety Inventory questionnaire and compared binaural beat audio (Binaural Group) with an identical soundtrack but without these added tones (Audio Group) and with a third group who received no specific intervention (No Intervention Group). Mean [95% confidence intervals] decreases in anxiety scores were 26.3%[19-33%] in the Binaural Group (p = 0.001 vs. Audio Group, p Binaural beat audio has the potential to decrease acute pre-operative anxiety significantly.

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...

Network Degradation Effects on Different Codec Types and Characteristics of Video Streaming

Directory of Open Access Journals (Sweden)

Jaroslav Frnda

2014-01-01

Full Text Available Nowadays, there is a quickly growing demand for the transmission of voice, video and data over an IP based network. Multimedia, whether we are talking about broadcast, audio and video transmission and others, from a global perspective is growing exponentially with time. With incoming requests from users, new technologies for data transfer are continually developing. Data must be delivered reliably and with the fewest losses at such high speed. Video quality as part of multimedia technology has a very important role nowadays. It is influenced by several factors, where each of them can have many forms and processing. Network performance is the major degradation effect that influences the quality of resulting image. Poor network performance (lack of link capacity, high network load… causes data packet losses or different delivery time for each packet. This work focuses exactly on these network phenomena. It examines the impact of different delays and packet losses on the quality parameters of triple play services, to evaluate the results using objective methods. The aim of this work is to bring a detailed view on the performance of video streaming over IP-based networks.

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

Audio production principles practical studio applications

CERN Document Server

Elmosnino, Stephane

2018-01-01

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

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

Directory of Open Access Journals (Sweden)

Giovany Vega

2016-06-01

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

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

Directory of Open Access Journals (Sweden)

Soraya Ferreira Alves

2016-12-01

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

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

Energy Technology Data Exchange (ETDEWEB)

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

2005-07-01

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

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

DEFF Research Database (Denmark)

Ljusev, Petar; Andersen, Michael Andreas E.

2005-01-01

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

An Exploratory Evaluation of User Interfaces for 3D Audio Mixing

DEFF Research Database (Denmark)

Gelineck, Steven; Korsgaard, Dannie Michael

2015-01-01

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

A 240W Monolithic Class-D Audio Amplifier Output Stage

DEFF Research Database (Denmark)

Nyboe, Flemming; Kaya, Cetin; Risbo, Lars

2006-01-01

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

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...

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

Science.gov (United States)

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

2010-01-01

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

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

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

Science.gov (United States)

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

2018-01-01

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

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

Science.gov (United States)

Dickinson Public Schools, ND. Instructional Media Center.

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

Audio Restoration

Science.gov (United States)

Esquef, Paulo A. A.

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

Design And Construction Of 300W Audio Power Amplifier For Classroom

Directory of Open Access Journals (Sweden)

Shune Lei Aung

2015-07-01

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

Créer des ressources audio pour le cours de FLE

Directory of Open Access Journals (Sweden)

Florence Gérard Lojacono

2010-01-01

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

StreamExplorer: A Multi-Stage System for Visually Exploring Events in Social Streams.

Science.gov (United States)

Wu, Yingcai; Chen, Zhutian; Sun, Guodao; Xie, Xiao; Cao, Nan; Liu, Shixia; Cui, Weiwei

2017-10-18

Analyzing social streams is important for many applications, such as crisis management. However, the considerable diversity, increasing volume, and high dynamics of social streams of large events continue to be significant challenges that must be overcome to ensure effective exploration. We propose a novel framework by which to handle complex social streams on a budget PC. This framework features two components: 1) an online method to detect important time periods (i.e., subevents), and 2) a tailored GPU-assisted Self-Organizing Map (SOM) method, which clusters the tweets of subevents stably and efficiently. Based on the framework, we present StreamExplorer to facilitate the visual analysis, tracking, and comparison of a social stream at three levels. At a macroscopic level, StreamExplorer uses a new glyph-based timeline visualization, which presents a quick multi-faceted overview of the ebb and flow of a social stream. At a mesoscopic level, a map visualization is employed to visually summarize the social stream from either a topical or geographical aspect. At a microscopic level, users can employ interactive lenses to visually examine and explore the social stream from different perspectives. Two case studies and a task-based evaluation are used to demonstrate the effectiveness and usefulness of StreamExplorer.Analyzing social streams is important for many applications, such as crisis management. However, the considerable diversity, increasing volume, and high dynamics of social streams of large events continue to be significant challenges that must be overcome to ensure effective exploration. We propose a novel framework by which to handle complex social streams on a budget PC. This framework features two components: 1) an online method to detect important time periods (i.e., subevents), and 2) a tailored GPU-assisted Self-Organizing Map (SOM) method, which clusters the tweets of subevents stably and efficiently. Based on the framework, we present Stream

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...

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

Directory of Open Access Journals (Sweden)

Rita Kumalasari

2017-05-01

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

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

Science.gov (United States)

Vigil, Kathleen Murray

2017-01-01

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

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

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...

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

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

NARCIS (Netherlands)

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

2015-01-01

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

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

NARCIS (Netherlands)

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

2010-01-01

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

The presentation of expert testimony via live audio-visual communication.

Science.gov (United States)

Miller, R D

1991-01-01

As part of a national effort to improve efficiency in court procedures, the American Bar Association has recommended, on the basis of a number of pilot studies, increased use of current audio-visual technology, such as telephone and live video communication, to eliminate delays caused by unavailability of participants in both civil and criminal procedures. Although these recommendations were made to facilitate court proceedings, and for the convenience of attorneys and judges, they also have the potential to save significant time for clinical expert witnesses as well. The author reviews the studies of telephone testimony that were done by the American Bar Association and other legal research groups, as well as the experience in one state forensic evaluation and treatment center. He also reviewed the case law on the issue of remote testimony. He then presents data from a national survey of state attorneys general concerning the admissibility of testimony via audio-visual means, including video depositions. Finally, he concludes that the option to testify by telephone provides a significant savings in precious clinical time for forensic clinicians in public facilities, and urges that such clinicians work actively to convince courts and/or legislatures in states that do not permit such testimony (currently the majority), to consider accepting it, to improve the effective use of scarce clinical resources in public facilities.

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…

The Vibe: A Versatile Vision-to-Audition Sensory Substitution Device

Directory of Open Access Journals (Sweden)

Sylvain Hanneton

2010-01-01

Full Text Available We describe a sensory substitution scheme that converts a video stream into an audio stream in real-time. It was initially developed as a research tool for studying human ability to learn new ways of perceiving the world: the Vibe can give us the ability to learn a kind of ‘vision’ by audition. It converts a video stream into a continuous stereophonic audio signal that conveys information coded from the video stream. The conversion from the video stream to the audio stream uses a kind of retina with receptive fields. Each receptive field controls a sound source and the user listens to a sound that is a mixture of all these sound sources. Compared to other existing vision-to-audition sensory substitution devices, the Vibe is highly versatile in particular because it uses a set of configurable units working in parallel. In order to demonstrate the validity and interest of this method of vision to audition conversion, we give the results of an experiment involving a pointing task to targets memorised through visual perception or through their auditory conversion by the Vibe. This article is also an opportunity to precisely draw the general specifications of this scheme in order to prepare its implementation on an autonomous/mobile hardware.

Can audio recording of outpatient consultations improve patient outcome?

DEFF Research Database (Denmark)

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

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

A 240W Monolithic Class-D Audio Amplifier Output Stage

OpenAIRE

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

2006-01-01

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

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

Science.gov (United States)

Oringel, Robert S.

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

Fault Diagnosis using Audio and Vibration Signals in a Circulating Pump

International Nuclear Information System (INIS)

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

2012-01-01

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

Rotenone persistence model for montane streams

Science.gov (United States)

Brown, Peter J.; Zale, Alexander V.

2012-01-01

The efficient and effective use of rotenone is hindered by its unknown persistence in streams. Environmental conditions degrade rotenone, but current label instructions suggest fortifying the chemical along a stream based on linear distance or travel time rather than environmental conditions. Our objective was to develop models that use measurements of environmental conditions to predict rotenone persistence in streams. Detailed measurements of ultraviolet radiation, water temperature, dissolved oxygen, total dissolved solids (TDS), conductivity, pH, oxidation–reduction potential (ORP), substrate composition, amount of organic matter, channel slope, and travel time were made along stream segments located between rotenone treatment stations and cages containing bioassay fish in six streams. The amount of fine organic matter, biofilm, sand, gravel, cobble, rubble, small boulders, slope, pH, TDS, ORP, light reaching the stream, energy dissipated, discharge, and cumulative travel time were each significantly correlated with fish death. By using logistic regression, measurements of environmental conditions were paired with the responses of bioassay fish to develop a model that predicted the persistence of rotenone toxicity in streams. This model was validated with data from two additional stream treatment reaches. Rotenone persistence was predicted by a model that used travel time, rubble, and ORP. When this model predicts a probability of less than 0.95, those who apply rotenone can expect incomplete eradication and should plan on fortifying rotenone concentrations. The significance of travel time has been previously identified and is currently used to predict rotenone persistence. However, rubble substrate, which may be associated with the degradation of rotenone by adsorption and volatilization in turbulent environments, was not previously considered.

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

Directory of Open Access Journals (Sweden)

Koji Iwano

2007-03-01

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

Audio-Visual Feedback for Self-monitoring Posture in Ballet Training

DEFF Research Database (Denmark)

Knudsen, Esben Winther; Hølledig, Malte Lindholm; Bach-Nielsen, Sebastian Siem

2017-01-01

An application for ballet training is presented that monitors the posture position (straightness of the spine and rotation of the pelvis) deviation from the ideal position in real-time. The human skeletal data is acquired through a Microsoft Kinect v2. The movement of the student is mirrored......-coded. In an experiment with 9-12 year-old dance students from a ballet school, comparing the audio-visual feedback modality with no feedback leads to an increase in posture accuracy (p

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.

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.

Turkish Music Genre Classification using Audio and Lyrics Features

Directory of Open Access Journals (Sweden)

Önder ÇOBAN

2017-05-01

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

The relationship between basic audio quality and overall listening experience.

Science.gov (United States)

Schoeffler, Michael; Herre, Jürgen

2016-09-01

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

ANALYSIS OF MULTIMODAL FUSION TECHNIQUES FOR AUDIO-VISUAL SPEECH RECOGNITION

Directory of Open Access Journals (Sweden)

D.V. Ivanko

2016-05-01

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

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.

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

Science.gov (United States)

2010-01-01

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

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...

Objective Audio Quality Assessment Based on Spectro-Temporal Modulation Analysis

OpenAIRE

Guo, Ziyuan

2011-01-01

Objective audio quality assessment is an interdisciplinary research area that incorporates audiology and machine learning. Although much work has been made on the machine learning aspect, the audiology aspect also deserves investigation. This thesis proposes a non-intrusive audio quality assessment algorithm, which is based on an auditory model that simulates human auditory system. The auditory model is based on spectro-temporal modulation analysis of spectrogram, which has been proven to be ...

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

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.

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

Directory of Open Access Journals (Sweden)

Md. Rabiul Islam

2014-01-01

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

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.

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.

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

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...

Effect of audio in-vehicle red light-running warning message on driving behavior based on a driving simulator experiment.

Science.gov (United States)

Yan, Xuedong; Liu, Yang; Xu, Yongcun

2015-01-01

Drivers' incorrect decisions of crossing signalized intersections at the onset of the yellow change may lead to red light running (RLR), and RLR crashes result in substantial numbers of severe injuries and property damage. In recent years, some Intelligent Transport System (ITS) concepts have focused on reducing RLR by alerting drivers that they are about to violate the signal. The objective of this study is to conduct an experimental investigation on the effectiveness of the red light violation warning system using a voice message. In this study, the prototype concept of the RLR audio warning system was modeled and tested in a high-fidelity driving simulator. According to the concept, when a vehicle is approaching an intersection at the onset of yellow and the time to the intersection is longer than the yellow interval, the in-vehicle warning system can activate the following audio message "The red light is impending. Please decelerate!" The intent of the warning design is to encourage drivers who cannot clear an intersection during the yellow change interval to stop at the intersection. The experimental results showed that the warning message could decrease red light running violations by 84.3 percent. Based on the logistic regression analyses, drivers without a warning were about 86 times more likely to make go decisions at the onset of yellow and about 15 times more likely to run red lights than those with a warning. Additionally, it was found that the audio warning message could significantly reduce RLR severity because the RLR drivers' red-entry times without a warning were longer than those with a warning. This driving simulator study showed a promising effect of the audio in-vehicle warning message on reducing RLR violations and crashes. It is worthwhile to further develop the proposed technology in field applications.

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

Directory of Open Access Journals (Sweden)

Butko Taras

2011-01-01

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

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

Science.gov (United States)

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

2018-01-01

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

An ESL Audio-Script Writing Workshop

Science.gov (United States)

Miller, Carla

2012-01-01

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

Digital signal processing methods and algorithms for audio conferencing systems

OpenAIRE

Lindström, Fredric

2007-01-01

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

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

Science.gov (United States)

Pimmer, Christoph; Mbvundula, Francis

2018-01-01

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

Impact of Audio-Coaching on the Position of Lung Tumors

International Nuclear Information System (INIS)

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

2008-01-01

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

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.

audio-ultrasonic waves by argon gas discharge

International Nuclear Information System (INIS)

Ragheb, M.S.

2010-01-01

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

Impact of oral health education by audio aids, braille and tactile models on the oral health status of visually impaired children of Bhopal City.

Science.gov (United States)

Gautam, Anjali; Bhambal, Ajay; Moghe, Swapnil

2018-01-01

Children with special needs face unique challenges in day-to-day practice. They are dependent on their close ones for everything. To improve oral hygiene in such visually impaired children, undue training and education are required. Braille is an important language for reading and writing for the visually impaired. It helps them understand and visualize the world via touch. Audio aids are being used to impart health education to the visually impaired. Tactile models help them perceive things which they cannot visualize and hence are an important learning tool. This study aimed to assess the improvement in oral hygiene by audio aids and Braille and tactile models in visually impaired children aged 6-16 years of Bhopal city. This was a prospective study. Sixty visually impaired children aged 6-16 years were selected and randomly divided into three groups (20 children each). Group A: audio aids + Braille, Group B: audio aids + tactile models, and Group C: audio aids + Braille + tactile models. Instructions were given for maintaining good oral hygiene and brushing techniques were explained to all children. After 3 months' time, the oral hygiene status was recorded and compared using plaque and gingival index. ANNOVA test was used. The present study showed a decrease in the mean plaque and gingival scores at all time intervals in individual group as compared to that of the baseline that was statistically significant. The study depicts that the combination of audio aids, Braille and tactile models is an effective way to provide oral health education and improve oral health status of visually impaired children.

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

Stream hydraulics and temperature determine the metabolism of geothermal Icelandic streams

Directory of Open Access Journals (Sweden)

Demars B. O.L.

2011-07-01

Full Text Available Stream ecosystem metabolism plays a critical role in planetary biogeochemical cycling. Stream benthic habitat complexity and the available surface area for microbes relative to the free-flowing water volume are thought to be important determinants of ecosystem metabolism. Unfortunately, the engineered deepening and straightening of streams for drainage purposes could compromise stream natural services. Stream channel complexity may be quantitatively expressed with hydraulic parameters such as water transient storage, storage residence time, and water spiralling length. The temperature dependence of whole stream ecosystem respiration (ER, gross primary productivity (GPP and net ecosystem production (NEP = GPP − ER has recently been evaluated with a “natural experiment” in Icelandic geothermal streams along a 5–25 °C temperature gradient. There remained, however, a substantial amount of unexplained variability in the statistical models, which may be explained by hydraulic parameters found to be unrelated to temperature. We also specifically tested the additional and predicted synergistic effects of water transient storage and temperature on ER, using novel, more accurate, methods. Both ER and GPP were highly related to water transient storage (or water spiralling length but not to the storage residence time. While there was an additional effect of water transient storage and temperature on ER (r2 = 0.57; P = 0.015, GPP was more related to water transient storage than temperature. The predicted synergistic effect could not be confirmed, most likely due to data limitation. Our interpretation, based on causal statistical modelling, is that the metabolic balance of streams (NEP was primarily determined by the temperature dependence of respiration. Further field and experimental work is required to test the predicted synergistic effect on ER. Meanwhile, since higher metabolic activities allow for higher pollutant degradation or uptake

Comparison of Linear Prediction Models for Audio Signals

Directory of Open Access Journals (Sweden)

2009-03-01

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

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

Directory of Open Access Journals (Sweden)

Qiuling Wu

2018-05-01

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

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...

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.

Study of audio speakers containing ferrofluid

Energy Technology Data Exchange (ETDEWEB)

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

2008-05-21

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

Audio-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...

Audio engineering 101 a beginner's guide to music production

CERN Document Server

Dittmar, Tim

2013-01-01

Audio Engineering 101 is a real world guide for starting out in the recording industry. If you have the dream, the ideas, the music and the creativity but don't know where to start, then this book is for you!Filled with practical advice on how to navigate the recording world, from an author with first-hand, real-life experience, Audio Engineering 101 will help you succeed in the exciting, but tough and confusing, music industry. Covering all you need to know about the recording process, from the characteristics of sound to a guide to microphones to analog versus digital

GaN Power Stage for Switch-mode Audio Amplification

DEFF Research Database (Denmark)

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

2015-01-01

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

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

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.

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.

"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…

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

Energy Technology Data Exchange (ETDEWEB)

Kierkegaard, P.

1998-10-01

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

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

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

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…

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

Science.gov (United States)

Aleman-Centeno, Josefina R.

1983-01-01

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

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

Directory of Open Access Journals (Sweden)

Kirsten E Smayda

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

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

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

Science.gov (United States)

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

2017-04-01

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

Approaches to building single-stage AC/AC conversion switch-mode audio power amplifiers

DEFF Research Database (Denmark)

Ljusev, Petar; Andersen, Michael Andreas E.

2004-01-01

This paper discusses the possible topologies and promising approaches towards direct single-phase AC-AC conversion of the mains voltage for audio applications. When compared to standard Class-D switching audio power amplifiers with a separate power supply, it is expected that direct conversion...

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

Science.gov (United States)

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

2012-01-01

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

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

Science.gov (United States)

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

2012-01-01

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

StreamSqueeze: a dynamic stream visualization for monitoring of event data

Science.gov (United States)

Mansmann, Florian; Krstajic, Milos; Fischer, Fabian; Bertini, Enrico

2012-01-01

While in clear-cut situations automated analytical solution for data streams are already in place, only few visual approaches have been proposed in the literature for exploratory analysis tasks on dynamic information. However, due to the competitive or security-related advantages that real-time information gives in domains such as finance, business or networking, we are convinced that there is a need for exploratory visualization tools for data streams. Under the conditions that new events have higher relevance and that smooth transitions enable traceability of items, we propose a novel dynamic stream visualization called StreamSqueeze. In this technique the degree of interest of recent items is expressed through an increase in size and thus recent events can be shown with more details. The technique has two main benefits: First, the layout algorithm arranges items in several lists of various sizes and optimizes the positions within each list so that the transition of an item from one list to the other triggers least visual changes. Second, the animation scheme ensures that for 50 percent of the time an item has a static screen position where reading is most effective and then continuously shrinks and moves to the its next static position in the subsequent list. To demonstrate the capability of our technique, we apply it to large and high-frequency news and syslog streams and show how it maintains optimal stability of the layout under the conditions given above.

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

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...

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

Science.gov (United States)

Anderson, Gerald D.; Olson, David B.

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

Impact of oral health education by audio aids, braille and tactile models on the oral health status of visually impaired children of Bhopal City

Directory of Open Access Journals (Sweden)

Anjali Gautam

2018-01-01

Full Text Available Context: Children with special needs face unique challenges in day-to-day practice. They are dependent on their close ones for everything. To improve oral hygiene in such visually impaired children, undue training and education are required. Braille is an important language for reading and writing for the visually impaired. It helps them understand and visualize the world via touch. Audio aids are being used to impart health education to the visually impaired. Tactile models help them perceive things which they cannot visualize and hence are an important learning tool. Aim: This study aimed to assess the improvement in oral hygiene by audio aids and Braille and tactile models in visually impaired children aged 6–16 years of Bhopal city. Settings and Design: This was a prospective study. Materials and Methods: Sixty visually impaired children aged 6–16 years were selected and randomly divided into three groups (20 children each. Group A: audio aids + Braille, Group B: audio aids + tactile models, and Group C: audio aids + Braille + tactile models. Instructions were given for maintaining good oral hygiene and brushing techniques were explained to all children. After 3 months' time, the oral hygiene status was recorded and compared using plaque and gingival index. Statistical Analysis Used: ANNOVA test was used. Results: The present study showed a decrease in the mean plaque and gingival scores at all time intervals in individual group as compared to that of the baseline that was statistically significant. Conclusions: The study depicts that the combination of audio aids, Braille and tactile models is an effective way to provide oral health education and improve oral health status of visually impaired children.

The Transfer of Learning Associated with Audio Feedback on Written Work

Directory of Open Access Journals (Sweden)

Tanya Martini

2014-11-01

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

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

DEFF Research Database (Denmark)

Laursen, Ditte

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

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

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