WorldWideScience

Sample records for well-reduced audio feature

  1. 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

    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......-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...... 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....

  2. Video salient event classification using audio features

    Science.gov (United States)

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

    2014-03-01

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

  3. 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.

  4. 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...

  5. Survey of compressed domain audio features and their expressiveness

    Science.gov (United States)

    Pfeiffer, Silvia; Vincent, Thomas

    2003-01-01

    We give an overview of existing audio analysis approaches in the compressed domain and incorporate them into a coherent formal structure. After examining the kinds of information accessible in an MPEG-1 compressed audio stream, we describe a coherent approach to determine features from them and report on a number of applications they enable. Most of them aim at creating an index to the audio stream by segmenting the stream into temporally coherent regions, which may be classified into pre-specified types of sounds such as music, speech, speakers, animal sounds, sound effects, or silence. Other applications centre around sound recognition such as gender, beat or speech recognition.

  6. 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.

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

    Directory of Open Access Journals (Sweden)

    Adit Jamdar

    2015-05-01

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

  8. 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.

  9. Feature Selection for Audio Surveillance in Urban Environment

    Directory of Open Access Journals (Sweden)

    KIKTOVA Eva

    2014-05-01

    Full Text Available This paper presents the work leading to the acoustic event detection system, which is designed to recognize two types of acoustic events (shot and breaking glass in urban environment. For this purpose, a huge front-end processing was performed for the effective parametric representation of an input sound. MFCC features and features computed during their extraction (MELSPEC and FBANK, then MPEG-7 audio descriptors and other temporal and spectral characteristics were extracted. High dimensional feature sets were created and in the next phase reduced by the mutual information based selection algorithms. Hidden Markov Model based classifier was applied and evaluated by the Viterbi decoding algorithm. Thus very effective feature sets were identified and also the less important features were found.

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

    OpenAIRE

    Saleh Al-Zhrani; Mubarak AlQahtani

    2010-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Saleh Al-Zhrani

    2010-01-01

    Full Text Available Problem statement: This study investigated zero crossing features and selected MPEG-7 audio descriptors for environment sound recognition applications such as audio forensics. Approach: The study implemented several experiments focusing on the problems of environment recognition from audio particularly for forensic applications. Results: It was investigated the effect of the temporal zero crossing feature as well as selected MPEG-7 audio low level descriptors on environment sound recognition. The performance was evaluated against a varying number of training sounds and samples per training file. Conclusion/Recommendations: Experimental results showed that higher recognition accuracy is achieved by increasing the number of training files and by decreasing the number of samples per training file. This study presented an audio environment recognition using zero crossing features and MPEG-7 Descriptors.

  12. Bimodal Log-linear Regression for Fusion of Audio and Visual Features

    NARCIS (Netherlands)

    Rudovic, Ognjen; Petridis, Stavros; Pantic, Maja

    2013-01-01

    One of the most commonly used audiovisual fusion approaches is feature-level fusion where the audio and visual features are concatenated. Although this approach has been successfully used in several applications, it does not take into account interactions between the features, which can be a problem

  13. Automatic Segmentation of News Items Based on Video and Audio Features

    Institute of Scientific and Technical Information of China (English)

    王伟强; 高文

    2002-01-01

    The automatic segmentation of news items is a key for implementing the automatic cataloging system of news video. This paper presents an approach which manages audio and video feature information to automatically segment news items. The integration of audio and visual analyses can overcome the weakness of the approach using only image analysis techniques. It makes the approach more adaptable to various situations of news items. The proposed approach detects silence segments in accompanying audio, and integrates them with shot segmentation results, as well as anchor shot detection results, to determine the boundaries among news items. Experimental results show that the integration of audio and video features is an effective approach to solving the problem of automatic segmentation of news items.

  14. 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.

  15. 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.

  16. An Analysis of Audio Features to Develop a Human Activity Recognition Model Using Genetic Algorithms, Random Forests, and Neural Networks

    Directory of Open Access Journals (Sweden)

    Carlos E. Galván-Tejada

    2016-01-01

    Full Text Available This work presents a human activity recognition (HAR model based on audio features. The use of sound as an information source for HAR models represents a challenge because sound wave analyses generate very large amounts of data. However, feature selection techniques may reduce the amount of data required to represent an audio signal sample. Some of the audio features that were analyzed include Mel-frequency cepstral coefficients (MFCC. Although MFCC are commonly used in voice and instrument recognition, their utility within HAR models is yet to be confirmed, and this work validates their usefulness. Additionally, statistical features were extracted from the audio samples to generate the proposed HAR model. The size of the information is necessary to conform a HAR model impact directly on the accuracy of the model. This problem also was tackled in the present work; our results indicate that we are capable of recognizing a human activity with an accuracy of 85% using the HAR model proposed. This means that minimum computational costs are needed, thus allowing portable devices to identify human activities using audio as an information source.

  17. Integrating Audio-Visual Features and Text Information for Story Segmentation of News Video

    Institute of Scientific and Technical Information of China (English)

    Liu Hua-yong; Zhou Dong-ru

    2003-01-01

    Video data are composed of multimodal information streams including visual, auditory and textual streams, so an approach of story segmentation for news video using multimodal analysis is described in this paper. The proposed approach detects the topic-caption frames, and integrates them with silence clips detection results, as well as shot segmentation results to locate the news story boundaries. The integration of audio-visual features and text information overcomes the weakness of the approach using only image analysis techniques. On test data with 135 400 frames, when the boundaries between news stories are detected, the accuracy rate 85.8% and the recall rate 97.5% are obtained. The experimental results show the approach is valid and robust.

  18. Integrating Audio-Visual Features and Text Information for Story Segmentation of News Video

    Institute of Scientific and Technical Information of China (English)

    LiuHua-yong; ZhouDong-ru

    2003-01-01

    Video data are composed of multimodal information streams including visual, auditory and textual streams, an approach of story segmentation for news video using multimodal analysis is described in this paper. The proposed approach detects the topic-caption frames, and integrates them with silence clips detection results, as well as shot segmentation results to locate the news story boundaries. The integration of audio-visual features and text information overcomes the weakness of the approach using only image analysis techniques. On test data with 135 400 frames, when the boundaries between news stories are detected, the accuracy rate 85.8% and the recall rate 97.5% are obtained. The experimental results show the approach is valid and robust.

  19. Online Learning for Classification of Low-rank Representation Features and Its Applications in Audio Segment Classification

    CERN Document Server

    Shi, Ziqiang; Zheng, Tieran; Deng, Shiwen

    2011-01-01

    In this paper, a novel framework based on trace norm minimization for audio segment is proposed. In this framework, both the feature extraction and classification are obtained by solving corresponding convex optimization problem with trace norm regularization. For feature extraction, robust principle component analysis (robust PCA) via minimization a combination of the nuclear norm and the $\\ell_1$-norm is used to extract low-rank features which are robust to white noise and gross corruption for audio segments. These low-rank features are fed to a linear classifier where the weight and bias are learned by solving similar trace norm constrained problems. For this classifier, most methods find the weight and bias in batch-mode learning, which makes them inefficient for large-scale problems. In this paper, we propose an online framework using accelerated proximal gradient method. This framework has a main advantage in memory cost. In addition, as a result of the regularization formulation of matrix classificatio...

  20. 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 ...

  1. Detecting paralinguistic events in audio stream using context in features and probabilistic decisions☆

    Science.gov (United States)

    Gupta, Rahul; Audhkhasi, Kartik; Lee, Sungbok; Narayanan, Shrikanth

    2017-01-01

    Non-verbal communication involves encoding, transmission and decoding of non-lexical cues and is realized using vocal (e.g. prosody) or visual (e.g. gaze, body language) channels during conversation. These cues perform the function of maintaining conversational flow, expressing emotions, and marking personality and interpersonal attitude. In particular, non-verbal cues in speech such as paralanguage and non-verbal vocal events (e.g. laughters, sighs, cries) are used to nuance meaning and convey emotions, mood and attitude. For instance, laughters are associated with affective expressions while fillers (e.g. um, ah, um) are used to hold floor during a conversation. In this paper we present an automatic non-verbal vocal events detection system focusing on the detect of laughter and fillers. We extend our system presented during Interspeech 2013 Social Signals Sub-challenge (that was the winning entry in the challenge) for frame-wise event detection and test several schemes for incorporating local context during detection. Specifically, we incorporate context at two separate levels in our system: (i) the raw frame-wise features and, (ii) the output decisions. Furthermore, our system processes the output probabilities based on a few heuristic rules in order to reduce erroneous frame-based predictions. Our overall system achieves an Area Under the Receiver Operating Characteristics curve of 95.3% for detecting laughters and 90.4% for fillers on the test set drawn from the data specifications of the Interspeech 2013 Social Signals Sub-challenge. We perform further analysis to understand the interrelation between the features and obtained results. Specifically, we conduct a feature sensitivity analysis and correlate it with each feature's stand alone performance. The observations suggest that the trained system is more sensitive to a feature carrying higher discriminability with implications towards a better system design. PMID:28713197

  2. 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....

  3. 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

  4. 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...... of the written paper through its specific use of media, a sonic awareness of aesthetics and materiality, and creative approach towards communication. The audio paper is a performative format working together with an affective and elaborate understanding of language. It is an experiment embracing intellectual...... arguments and creative work, papers and performances, written scholarship and sonic aesthetics. For this special issue of Seismograf, the guidelines for authors and peer reviewers mainly focused on the format. Topic-wise we encouraged dealing with site-specificity and topics related to the island Amager...

  5. Concept Framework for Audio Information Retrieval: ARF

    Institute of Scientific and Technical Information of China (English)

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

    2003-01-01

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

  6. Multipurpose audio watermarking algorithm

    Institute of Scientific and Technical Information of China (English)

    Ning CHEN; Jie ZHU

    2008-01-01

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

  7. AC-3 audio coder

    Science.gov (United States)

    Todd, Craig

    1995-12-01

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

  8. 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.

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

    Directory of Open Access Journals (Sweden)

    Theodoros Giannakopoulos

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

  10. Balancing Audio

    DEFF Research Database (Denmark)

    Walther-Hansen, Mads

    2016-01-01

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

  11. Structure Learning in Audio

    DEFF Research Database (Denmark)

    Nielsen, Andreas Brinch

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

  12. Fuzzy Logic-Based Audio Pattern Recognition

    Science.gov (United States)

    Malcangi, M.

    2008-11-01

    Audio and audio-pattern recognition is becoming one of the most important technologies to automatically control embedded systems. Fuzzy logic may be the most important enabling methodology due to its ability to rapidly and economically model such application. An audio and audio-pattern recognition engine based on fuzzy logic has been developed for use in very low-cost and deeply embedded systems to automate human-to-machine and machine-to-machine interaction. This engine consists of simple digital signal-processing algorithms for feature extraction and normalization, and a set of pattern-recognition rules manually tuned or automatically tuned by a self-learning process.

  13. 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...

  14. Audio Classification from Time-Frequency Texture

    CERN Document Server

    Yu, Guoshen

    2008-01-01

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

  15. Audio Indexing for Efficiency

    Science.gov (United States)

    Rahnlom, Harold F.; Pedrick, Lillian

    1978-01-01

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

  16. Audio-Visual Classification of Sports Types

    DEFF Research Database (Denmark)

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

    2015-01-01

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

  17. Back to basics audio

    CERN Document Server

    Nathan, Julian

    1998-01-01

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

  18. 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.

  19. Principles of Audio Watermarking

    Directory of Open Access Journals (Sweden)

    Martin Hrncar

    2008-01-01

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

  20. Audio Papers - A Manifesto

    DEFF Research Database (Denmark)

    Krogh Groth, Sanne; Samson, Kristine

    2016-01-01

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

  1. Digital Audio Legal Recorder

    Data.gov (United States)

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

  2. Robust audio hashing for audio authentication watermarking

    Science.gov (United States)

    Zmudzinski, Sascha; Steinebach, Martin

    2008-02-01

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

  3. 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.

  4. The Study of Audio Watermarking

    Institute of Scientific and Technical Information of China (English)

    王景; 唐晟

    2011-01-01

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

  5. The Schema Features and Aesthetic Functions of the Foreign Language Teaching with Electric Audio-visual Aids%外语电化教学的图式特征与美育功能

    Institute of Scientific and Technical Information of China (English)

    齐欣

    2015-01-01

    外语电化教学对传统外语教学模式提出挑战的同时,其自身也面临着诸多的挑战,需要更多的理论支撑和功能研究。基于图式理论和美育教育,对外语电化教学图式特征及其隐性、感性、个性三种美育功能的创新审视,进一步丰富了外语电化教学的理论基础,并强调了其美育功能实现的必要性。%While the foreign language teaching with electric audio-visual aids brings about challenges to the traditional language teaching,it is also faced with many challenges,and more studies on its theoretical basis and functions are encouraged. On the basis of Schema Theory and aesthetic education,this paper makes an innovative examination of the schema features of foreign language teaching with electric audio-visual aids and its implicit,emotional,and personalized aesthetic functions,further enriches its theoretical basis and emphasizes the necessity of achieving its aesthetic functions.

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

    DEFF Research Database (Denmark)

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

    2015-01-01

    by a qualitative analysis of subject responses. Distraction ratings were collected for one hundred randomly created audio-on-audio interference situations with music target and interferer programs. The selected features were related to the overall loudness, loudness ratio, perceptual evaluation of audio source......There are many situations in which multiple audio programs are replayed over loudspeakers in the same acoustic environment, allowing listeners to focus on their desired target program. Where this situation is deliberately created and the different program items are centrally controlled, each...... listener can be viewed as having a personal sound zone system. In order to evaluate and optimize such situations in a perceptually relevant manner, the authors created a predictive model using the features that contribute to the distraction from unwanted sounds. Feature extraction was motivated...

  7. Watermarking-Based Digital Audio Data Authentication

    Directory of Open Access Journals (Sweden)

    Jana Dittmann

    2003-09-01

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

  8. Portable Audio Design

    DEFF Research Database (Denmark)

    Groth, Sanne Krogh

    2014-01-01

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

  9. Forensic audio watermark detection

    Science.gov (United States)

    Steinebach, Martin; Zmudzinski, Sascha; Petrautzki, Dirk

    2012-03-01

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

  10. Introduction to AVS Audio

    Institute of Scientific and Technical Information of China (English)

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

    2006-01-01

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

  11. A Physiologically Inspired Method for Audio Classification

    Directory of Open Access Journals (Sweden)

    David V. Anderson

    2005-06-01

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

  12. Voice activity detection using audio-visual information

    DEFF Research Database (Denmark)

    Petsatodis, Theodore; Pnevmatikakis, Aristodemos; Boukis, Christos

    2009-01-01

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

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

    NARCIS (Netherlands)

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

    2012-01-01

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

  14. Layered indexing of home video based on audio signals

    Science.gov (United States)

    Ogawa, Tomomi; Aizawa, Kiyoharu

    2003-12-01

    In this paper, we propose a home video indexing using an audio information to detect an event both a rules-based method and a GMM-based method. Although exclusive audio segmentation and classification was usually used, various sounds overlap in practice, in which case an audio in which various sound overlapped is expressed by a labeling layered index. With the rules-based method, low-level audio features are used to determine indexes, which are classied such as speech, silence, music, and EVN(Environment Noise). The GMM-based method which uses the same features as the rule based method also classifies an audio into the four classes. Smoothing is applied in order to determine the index. We show experiments in a few home video data.

  15. Dynamic Bayesian Networks for Audio-Visual Speech Recognition

    Directory of Open Access Journals (Sweden)

    Liang Luhong

    2002-01-01

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

  16. 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.

  17. DAFX Digital Audio Effects

    CERN Document Server

    2011-01-01

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

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

    Science.gov (United States)

    Rui, Rui; Bao, Chang-chun

    2013-03-01

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

  19. 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…

  20. Categorizing Video Game Audio

    DEFF Research Database (Denmark)

    Westerberg, Andreas Rytter; Schoenau-Fog, Henrik

    2015-01-01

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

  1. Embedded Audio Without Beeps

    DEFF Research Database (Denmark)

    Overholt, Daniel; Møbius, Nikolaj Friis

    2014-01-01

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

  2. 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.

  3. Circuit Bodging: Audio Multiplexer

    NARCIS (Netherlands)

    Allen, B.

    2010-01-01

    Audio amplifiers usually come with a single, glaring design flaw: Not enough auxiliary inputs. Not only that, but you’re usually required to press a button to switch between the amplifier’s limited number of inputs. This is unacceptable - we have better things to do than change input channels! In

  4. 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

  5. Efficient audio signal processing for embedded systems

    Science.gov (United States)

    Chiu, Leung Kin

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

  6. Processing features of audio and video files

    Directory of Open Access Journals (Sweden)

    E. N. Vydalko

    2012-11-01

    Full Text Available Currently the analog videotape recorders using is a thing of the past. Therefore, digital video recording became actual and attractive for the users who put image quality above all else. It is important to make a video recording in digital format without digital signal into analog signal converting. The last leads to a significant loss of quality records. A program processes video stream of digital cable TV is described in this article. Also it can convert video stream of digital cable TV into a format that can easily be used by any computer or DVD-player in digital form.

  7. Quality Enhancement of Compressed Audio Based on Statistical Conversion

    Directory of Open Access Journals (Sweden)

    Mouchtaris Athanasios

    2008-01-01

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

  8. Quality Enhancement of Compressed Audio Based on Statistical Conversion

    Directory of Open Access Journals (Sweden)

    Chris Kyriakakis

    2008-07-01

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

  9. Parametric Coding of Stereo Audio

    Directory of Open Access Journals (Sweden)

    Erik Schuijers

    2005-06-01

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

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

    Directory of Open Access Journals (Sweden)

    Lhoucine Bahatti

    2016-05-01

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

  11. 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

  12. Audio in Place

    DEFF Research Database (Denmark)

    Chamberlain, Alan; Bødker, Mads; Hazzard, Adrian

    2016-01-01

    experience in the world. Vision, and to some extent the tactile senses have been dominant modalities discussed in experiential terms within HCI. This workshop suggests that there is a need to better understand how sound can be used for shaping and augmenting the experiential qualities of places through......Audio-based content, location and mobile technologies can offer a multitude of interactional possibilities when combined in innovative and creative ways. It is important not to underestimate impact of the interplay between location, place and sound. Even if intangible and ephemeral, sounds impact...

  13. 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)

  14. Audio Watermarking with Error Correction

    CERN Document Server

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

    2011-01-01

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

  15. Audio Watermarking with Error Correction

    Directory of Open Access Journals (Sweden)

    Aman Chadha

    2011-09-01

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

  16. Digital audio and video broadcasting by satellite

    Science.gov (United States)

    Yoshino, Takehiko

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

  17. Digital Audio Collections

    Directory of Open Access Journals (Sweden)

    Jason Tenter

    2010-11-01

    Full Text Available

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

  18. 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...

  19. Digital Audio Watermarking: An Overview

    Directory of Open Access Journals (Sweden)

    Bhuvnesh Kumar Singh

    2013-10-01

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

  20. 一种基于 MDCT 量化系数统计特征的A AC 音频隐写分析方法%A steganalysis method of AAC audio based on statistical features of MDCT quantized coefficients

    Institute of Scientific and Technical Information of China (English)

    王昱洁; 杨萍; 蒋薇薇

    2015-01-01

    文章提出了一种基于MDCT量化系数统计特征的AAC音频隐写分析方法。将AAC音频进行部分解码得到MDCT量化系数,在MDCT量化系数中提取广义高斯分布模型的参数、量化系数分布直方图的频域统计矩、帧内和帧间MDCT量化系数的Markov转移矩阵的部分数据作为隐写分析的特征,最后采用支持向量机进行分类。通过对不同比特率的AAC音频的实验结果表明,文中提出的AAC音频隐写分析方法对于MDCT量化系数中的直接扩频隐写方式的检测效果较好,对于比特率为128 kb/s的AAC音频,在隐写容量较低的情况下也能达到较高的检测率。%A steganalysis method of AAC audio based on statistical features of MDCT quantized coeffi‐cients is proposed .Firstly ,AAC audio is partly decoded to get MDCT quantized coefficients ,and then the parameters of generalized Gaussian distribution (GGD) model ,the statistical moments in fre‐quency domain of the distribution histogram of quantized coefficients ,and some data of the Markov transition matrix of the MDCT quantized coefficients in a frame and between frames are extracted from the MDCT quantized coefficients as the features of steganalysis .Finally ,the support vector machine is used as a classifier .The experimental results of AAC audio at different bitrates reveal that ,the pro‐posed steganalysis method of AAC audio has a good detection effect on the hiding method of the direct spread spectrum modulation on the MDCT quantized coefficients ,and for the AAC audio at 128 kb/s bitrate ,the detection accuracy is high even under the condition of low capacity of steganography .

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

    Directory of Open Access Journals (Sweden)

    Urel Sawyers

    2005-11-01

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

  2. A centralized audio presentation manager

    Energy Technology Data Exchange (ETDEWEB)

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

    1994-05-16

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

  3. Tag Based Audio Search Engine

    Directory of Open Access Journals (Sweden)

    Parameswaran Vellachu

    2012-03-01

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

  4. 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...

  5. WLAN Technologies for Audio Delivery

    Directory of Open Access Journals (Sweden)

    Nicolas-Alexander Tatlas

    2007-01-01

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

  6. 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,

  7. 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....

  8. Audio Steganography with Embedded Text

    Science.gov (United States)

    Teck Jian, Chua; Chai Wen, Chuah; Rahman, Nurul Hidayah Binti Ab.; Hamid, Isredza Rahmi Binti A.

    2017-08-01

    Audio steganography is about hiding the secret message into the audio. It is a technique uses to secure the transmission of secret information or hide their existence. It also may provide confidentiality to secret message if the message is encrypted. To date most of the steganography software such as Mp3Stego and DeepSound use block cipher such as Advanced Encryption Standard or Data Encryption Standard to encrypt the secret message. It is a good practice for security. However, the encrypted message may become too long to embed in audio and cause distortion of cover audio if the secret message is too long. Hence, there is a need to encrypt the message with stream cipher before embedding the message into the audio. This is because stream cipher provides bit by bit encryption meanwhile block cipher provide a fixed length of bits encryption which result a longer output compare to stream cipher. Hence, an audio steganography with embedding text with Rivest Cipher 4 encryption cipher is design, develop and test in this project.

  9. Modeling Audio Fingerprints: Structure, Distortion, Capacity

    OpenAIRE

    Doets, P.J.O.

    2010-01-01

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

  10. 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.

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

    Directory of Open Access Journals (Sweden)

    Iwano Koji

    2007-01-01

    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.

  12. 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.

  13. Location audio simplified capturing your audio and your audience

    CERN Document Server

    Miles, Dean

    2014-01-01

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

  14. Evaluation of Audio Compression Artifacts

    Directory of Open Access Journals (Sweden)

    M. Herrera Martinez

    2007-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Jon Baggaley

    2006-06-01

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

  16. Amplificador de audio Clase D

    OpenAIRE

    2012-01-01

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

  17. 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

  18. 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...

  19. Audio watermark a comprehensive foundation using Matlab

    CERN Document Server

    Lin, Yiqing

    2015-01-01

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

  20. 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...

  1. The Audio-Visual Man.

    Science.gov (United States)

    Babin, Pierre, Ed.

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

  2. 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.

  3. 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…

  4. Robust Unsupervised Speaker Segmentation for Audio Diarization

    OpenAIRE

    Kadri, Hachem; Davy, Manuel; Ellouze, Noureddine

    2010-01-01

    Audio diarization is the process of partitioning an input audio stream into homogeneous regions according to their specific audio sources. These sources can include audio type (speech, music, background noise, ect.), speaker identity and channel characteristics. With the continually increasing number of larges volumes of spoken documents including broadcasts, voice mails, meetings and telephone conversations, diarization has received a great deal of interest in recent years which significantl...

  5. Spatial audio quality perception (part 1)

    DEFF Research Database (Denmark)

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

    2015-01-01

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

  6. Audio-Visual Aids: Historians in Blunderland.

    Science.gov (United States)

    Decarie, Graeme

    1988-01-01

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

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

    Science.gov (United States)

    Morand, J J

    1989-01-01

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

  8. Spatial audio quality perception (part 1)

    DEFF Research Database (Denmark)

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

    2015-01-01

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

  9. Design of an Audio Interface for Patmos

    OpenAIRE

    Ausin, Daniel Sanz; Goerge, Fabian

    2017-01-01

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

  10. Two-dimensional audio watermark for MPEG AAC audio

    Science.gov (United States)

    Tachibana, Ryuki

    2004-06-01

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

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

    Science.gov (United States)

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

    2001-05-01

    This paper describes a new interview data collection system that uses a laptop personal computer equipped with a touch-sensitive video monitor. The touch-screen-based audio computer-assisted self-interviewing system, or touch screen audio-CASI, enhances the ease of use of conventional audio CASI systems while simultaneously providing the privacy of self-administered questionnaires. We describe touch screen audio-CASI design features and operational characteristics. In addition, we present data from a recent clinic-based experiment indicating that the touch audio-CASI system is stable, robust, and suitable for administering relatively long and complex questionnaires on sensitive topics, including drug use and sexual behaviors associated with HIV and other sexually transmitted diseases.

  12. Haptic and Audio Interaction Design

    DEFF Research Database (Denmark)

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

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

    Directory of Open Access Journals (Sweden)

    Sunita V. Dhavale

    2014-02-01

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

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

    Directory of Open Access Journals (Sweden)

    Nitin Arora

    2010-06-01

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

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

    Science.gov (United States)

    Lee, Ki-Seung

    2008-08-01

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

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

    Science.gov (United States)

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

    2012-04-01

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

  17. 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.

  18. Review of AVS Audio Coding Standard

    Institute of Scientific and Technical Information of China (English)

    ZHANG Tao; ZHANG Caixia; ZHAO Xin

    2016-01-01

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

  19. Use of Cue Sheets in Audio Digitization

    Directory of Open Access Journals (Sweden)

    Austin Dixon

    2014-01-01

    Full Text Available Audio digitization is becoming essential to many libraries. As more and more audio files are being digitally preserved, the workflows for handling those digital objects need to be examined to ensure efficiency. In some instances, files are being manually manipulated when it would be more efficient to manipulate them programmatically. This article describes a time-saving solution to the problem of how to split master audio files into sub-item tracks.

  20. 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.

  1. Audio-visual Feature Fusion Person Identification Based on SVM and Score Normalization%基于SVM和归一化技术的音视频特征融合身份识别

    Institute of Scientific and Technical Information of China (English)

    丁辉; 安今朝

    2012-01-01

    In order to solve the problem of low recognition rate of face recognition and speech recognition under the wicked noise conditions. Based on the studies of feature level fusion theory and combined with Normalization and SVM theory, a novel model for face features and speech features fusion recognition is presented in this paper. First, we extract the face features and speech features correspondingly, then we fuse the two features on the feature level in order to obtain the fusion feature, after the calculation of the distance between the test people and template people we normalize the matching distance so as to reduce the computational and to improve the recognition accuracy. Al the last, we put the normalization matching distance into SVM can we obtain the recognition result. Trie experiment show that the fusion system performs well both in response time and system accuracy especially in noisy background.%针对噪声环境下人脸识别率和说话人识别率低的问题,在研究特征层融合的基础上,结合归一化技术和SVM理论,提出了一种融合人脸和语音的多生物特征识别模型.首先采用离散余弦变换和局部保持投影算法提取人脸特征及SVM方法提取语音特征,在特征层进行融合得到融合特征后,计算测试身份与模板问的距离,为了减少计算量和提高识别性能,对匹配距离进行归一化处理,最后输入到SVM进行识别.仿真结果表明,在噪声环境下,当信噪比降低时,融合识别率要明显高于单个系统的识别率,达到了身份识别的目的.

  2. Optimization of audio - ultrasonic plasma system parameters

    Science.gov (United States)

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

    2016-10-01

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

  3. Distortion Estimation in Compressed Music Using Only Audio Fingerprints

    NARCIS (Netherlands)

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

    2008-01-01

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

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

    NARCIS (Netherlands)

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

    2006-01-01

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

  5. A listening test system for automotive audio

    DEFF Research Database (Denmark)

    Christensen, Flemming; Geoff, Martin; Minnaar, Pauli;

    2005-01-01

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

  6. Enhancing Manual Scan Registration Using Audio Cues

    Science.gov (United States)

    Ntsoko, T.; Sithole, G.

    2014-04-01

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

  7. Audio-Visual Aids in Universities

    Science.gov (United States)

    Douglas, Jackie

    1970-01-01

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

  8. Digital Advances in Contemporary Audio Production.

    Science.gov (United States)

    Shields, Steven O.

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

  9. Audio coding in wireless acoustic sensor networks

    DEFF Research Database (Denmark)

    Zahedi, Adel; Østergaard, Jan; Jensen, Søren Holdt

    2015-01-01

    ) for the resulting remote DSC problem under covariance matrix distortion constraints. We further show that for this problem, the Gaussian source is the worst to code. Thus, the Gaussian RDF provides an upper bound to other sources such as audio signals. We then turn our attention to audio signals. We consider...

  10. The HDTV digital audio matrix

    Science.gov (United States)

    Mason, A. J.

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

  11. Stego-audio Using Genetic Algorithm Approach

    Directory of Open Access Journals (Sweden)

    V. Santhi

    2014-06-01

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

  12. 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)

  13. “Wrapping” X3DOM around Web Audio API

    Directory of Open Access Journals (Sweden)

    Andreas Stamoulias

    2015-12-01

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

  14. A Robust Zero-Watermarking Algorithm for Audio

    Directory of Open Access Journals (Sweden)

    Jie Zhu

    2008-03-01

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

  15. Audio Sensing Aid based Wireless Microphone Emulation Attacks Detection

    Directory of Open Access Journals (Sweden)

    Wang Shan-shan

    2013-10-01

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

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

    CERN Document Server

    Dangarwala, Kruti

    2010-01-01

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

  17. Content-based audio authentication using a hierarchical patchwork watermark embedding

    Science.gov (United States)

    Gulbis, Michael; Müller, Erika

    2010-05-01

    Content-based audio authentication watermarking techniques extract perceptual relevant audio features, which are robustly embedded into the audio file to protect. Manipulations of the audio file are detected on the basis of changes between the original embedded feature information and the anew extracted features during verification. The main challenges of content-based watermarking are on the one hand the identification of a suitable audio feature to distinguish between content preserving and malicious manipulations. On the other hand the development of a watermark, which is robust against content preserving modifications and able to carry the whole authentication information. The payload requirements are significantly higher compared to transaction watermarking or copyright protection. Finally, the watermark embedding should not influence the feature extraction to avoid false alarms. Current systems still lack a sufficient alignment of watermarking algorithm and feature extraction. In previous work we developed a content-based audio authentication watermarking approach. The feature is based on changes in DCT domain over time. A patchwork algorithm based watermark was used to embed multiple one bit watermarks. The embedding process uses the feature domain without inflicting distortions to the feature. The watermark payload is limited by the feature extraction, more precisely the critical bands. The payload is inverse proportional to segment duration of the audio file segmentation. Transparency behavior was analyzed in dependence of segment size and thus the watermark payload. At a segment duration of about 20 ms the transparency shows an optimum (measured in units of Objective Difference Grade). Transparency and/or robustness are fast decreased for working points beyond this area. Therefore, these working points are unsuitable to gain further payload, needed for the embedding of the whole authentication information. In this paper we present a hierarchical extension

  18. Authenticity examination of compressed audio recordings using detection of multiple compression and encoders' identification.

    Science.gov (United States)

    Korycki, Rafal

    2014-05-01

    Since the appearance of digital audio recordings, audio authentication has been becoming increasingly difficult. The currently available technologies and free editing software allow a forger to cut or paste any single word without audible artifacts. Nowadays, the only method referring to digital audio files commonly approved by forensic experts is the ENF criterion. It consists in fluctuation analysis of the mains frequency induced in electronic circuits of recording devices. Therefore, its effectiveness is strictly dependent on the presence of mains signal in the recording, which is a rare occurrence. Recently, much attention has been paid to authenticity analysis of compressed multimedia files and several solutions were proposed for detection of double compression in both digital video and digital audio. This paper addresses the problem of tampering detection in compressed audio files and discusses new methods that can be used for authenticity analysis of digital recordings. Presented approaches consist in evaluation of statistical features extracted from the MDCT coefficients as well as other parameters that may be obtained from compressed audio files. Calculated feature vectors are used for training selected machine learning algorithms. The detection of multiple compression covers up tampering activities as well as identification of traces of montage in digital audio recordings. To enhance the methods' robustness an encoder identification algorithm was developed and applied based on analysis of inherent parameters of compression. The effectiveness of tampering detection algorithms is tested on a predefined large music database consisting of nearly one million of compressed audio files. The influence of compression algorithms' parameters on the classification performance is discussed, based on the results of the current study.

  19. Performance enhancement for audio-visual speaker identification using dynamic facial muscle model.

    Science.gov (United States)

    Asadpour, Vahid; Towhidkhah, Farzad; Homayounpour, Mohammad Mehdi

    2006-10-01

    Science of human identification using physiological characteristics or biometry has been of great concern in security systems. However, robust multimodal identification systems based on audio-visual information has not been thoroughly investigated yet. Therefore, the aim of this work to propose a model-based feature extraction method which employs physiological characteristics of facial muscles producing lip movements. This approach adopts the intrinsic properties of muscles such as viscosity, elasticity, and mass which are extracted from the dynamic lip model. These parameters are exclusively dependent on the neuro-muscular properties of speaker; consequently, imitation of valid speakers could be reduced to a large extent. These parameters are applied to a hidden Markov model (HMM) audio-visual identification system. In this work, a combination of audio and video features has been employed by adopting a multistream pseudo-synchronized HMM training method. Noise robust audio features such as Mel-frequency cepstral coefficients (MFCC), spectral subtraction (SS), and relative spectra perceptual linear prediction (J-RASTA-PLP) have been used to evaluate the performance of the multimodal system once efficient audio feature extraction methods have been utilized. The superior performance of the proposed system is demonstrated on a large multispeaker database of continuously spoken digits, along with a sentence that is phonetically rich. To evaluate the robustness of algorithms, some experiments were performed on genetically identical twins. Furthermore, changes in speaker voice were simulated with drug inhalation tests. In 3 dB signal to noise ratio (SNR), the dynamic muscle model improved the identification rate of the audio-visual system from 91 to 98%. Results on identical twins revealed that there was an apparent improvement on the performance for the dynamic muscle model-based system, in which the identification rate of the audio-visual system was enhanced from 87

  20. Implementing Audio-CASI on Windows' Platforms.

    Science.gov (United States)

    Cooley, Philip C; Turner, Charles F

    1998-01-01

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

  1. 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

  2. Audio-visual voice activity detection

    Institute of Scientific and Technical Information of China (English)

    LIU Peng; WANG Zuo-ying

    2006-01-01

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

  3. Quantization of wavelet packet audio coding

    Institute of Scientific and Technical Information of China (English)

    Tan Jianguo; Zhang Wenjun; Liu Peilin

    2006-01-01

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

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

    Directory of Open Access Journals (Sweden)

    N. A. Haleem

    2016-09-01

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

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

    Directory of Open Access Journals (Sweden)

    Rachel N Denison

    2013-01-01

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

  6. The Effects of Audio-Visual Recorded and Audio Recorded Listening Tasks on the Accuracy of Iranian EFL Learners' Oral Production

    Science.gov (United States)

    Drood, Pooya; Asl, Hanieh Davatgari

    2016-01-01

    The ways in which task in classrooms has developed and proceeded have receive great attention in the field of language teaching and learning in the sense that they draw attention of learners to the competing features such as accuracy, fluency, and complexity. English audiovisual and audio recorded materials have been widely used by teachers and…

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

    OpenAIRE

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

    2003-01-01

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

  8. Audio watermarking for live performance

    Science.gov (United States)

    Tachibana, Ryuki

    2003-06-01

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

  9. Audio-visual affective expression recognition

    Science.gov (United States)

    Huang, Thomas S.; Zeng, Zhihong

    2007-11-01

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

  10. Presence and the utility of audio spatialization

    DEFF Research Database (Denmark)

    Bormann, Karsten

    2005-01-01

    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......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...... the playing radio would be stationary (a passive sound source) while subjects searched for some other object. Independent of this, the music emitted by the radio would be either fully spatialized or directional but nonattenuated. Findings include that for subjects searching for the active sound source, being...

  11. Spatial audio reproduction with primary ambient extraction

    CERN Document Server

    He, JianJun

    2017-01-01

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

  12. Definición de audio

    OpenAIRE

    Montañez, Luis A.; Cabrera, Juan G.

    2015-01-01

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

  13. 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.

  14. Web Audio/Video Streaming Tool

    Science.gov (United States)

    Guruvadoo, Eranna K.

    2003-01-01

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

  15. 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...

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

    Science.gov (United States)

    Beasley, Augie

    1995-01-01

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

  17. Audio Mining with emphasis on Music Genre Classification

    DEFF Research Database (Denmark)

    Meng, Anders

    2004-01-01

    in searching / retrieving audio effectively is needed. Currently, search engines such as e.g. Google, AltaVista etc. do not search into audio files, but uses either the textual information attached to the audio file or the textual information around the audio. Also in the hearing aid industries around...

  18. Implementation of Audio signal by using wavelet transform

    Directory of Open Access Journals (Sweden)

    Chakresh kumar,

    2010-10-01

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

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

    Directory of Open Access Journals (Sweden)

    Maxime eAmbard

    2015-10-01

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

  20. Audio description as an accessibility enhancer

    OpenAIRE

    Martins, Cláudia Susana Nunes

    2012-01-01

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

  1. PENGGUNAAN MEDIA AUDIO DALAM PEMBELAJARAN STENOGRAFI

    Directory of Open Access Journals (Sweden)

    S Martono

    2011-06-01

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

  2. PENGGUNAAN MEDIA AUDIO DALAM PEMBELAJARAN STENOGRAFI

    Directory of Open Access Journals (Sweden)

    S Martono

    2007-06-01

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

  3. Audio Video Compression Stream Synthesis and Implementation

    Institute of Scientific and Technical Information of China (English)

    徐燕凌; 方向忠; 周源华

    2004-01-01

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

  4. Design and implementation of an audio indicator

    Science.gov (United States)

    Zheng, Shiyong; Li, Zhao; Li, Biqing

    2017-04-01

    This page proposed an audio indicator which designed by using C9014, LED by operational amplifier level indicator, the decimal count/distributor of CD4017. The experimental can control audibly neon and holiday lights through the signal. Input audio signal after C9014 composed of operational amplifier for power amplifier, the adjust potentiometer extraction amplification signal input voltage CD4017 distributors make its drive to count, then connect the LED display running situation of the circuit. This simple audio indicator just use only U1 and can produce two colors LED with the audio signal tandem come pursuit of the running effect, from LED display the running of the situation takes can understand the general audio signal. The variation in the audio and the frequency of the signal and the corresponding level size. In this light can achieve jump to change, slowly, atlas, lighting four forms, used in home, hotel, discos, theater, advertising and other fields, and a wide range of USES, rU1h life in a modern society.

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

    OpenAIRE

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

    2015-01-01

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

  6. Audio Journal in an ELT Context

    Directory of Open Access Journals (Sweden)

    Neşe Aysin Siyli

    2012-09-01

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

  7. Unsupervised Learning of Structural Representation of Percussive Audio Using a Hierarchical Dirichlet Process Hidden Markov Model

    DEFF Research Database (Denmark)

    Antich, Jose Luis Diez; Paterna, Mattia; Marxer, Richard

    2016-01-01

    A method is proposed that extracts a structural representation of percussive audio in an unsupervised manner. It consists of two parts: 1) The input signal is segmented into blocks of approximately even duration, aligned to a metrical grid, using onset and timbre feature extraction, agglomerative...

  8. Determination of over current protection thresholds for class D audio amplifiers

    DEFF Research Database (Denmark)

    Nyboe, Flemming; Risbo, L; Andreani, Pietro

    2005-01-01

    Monolithic class-D audio amplifiers typically feature built-in over current protection circuitry that shuts down the amplifier in case of a short circuit on the output speaker terminals. To minimize cost, the threshold at which the device shuts down must be set just above the maximum current...

  9. 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...

  10. 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.

  11. 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.

  12. A High-Voltage Class D Audio Amplifier for Dielectric Elastomer Transducers

    DEFF Research Database (Denmark)

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

    2014-01-01

    Dielectric Elastomer (DE) transducers have emerged as a very interesting alternative to the traditional electrodynamic transducer. Lightweight, small size and high maneuverability are some of the key features of the DE transducer. An amplifier for the DE transducer suitable for audio applications...... is proposed and analyzed. The amplifier addresses the issue of a high impedance load, ensuring a linear response over the midrange region of the audio bandwidth (100 Hz – 3.5 kHz). THD+N below 0.1% are reported for the ± 300 V prototype amplifier producing a maximum of 125 Var at a peak efficiency of 95 %....

  13. Synchronization and comparison of Lifelog audio recordings

    DEFF Research Database (Denmark)

    Nielsen, Andreas Brinch; Hansen, Lars Kai

    2008-01-01

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

  14. Information Security using Audio Steganography -A Survey

    Directory of Open Access Journals (Sweden)

    B. Santhi

    2012-07-01

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

  15. Enhancing Navigation Skills through Audio Gaming.

    Science.gov (United States)

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

    2010-01-01

    We present the design, development and initial cognitive evaluation of an Audio-based Environment Simulator (AbES). This software allows a blind user to navigate through a virtual representation of a real space for the purposes of training orientation and mobility skills. Our findings indicate that users feel satisfied and self-confident when interacting with the audio-based interface, and the embedded sounds allow them to correctly orient themselves and navigate within the virtual world. Furthermore, users are able to transfer spatial information acquired through virtual interactions into real world navigation and problem solving tasks.

  16. 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.

  17. 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.

  18. 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.

  19. 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.

  20. 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.

  1. Music information retrieval in compressed audio files: a survey

    Science.gov (United States)

    Zampoglou, Markos; Malamos, Athanasios G.

    2014-07-01

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

  2. 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...

  3. 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...

  4. Audio Mining with emphasis on Music Genre Classification

    DEFF Research Database (Denmark)

    Meng, Anders

    2004-01-01

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

  5. Audio-visual materials usage preference among agricultural ...

    African Journals Online (AJOL)

    Audio-visual materials usage preference among agricultural extension workers in rivers ... AFRICAN JOURNALS ONLINE (AJOL) · Journals · Advanced Search · USING ... The use of audio-visual materials in the dissemination of agricultural ...

  6. Cross-modal retrieval of scripted speech audio

    Science.gov (United States)

    Owen, Charles B.; Makedon, Fillia

    1997-12-01

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

  7. Calibration of an audio frequency noise generator

    DEFF Research Database (Denmark)

    Diamond, Joseph M.

    1966-01-01

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

  8. Structuring Broadcast Audio for Information Access

    Directory of Open Access Journals (Sweden)

    Jean-Luc Gauvain

    2003-02-01

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

  9. Transparency benchmarking on audio watermarks and steganography

    Science.gov (United States)

    Kraetzer, Christian; Dittmann, Jana; Lang, Andreas

    2006-02-01

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

  10. Audio-visual integration in schizophrenia

    NARCIS (Netherlands)

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

    2003-01-01

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

  11. Audio-visual integration in schizophrenia.

    NARCIS (Netherlands)

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

    2003-01-01

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

  12. 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...

  13. 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...

  14. 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.

  15. Agency Video, Audio and Imagery Library

    Science.gov (United States)

    Grubbs, Rodney

    2015-01-01

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

  16. Building Digital Audio Preservation Infrastructure and Workflows

    Science.gov (United States)

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

    2010-01-01

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

  17. 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...

  18. Consuming audio: an introduction to Tweak Theory

    NARCIS (Netherlands)

    Perlman, Marc

    2014-01-01

    abstractAudio technology is a medium for music, and when we pay attention to it we tend to speculate about its effects on the music it transmits. By now there are well-established traditions of commentary (many of them critical) about the impact of musical reproduction on musical production.

  19. 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…

  20. 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.

  1. Relevant Research on Audio-Tutorial Methods

    Science.gov (United States)

    Novak, Joseph D.

    1970-01-01

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

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

    DEFF Research Database (Denmark)

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

    2014-01-01

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

  3. 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...

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

    OpenAIRE

    Pradeep Kumar Singh; R.K.Aggrawal

    2010-01-01

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

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

    DEFF Research Database (Denmark)

    Ljusev, Petar; Andersen, Michael Andreas E.

    2005-01-01

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

  6. Progressive Syntax-Rich Coding of Multichannel Audio Sources

    Directory of Open Access Journals (Sweden)

    Dai Yang

    2003-09-01

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

  7. 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...

  8. 47 CFR 73.403 - Digital audio broadcasting service requirements.

    Science.gov (United States)

    2010-10-01

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

  9. The KUSC Classical Music Dataset for Audio Key Finding

    Directory of Open Access Journals (Sweden)

    Ching-Hua Chuan

    2014-08-01

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

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

    Directory of Open Access Journals (Sweden)

    W. Bastiaan Kleijn

    2005-06-01

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

  11. 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.

  12. 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.

  13. Automatic Speech Segmentation Based On Audio and Optical Flow Visual Classification

    Directory of Open Access Journals (Sweden)

    Behnam Torabi

    2014-10-01

    Full Text Available Automatic speech segmentation as an important part of speech recognition system (ASR is highly noise dependent. Noise is made by changes in the communication channel, background, level of speaking etc. In recent years, many researchers have proposed noise cancelation techniques and have added visual features from speaker’s face to reduce the effect of noise on ASR systems. Removing noise from audio signals depends on the type of the noise; so it cannot be used as a general solution. Adding visual features improve this lack of efficiency, but advanced methods of this type need manual extraction of visual features. In this paper we propose a completely automatic system which uses optical flow vectors from speaker’s image sequence to obtain visual features. Then, Hidden Markov Models are trained to segment audio signals from image sequences and audio features based on extracted optical flow. The developed segmentation system based on such method acts totally automatic and become more robust to noise.

  14. Predicting the perception of performed dynamics in music audio with ensemble learning.

    Science.gov (United States)

    Elowsson, Anders; Friberg, Anders

    2017-03-01

    By varying the dynamics in a musical performance, the musician can convey structure and different expressions. Spectral properties of most musical instruments change in a complex way with the performed dynamics, but dedicated audio features for modeling the parameter are lacking. In this study, feature extraction methods were developed to capture relevant attributes related to spectral characteristics and spectral fluctuations, the latter through a sectional spectral flux. Previously, ground truths ratings of performed dynamics had been collected by asking listeners to rate how soft/loud the musicians played in a set of audio files. The ratings, averaged over subjects, were used to train three different machine learning models, using the audio features developed for the study as input. The highest result was produced from an ensemble of multilayer perceptrons with an R(2) of 0.84. This result seems to be close to the upper bound, given the estimated uncertainty of the ground truth data. The result is well above that of individual human listeners of the previous listening experiment, and on par with the performance achieved from the average rating of six listeners. Features were analyzed with a factorial design, which highlighted the importance of source separation in the feature extraction.

  15. Comparing audio and video data for rating communication.

    Science.gov (United States)

    Williams, Kristine; Herman, Ruth; Bontempo, Daniel

    2013-09-01

    Video recording has become increasingly popular in nursing research, adding rich nonverbal, contextual, and behavioral information. However, benefits of video over audio data have not been well established. We compared communication ratings of audio versus video data using the Emotional Tone Rating Scale. Twenty raters watched video clips of nursing care and rated staff communication on 12 descriptors that reflect dimensions of person-centered and controlling communication. Another group rated audio-only versions of the same clips. Interrater consistency was high within each group with Interclass Correlation Coefficient (ICC) (2,1) for audio .91, and video = .94. Interrater consistency for both groups combined was also high with ICC (2,1) for audio and video = .95. Communication ratings using audio and video data were highly correlated. The value of video being superior to audio-recorded data should be evaluated in designing studies evaluating nursing care.

  16. Audio Steganography Techniques-A Survey

    Directory of Open Access Journals (Sweden)

    Navneet Kaur

    2014-06-01

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

  17. Enlace optoelectrónico de audio

    OpenAIRE

    García Lozano, Jesús

    2012-01-01

    En este proyecto se diseña e implementa un sistema capaz de transmitir audio mediante luz infrarroja. Se pueden diferenciar dos grandes partes del proyecto, una el módulo emisor y la otra el módulo receptor. La señal es introducida en el módulo emisor a partir de cualquier reproductor de audio. Esta señal es sometida a un proceso de modulación FM para mejorar la comunicación entre emisor y receptor, puesto que la transmisión de la señal en banda base es más vulnerable a ruidos. Una vez modula...

  18. Indexing spoken audio by LSA and SOMs

    OpenAIRE

    2000-01-01

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

  19. Digitisation of the CERN Audio Archives

    CERN Multimedia

    Maximilien Brice

    2006-01-01

    Since the creation of CERN in 1954 until mid 1980s, the audiovisual service has recorded hundreds of hours of moments of life at CERN on audio tapes. These moments range from inaugurations of new facilities to VIP speeches and general interest cultural seminars The preservation process started in June 2005 On these pictures, we see Waltraud Hug working on an open-reel tape.

  20. Museum audio guides as an accessibility enhancer

    OpenAIRE

    Martins, Cláudia Susana Nunes

    2012-01-01

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

  1. PDE-SVD Based Audio Denoising

    OpenAIRE

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

    2012-01-01

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

  2. Capacity-optimized mp2 audio watermarking

    Science.gov (United States)

    Steinebach, Martin; Dittmann, Jana

    2003-06-01

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

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

    Directory of Open Access Journals (Sweden)

    Luca Cossettini

    2009-11-01

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

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

    DEFF Research Database (Denmark)

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

    2013-01-01

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

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

    DEFF Research Database (Denmark)

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

    2013-01-01

    Spatial audio solutions have been around for a long time in real-time applications, but yielding spatial cues that more closely simulate real life accuracy has been a computational issue, and has often been solved by hardware solutions. This has long been a restriction, but now with more powerful......-related transfer function (HRTF) system implemen- tation in a study in relation to precision, speed and navi- gational performance in localizing audio sources in a virtual environment. We found that a system using HRTFs is signif- icantly better at all three performance tasks than a system using panning....

  6. A new alley in Opinion Mining using Senti Audio Visual Algorithm

    Directory of Open Access Journals (Sweden)

    Mukesh Rawat,

    2016-02-01

    Full Text Available People share their views about products and services over social media, blogs, forums etc. If someone is willing to spend resources and money over these products and services will definitely learn about them from the past experiences of their peers. Opinion mining plays vital role in knowing increasing interests of a particular community, social and political events, making business strategies, marketing campaigns etc. This data is in unstructured form over internet but analyzed properly can be of great use. Sentiment analysis focuses on polarity detection of emotions like happy, sad or neutral. In this paper we proposed an algorithm i.e. Senti Audio Visual for examining Video as well as Audio sentiments. A review in the form of video/audio may contain several opinions/emotions, this algorithm will classify the reviews with the help of Baye’s Classifiers to three different classes i.e., positive, negative or neutral. The algorithm will use smiles, cries, gazes, pauses, pitch, and intensity as relevant Audio Visual features.

  7. SCALABLE PERCEPTUAL AUDIO REPRESENTATION WITH AN ADAPTIVE THREE TIME-SCALE SINUSOIDAL SIGNAL MODEL

    Institute of Scientific and Technical Information of China (English)

    Al-Moussawy Raed; Yin Junxun; Song Shaopeng

    2004-01-01

    This work is concerned with the development and optimization of a signal model for scalable perceptual audio coding at low bit rates. A complementary two-part signal model consisting of Sines plus Noise (SN) is described. The paper presents essentially a fundamental enhancement to the sinusoidal modeling component. The enhancement involves an audio signal scheme based on carrying out overlap-add sinusoidal modeling at three successive time scales,large, medium, and small. The sinusoidal modeling is done in an analysis-by-synthesis overlapadd manner across the three scales by using a psychoacoustically weighted matching pursuits.The sinusoidal modeling residual at the first scale is passed to the smaller scales to allow for the modeling of various signal features at appropriate resolutions. This approach greatly helps to correct the pre-echo inherent in the sinusoidal model. This improves the perceptual audio quality upon our previous work of sinusoidal modeling while using the same number of sinusoids. The most obvious application for the SN model is in scalable, high fidelity audio coding and signal modification.

  8. High-performance combination method of electric network frequency and phase for audio forgery detection in battery-powered devices.

    Science.gov (United States)

    Savari, Maryam; Abdul Wahab, Ainuddin Wahid; Anuar, Nor Badrul

    2016-09-01

    Audio forgery is any act of tampering, illegal copy and fake quality in the audio in a criminal way. In the last decade, there has been increasing attention to the audio forgery detection due to a significant increase in the number of forge in different type of audio. There are a number of methods for forgery detection, which electric network frequency (ENF) is one of the powerful methods in this area for forgery detection in terms of accuracy. In spite of suitable accuracy of ENF in a majority of plug-in powered devices, the weak accuracy of ENF in audio forgery detection for battery-powered devices, especially in laptop and mobile phone, can be consider as one of the main obstacles of the ENF. To solve the ENF problem in terms of accuracy in battery-powered devices, a combination method of ENF and phase feature is proposed. From experiment conducted, ENF alone give 50% and 60% accuracy for forgery detection in mobile phone and laptop respectively, while the proposed method shows 88% and 92% accuracy respectively, for forgery detection in battery-powered devices. The results lead to higher accuracy for forgery detection with the combination of ENF and phase feature.

  9. Content-Based Hierarchical Analysis of News Video Using Audio and Visual Information

    Institute of Scientific and Technical Information of China (English)

    2001-01-01

    A schema for content-based analysis of broadcast news video is presented. First, we separate commercials from news using audiovisual features. Then, we automatically organize news programs into a content hierarchy at various levels of abstraction via effective integration of video, audio, and text data available from the news programs. Based on these news video structure and content analysis technologies, a TV news video Library is generated, from which users can retrieve definite news story according to their demands.

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

    OpenAIRE

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

    2016-01-01

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

  11. Audio frequency in vivo optical coherence elastography

    Energy Technology Data Exchange (ETDEWEB)

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

    2009-05-21

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

  12. Non Audio-Video gesture recognition system

    DEFF Research Database (Denmark)

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

    2016-01-01

    Gesture recognition is a topic in computer science and language technology with the goal of interpreting human gestures via mathematical algorithms. Gestures can originate from any bodily motion or state but commonly originate from the face or hand. Current research focus includes on the emotion...... that can be connected to any computer on the market. The paper proposes an equation that relates the distance and voltage for a Sharp GP2Y0A21 and GP2D120 sensors in the situation that a hand is used as the reflective object. In the end, the presented system is compared with other audio/video system...

  13. On Steganography in Lost Audio Packets

    CERN Document Server

    Mazurczyk, Wojciech; Szczypiorski, Krzysztof

    2011-01-01

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

  14. Mixing audio concepts, practices and tools

    CERN Document Server

    Izhaki, Roey

    2013-01-01

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

  15. Audio marketing v ČR

    OpenAIRE

    Timanov, Vladimir

    2015-01-01

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

  16. Predistortion of a Bidirectional Cuk Audio Amplifier

    DEFF Research Database (Denmark)

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

    2014-01-01

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

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

    Science.gov (United States)

    Martin, R R; Haroldson, S K

    1992-06-01

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

  18. 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.

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

    Institute of Scientific and Technical Information of China (English)

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

    2004-01-01

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

  20. A content-based digital audio watermarking algorithm

    Science.gov (United States)

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

    2015-12-01

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

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

    Institute of Scientific and Technical Information of China (English)

    2011-01-01

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

  2. Origin, Development and Trend of Audio Book---Coping Strategies of Library in the Face of New Audio Resources%“听书”形态的起源、发展与趋势--兼论图书馆面对新型音频资源的应对策略

    Institute of Scientific and Technical Information of China (English)

    张鹏; 王铮

    2016-01-01

    在网络化、数字化、移动化的背景下,传统的听书形态发生了新的变化。文章首先回顾了听书形态的概念演变,分析了听书发展的历史、听书与载体的关系,以及听书的普及化、市场化、资源化特征,在此基础上分析了听书新形态所带来的音频资源变革,最后讨论了图书馆面对新型音频资源的应对策略。%Traditional audio book pattern has changed in the environment of Internet, digitization and mobile. This article reviews the conception evaluation of audio book, and then analyzes the history of audio book, the connection between audio book and record medi-um, and the feature of popularization, marketization and resource of audio book. Based on above research, the article analyzes the new pattern of audio book and the revolution on audio content resource, and discusses the strategy on new audio content resources for library.

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

    DEFF Research Database (Denmark)

    Dimitrov, Smilen; Serafin, Stefania

    2011-01-01

    Technology Devices International Ltd [FTDI] company) can be demonstrated to behave as a full-duplex, mono, 8-bit 44.1 kHz soundcard, through an implementation of: a PC audio driver for ALSA (Advanced Linux Sound Architecture); a matching program for the Arduino's ATmega microcontroller - and nothing more...

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

    Science.gov (United States)

    George, Rohini

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

  5. Semantic Labeling of Nonspeech Audio Clips

    Directory of Open Access Journals (Sweden)

    Xiaojuan Ma

    2010-01-01

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

  6. Simple Solutions for Space Station Audio Problems

    Science.gov (United States)

    Wood, Eric

    2016-01-01

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

  7. Person identification for mobile robot using audio-visual modality

    Science.gov (United States)

    Kim, Young-Ouk; Chin, Sehoon; Lee, Jihoon; Paik, Joonki

    2005-10-01

    Recently, we experienced significant advancement in intelligent service robots. The remarkable features of an intelligent robot include tracking and identification of person using biometric features. The human-robot interaction is very important because it is one of the final goals of an intelligent service robot. Many researches are concentrating in two fields. One is self navigation of a mobile robot and the other is human-robot interaction in natural environment. In this paper we will present an effective person identification method for HRI (Human Robot Interaction) using two different types of expert systems. However, most of mobile robots run under uncontrolled and complicated environment. It means that face and speech information can't be guaranteed under varying conditions, such as lighting, noisy sound, orientation of a robot. According to a value of illumination and signal to noise ratio around mobile a robot, our proposed fuzzy rule make a reasonable person identification result. Two embedded HMM (Hidden Marhov Model) are used for each visual and audio modality to identify person. The performance of our proposed system and experimental results are compared with single modality identification and simply mixed method of two modality.

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

    Institute of Scientific and Technical Information of China (English)

    XU Shuzheng; ZHANG Peng; WANG Pengjun; YANG Huazhong

    2009-01-01

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

  9. 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

  10. Circular microphone array for multi channel audio recording

    NARCIS (Netherlands)

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

    2004-01-01

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

  11. 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

  12. 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

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

    Science.gov (United States)

    McCrocklin, Shannon

    2012-01-01

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

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

    NARCIS (Netherlands)

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

    2008-01-01

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

  15. An Audio Stream Redirector for the Ethernet Speaker

    Science.gov (United States)

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

    2004-01-01

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

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

    Science.gov (United States)

    Baggaley, Jon; Klaas, Jim

    2006-01-01

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

  17. 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...

  18. DOA Estimation of Audio Sources in Reverberant Environments

    DEFF Research Database (Denmark)

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

    2016-01-01

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

  19. Using Audio Books to Improve Reading and Academic Performance

    Science.gov (United States)

    Montgomery, Joel R.

    2009-01-01

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

  20. 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

  1. 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...

  2. 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...

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

    NARCIS (Netherlands)

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

    2008-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Haider Ismael Shahadi

    2011-09-01

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

  5. A Dither Modulation Audio Watermarking Algorithm Based on HAS

    Directory of Open Access Journals (Sweden)

    Yi-bo Huang

    2012-11-01

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

  6. Effect of downsampling and compressive sensing on audio-based continuous cough monitoring.

    Science.gov (United States)

    Casaseca-de-la-Higuera, Pablo; Lesso, Paul; McKinstry, Brian; Pinnock, Hilary; Rabinovich, Roberto; McCloughan, Lucy; Monge-Álvarez, Jesús

    2015-01-01

    This paper presents an efficient cough detection system based on simple decision-tree classification of spectral features from a smartphone audio signal. Preliminary evaluation on voluntary coughs shows that the system can achieve 98% sensitivity and 97.13% specificity when the audio signal is sampled at full rate. With this baseline system, we study possible efficiency optimisations by evaluating the effect of downsampling below the Nyquist rate and how the system performance at low sampling frequencies can be improved by incorporating compressive sensing reconstruction schemes. Our results show that undersampling down to 400 Hz can still keep sensitivity and specificity values above 90% despite of aliasing. Furthermore, the sparsity of cough signals in the time domain allows keeping performance figures close to 90% when sampling at 100 Hz using compressive sensing schemes.

  7. 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.

  8. Transitioning from analog to digital audio recording in childhood speech sound disorders

    Science.gov (United States)

    Shriberg, Lawrence D.; McSweeny, Jane L.; Anderson, Bruce E.; Campbell, Thomas F.; Chial, Michael R.; Green, Jordan R.; Hauner, Katherina K.; Moore, Christopher A.; Rusiewicz, Heather L.; Wilson, David L.

    2014-01-01

    Few empirical findings or technical guidelines are available on the current transition from analog to digital audio recording in childhood speech sound disorders. Of particular concern in the present context was whether a transition from analog- to digital-based transcription and coding of prosody and voice features might require re-standardizing a reference database for research in childhood speech sound disorders. Two research transcribers with different levels of experience glossed, transcribed, and prosody-voice coded conversational speech samples from eight children with mild to severe speech disorders of unknown origin. The samples were recorded, stored, and played back using representative analog and digital audio systems. Effect sizes calculated for an array of analog versus digital comparisons ranged from negligible to medium, with a trend for participants’ speech competency scores to be slightly lower for samples obtained and transcribed using the digital system. We discuss the implications of these and other findings for research and clinical practise. PMID:16019779

  9. El tratamiento documental del mensaje audiovisual Documentary treatment of the audio-visual message

    Directory of Open Access Journals (Sweden)

    Blanca Rodríguez Bravo

    2005-06-01

    Full Text Available Se analizan las peculiaridades del documento audiovisual y el tratamiento documental que sufre en las emisoras de televisión. Observando a las particularidades de la imagen que condicionan su análisis y recuperación, se establecen las etapas y procedimientos para representar el mensaje audiovisual con vistas a su reutilización. Por último se realizan algunas consideraciones acerca del procesamiento automático del video y de los cambios introducidos por la televisión digital.Peculiarities of the audio-visual document and the treatment it undergoes in TV broadcasting stations are analyzed. The particular features of images condition their analysis and recovery; this paper establishes stages and proceedings for the representation of audio-visual messages with a view to their re-usability Also, some considerations about the automatic processing of the video and the changes introduced by digital TV are made.

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

    Directory of Open Access Journals (Sweden)

    Beracoechea JA

    2006-01-01

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

  11. 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.

  12. Enhanced Audio LSB Steganography for Secure Communication

    Directory of Open Access Journals (Sweden)

    Muhammad Junaid Hussain

    2016-01-01

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

  13. Particle Filtering on the Audio Localization Manifold

    CERN Document Server

    Ettinger, Evan

    2010-01-01

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

  14. 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.

  15. A direct broadcast satellite-audio experiment

    Science.gov (United States)

    Vaisnys, Arvydas; Abbe, Brian; Motamedi, Masoud

    1992-03-01

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

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

    NARCIS (Netherlands)

    Pocta, P.; Beerends, J.G.

    2015-01-01

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

  17. 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…

  18. 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…

  19. Beyond podcasting: creative approaches to designing educational audio

    Directory of Open Access Journals (Sweden)

    Andrew Middleton

    2009-12-01

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

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

    Science.gov (United States)

    Koenig, Bruce E; Lacey, Douglas S

    2012-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Saadia Zahid

    2015-01-01

    Full Text Available Audio segmentation is a basis for multimedia content analysis which is the most important and widely used application nowadays. An optimized audio classification and segmentation algorithm is presented in this paper that segments a superimposed audio stream on the basis of its content into four main audio types: pure-speech, music, environment sound, and silence. An algorithm is proposed that preserves important audio content and reduces the misclassification rate without using large amount of training data, which handles noise and is suitable for use for real-time applications. Noise in an audio stream is segmented out as environment sound. A hybrid classification approach is used, bagged support vector machines (SVMs with artificial neural networks (ANNs. Audio stream is classified, firstly, into speech and nonspeech segment by using bagged support vector machines; nonspeech segment is further classified into music and environment sound by using artificial neural networks and lastly, speech segment is classified into silence and pure-speech segments on the basis of rule-based classifier. Minimum data is used for training classifier; ensemble methods are used for minimizing misclassification rate and approximately 98% accurate segments are obtained. A fast and efficient algorithm is designed that can be used with real-time multimedia applications.

  2. Sampling Function of Degree 2 for DVD-Audio

    Science.gov (United States)

    Toraichi, Kazuo; Nakamura, Koji

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

  3. 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...

  4. Robust message authentication code algorithm for digital audio recordings

    Science.gov (United States)

    Zmudzinski, Sascha; Steinebach, Martin

    2007-02-01

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

  5. Aurally Aided Visual Search Performance Comparing Virtual Audio Systems

    DEFF Research Database (Denmark)

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

    2014-01-01

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

  6. 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...... for performance and out of band spectral amplitudes. The basic principle in MCM is to use programmable logic to combine two or more Pulse Width Modulated (PWM) audio signals at different switching frequencies. In this way the out of band spectrum will be lowered compared with conventional class D amplifiers...

  7. Lattice Vector Quantization Applied to Speech and Audio Coding

    Institute of Scientific and Technical Information of China (English)

    Minjie Xie

    2012-01-01

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

  8. 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

  9. A Review on Audio-visual Translation Studies

    Institute of Scientific and Technical Information of China (English)

    李瑶

    2008-01-01

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

  10. 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.

  11. Aurally Aided Visual Search Performance Comparing Virtual Audio Systems

    DEFF Research Database (Denmark)

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

    2014-01-01

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

  12. 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 %....

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

    DEFF Research Database (Denmark)

    Ljusev, Petar; Andersen, Michael Andreas E.

    2005-01-01

    This paper presents a new class of switching-mode audio power amplifiers, which are capable of direct energy conversion from the AC mains to the audio output. They represent an ultimate integration of a switching-mode power supply and a Class D audio power amplifier, where the intermediate DC bus...... has been replaced with a high frequency AC link. When compared to the conventional Class D amplifiers with a separate DC power supply, the proposed single conversion stage amplifier provides simple and compact solution with better efficiency and higher level of integration, leading to reduced...

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

    Directory of Open Access Journals (Sweden)

    Frank Kurth

    2007-01-01

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

  15. Semi-fragile Audio Watermarking Scheme Based on the Approximate Components Energy%基于近似分量能量的半脆弱音频水印算法

    Institute of Scientific and Technical Information of China (English)

    宁超魁; 和红杰; 陈帆; 尹忠科

    2013-01-01

    为提高半脆弱音频水印算法的安全性,本文提出一种基于近似分量能量的半脆弱音频水印算法.该算法将每个音频帧分为两段,分别用于提取音频帧特征和嵌入其它音频帧的水印信息.本文利用音频段近似分量能量以α为底的对数作为音频帧特征,基于密钥将音频帧特征加密后随机嵌入到其他音频帧另一段的混合域中,检测时根据音频帧及其相邻帧水印的不一致性判断音频帧的真实性.兼顾音频帧特征的鲁棒性和分布特性讨论α的取值,实验结果表明该算法能准确定位被篡改的音频帧且能有效抵抗拼贴攻击.%In order to improve the security of the semi-fragile audio watermarking scheme, the semi-fragile audio watermarking algorithm based on the approximate components energy(ACE) was proposed. Every audio frame was divided into two sections. One section was used to extract the feature of the audio frame, and the other was used to hide the watermark data of the other audio frame. The feature of an audio frame was the logarithm of ACE of the chosen audio section form the audio frame to the base α. For each audio frame, the feature was encrypted and randomly embedded in the hybrid domain of another audio frame based on the secret key. The validity of an audio frame was determined by the inconsistency of itself and its neighborhood audio frames. This paper also discussed the value of α from the viewpoint of the robustness and distribution of the audio frame feature. Experimental results demonstrate that the proposed scheme can localize the tampered regions accurately and resist collage attacks effectively.

  16. Audio-Visual Integration of Emotional Information

    Directory of Open Access Journals (Sweden)

    Penny Bergman

    2011-10-01

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

  17. Analysis of musical expression in audio signals

    Science.gov (United States)

    Dixon, Simon

    2003-01-01

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

  18. Multi-Level Audio Classification Architecture

    Directory of Open Access Journals (Sweden)

    Jozef Vavrek

    2015-01-01

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

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

    DEFF Research Database (Denmark)

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

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

  20. Perancangan Sistem Audio Mobil berbasiskan Sistem Pakar dan Web

    Directory of Open Access Journals (Sweden)

    Djunaidy Santoso

    2011-11-01

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

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

    NARCIS (Netherlands)

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

    2004-01-01

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

  2. Audio CAPTCHA for SIP-Based VoIP

    Science.gov (United States)

    Soupionis, Yannis; Tountas, George; Gritzalis, Dimitris

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

  3. 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. 

  4. Reception of infrasound and audio current in derma nerves

    Institute of Scientific and Technical Information of China (English)

    Jianwen Li; Ziyu Li; Xuezong Ma

    2010-01-01

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

  5. Proper Use of Audio-Visual Aids: Essential for Educators.

    Science.gov (United States)

    Dejardin, Conrad

    1989-01-01

    Criticizes educators as the worst users of audio-visual aids and among the worst public speakers. Offers guidelines for the proper use of an overhead projector and the development of transparencies. (DMM)

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

    NARCIS (Netherlands)

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

    2009-01-01

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

  7. 37 CFR 201.27 - Initial notice of distribution of digital audio recording devices or media.

    Science.gov (United States)

    2010-07-01

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

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

    DEFF Research Database (Denmark)

    Dimitrov, Smilen; Serafin, Stefania

    2011-01-01

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

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

    Institute of Scientific and Technical Information of China (English)

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

    2005-01-01

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

  10. Automatic summarization of audio-visual soccer feeds

    OpenAIRE

    Chen F; De Vleeschouwer C; Duxans Barrobes H.; Gregorio Escalada J.; Conejero D.

    2010-01-01

    This paper presents a fully automatic system for soccer game summarization. The system takes audio-visual content as an input, and builds on the integration of two independent but complementary contributions (i) to identify crucial periods of the soccer game in a fully automatic way, and (ii) to summarize the soccer game as a function of individual narrative preferences of the user. The process involves both audio and video analysis, and handles the personalized summarization challenge as a r...

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

    Science.gov (United States)

    Sánchez, J; Lumbreras, M

    1999-01-01

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

  12. 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.

  13. Ferrite bead effect on Class-D amplifier audio quality

    OpenAIRE

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

    2014-01-01

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

  14. 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.

  15. Revisiting the problem of audio-based hit song prediction using convolutional neural networks

    OpenAIRE

    Yang, Li-Chia; Chou, Szu-Yu; Liu, Jen-Yu; Yang, Yi-Hsuan; Chen, Yi-an

    2017-01-01

    Being able to predict whether a song can be a hit has impor- tant applications in the music industry. Although it is true that the popularity of a song can be greatly affected by exter- nal factors such as social and commercial influences, to which degree audio features computed from musical signals (whom we regard as internal factors) can predict song popularity is an interesting research question on its own. Motivated by the recent success of deep learning techniques, we attempt to ex- tend...

  16. 特定类型音频流泛化识别方法%A Generic Method of Recognizing Specific Type Audio Stream

    Institute of Scientific and Technical Information of China (English)

    罗森林; 李金玉; 潘丽敏

    2011-01-01

    提出一种基于Mel频率倒谱系数(MFCC)和AdaBoost算法的特定类型音频流泛化识别方法,通过分析特定类型音频流的子类别间的共性和差异性,利用共性特征进行泛化识别,能够准确地检测并定位音频流中特定类型的音频.文中将枪声作为特定类型音频进行研究,通过提取各种枪声子类别的共性,弱化子类间的差异得到一个泛化的枪声模板,利用一个模板就可以支持多子类的准确识别.实验结果表明,算法的识别准确率为87.6%,查全率达到91.8%.%To meet the security demand of audio information, a generic method of recognizing specific type audio stream based on MFCC and AdaBoost is proposed in this paper, which can detect and locate the specific audio fragment from the audio stream accurately. The generality and differences between subcategories of the audio stream was analyzed to achieve the generic recognition. Multi-type gunshot was considered as the specific type of audio stream. The generic template was obtained by extracting the common features and reducing the different features of the gunshot audio, which could support the accurate identification of multiple sub-classes. The experiments show that the recognition accuracy of the proposed method is 87. 6% and the recall rate reaches 91. 8%.

  17. Digital Audio Radio Broadcast Systems Laboratory Testing Nearly Complete

    Science.gov (United States)

    2005-01-01

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

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

    Science.gov (United States)

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

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

  19. The Fungible Audio-Visual Mapping and its Experience

    Directory of Open Access Journals (Sweden)

    Adriana Sa

    2014-12-01

    Full Text Available This article draws a perceptual approach to audio-visual mapping. Clearly perceivable cause and effect relationships can be problematic if one desires the audience to experience the music. Indeed perception would bias those sonic qualities that fit previous concepts of causation, subordinating other sonic qualities, which may form the relations between the sounds themselves. The question is, how can an audio-visual mapping produce a sense of causation, and simultaneously confound the actual cause-effect relationships. We call this a fungible audio-visual mapping. Our aim here is to glean its constitution and aspect. We will report a study, which draws upon methods from experimental psychology to inform audio-visual instrument design and composition. The participants are shown several audio-visual mapping prototypes, after which we pose quantitative and qualitative questions regarding their sense of causation, and their sense of understanding the cause-effect relationships. The study shows that a fungible mapping requires both synchronized and seemingly non-related components – sufficient complexity to be confusing. As the specific cause-effect concepts remain inconclusive, the sense of causation embraces the whole. 

  20. Lossless Audio Watermarking Based on the Alpha Statistic Modulation

    Directory of Open Access Journals (Sweden)

    Sunita V. Dhavale

    2012-09-01

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

  1. Joint application of audio spectral envelope and tonality index in an e-asthma monitoring system.

    Science.gov (United States)

    Wiśniewski, Marcin; Zieliński, Tomasz P

    2015-05-01

    This paper presents in detail a recently introduced highly efficient method for automatic detection of asthmatic wheezing in breathing sounds. The fluctuation in the audio spectral envelope (ASE) from the MPEG-7 standard and the value of the tonality index (TI) from the MPEG-2 Audio specification are jointly used as discriminative features for wheezy sounds, while the support vector machine (SVM) with a polynomial kernel serves as a classifier. The advantages of the proposed approach are described in the paper (e.g., detecting weak wheezes, very good ROC characteristics, independence from noise color). Since the method is not computationally complex, it is suitable for remote asthma monitoring using mobile devices (personal medical assistants). The main contribution of this paper consists of presenting all the implementation details concerning the proposed approach for the first time, i.e., the pseudocode of the method and adjusting the values of the ASE and TI parameters after which only one (not two) FFT is required for analysis of a next overlapping signal fragment. The efficiency of the method has also been additionally confirmed by the AdaBoost classifier with a built-in mechanism to feature ranking, as well as a previously performed minimal-redundancy-maximal-relevance test.

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

    Directory of Open Access Journals (Sweden)

    Cheng Ji-Shiung

    2003-01-01

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

  3. Say What? The Role of Audio in Multimedia Video

    Science.gov (United States)

    Linder, C. A.; Holmes, R. M.

    2011-12-01

    Audio, including interviews, ambient sounds, and music, is a critical-yet often overlooked-part of an effective multimedia video. In February 2010, Linder joined scientists working on the Global Rivers Observatory Project for two weeks of intensive fieldwork in the Congo River watershed. The team's goal was to learn more about how climate change and deforestation are impacting the river system and coastal ocean. Using stills and video shot with a lightweight digital SLR outfit and audio recorded with a pocket-sized sound recorder, Linder documented the trials and triumphs of working in the heart of Africa. Using excerpts from the six-minute Congo multimedia video, this presentation will illustrate how to record and edit an engaging audio track. Topics include interview technique, collecting ambient sounds, choosing and using music, and editing it all together to educate and entertain the viewer.

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

    DEFF Research Database (Denmark)

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

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

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

    Directory of Open Access Journals (Sweden)

    Amir Reza Fazli

    2012-06-01

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

  6. A novel audio watermarking scheme using multiscale wavelet modulation

    Institute of Scientific and Technical Information of China (English)

    JI Bing; ZHANG De; JI Xiaoyong

    2004-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Jim Rudolph

    2004-08-01

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

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

    Directory of Open Access Journals (Sweden)

    Muhammad Shoaib

    2016-10-01

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

  9. 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...

  10. Bimodal audio-visual training enhances auditory adaptation process.

    Science.gov (United States)

    Kawase, Tetsuaki; Sakamoto, Shuichi; Hori, Yoko; Maki, Atsuko; Suzuki, Yôiti; Kobayashi, Toshimitsu

    2009-09-23

    Effects of auditory training with bimodal audio-visual stimuli on monomodal aural speech intelligibility were examined in individuals with normal hearing using highly degraded noise-vocoded speech sound. Visual cue simultaneously presented with auditory stimuli during the training session significantly improved auditory speech intelligibility not only for words used in the training session, but also untrained words, when compared with the auditory training using only auditory stimuli. Visual information is generally considered to complement insufficient speech information conveyed by the auditory system during audio-visual speech perception. However, the present results showed another beneficial effect of audio-visual training that the visual cue enhances the auditory adaptation process to the degraded new speech sound, which is different from those given during bimodal training.

  11. Audio system using binaural synthesis for multimodal telepresence applications

    DEFF Research Database (Denmark)

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

    2013-01-01

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

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

    DEFF Research Database (Denmark)

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

    2008-01-01

    While switch-mode audio power amplifiers allow compact implementations and high output power levels due to their high power efficiency, they are very well known for creating electromagnetic interference (EMI) with other electronic equipment, in particular radio receivers. Lowering the EMI of switch......-mode audio power amplifiers while keeping the performance measures to excellent levels is therefore of high general interest. A modulator utilizing multiple carrier signals to generate a two level pulse train will be shown in this paper. The performance of the modulator will be compared in simulation...... to existing modulation topologies. The lower EMI as well as the preserved audio performance will be shown in simulation as well as in measurement results on a prototype....

  13. Efficiency Optimization in Class-D Audio Amplifiers

    DEFF Research Database (Denmark)

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

    2015-01-01

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

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

    CERN Document Server

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

    2013-01-01

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

  15. Optimizing dictionary learning parameters for solving Audio Inpainting problem

    Directory of Open Access Journals (Sweden)

    Václav Mach

    2013-01-01

    Full Text Available Recovering missing or distorted audio signal sam-ples has been recently improved by solving an Audio Inpaintingproblem. This paper aims to connect this problem with K-SVD dictionary learning to improve reconstruction error formissing signal insertion problem. Our aim is to adapt an initialdictionary to the reliable signal to be more accurate in missingsamples estimation. This approach is based on sparse signalsreconstruction and optimization problem. In the paper two staplealgorithms, connection between them and emerging problemsare described. We tried to find optimal parameters for efficientdictionary learning.

  16. 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....

  17. Design of a WAV audio player based on K20

    Directory of Open Access Journals (Sweden)

    Xu Yu

    2016-01-01

    Full Text Available The designed player uses the Freescale Company’s MK20DX128VLH7 as the core control ship, and its hardware platform is equipped with VS1003 audio decoder, OLED display interface, USB interface and SD card slot. The player uses the open source embedded real-time operating system μC/OS-II, Freescale USB Stack V4.1.1 and FATFS, and a graphical user interface is developed to improve the user experience based on CGUI. In general, the designed WAV audio player has a strong applicability and a good practical value.

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

    Science.gov (United States)

    Steinebach, Martin; Zmudzinski, Sascha

    2008-02-01

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

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

    DEFF Research Database (Denmark)

    Knott, Arnold; Ploug, Rasmus Overgaard

    2012-01-01

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

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

    DEFF Research Database (Denmark)

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

    2014-01-01

    It was recently shown that delta-sigma quantization (DSQ) can be used for optimal multiple description (MD) coding of Gaussian sources. The DSQ scheme combined oversampling, prediction, and noise-shaping in order to trade off side distortion for central distortion in MD coding. It was shown......, it is possible to obtain good quality audio in the presence of packet losses. Simulations on real audio reveal that, contrary to existing designs, it is straightforward to obtain a large number of trade-off points between side distortion and central distortion, which makes the proposed coder suitable for a wide...

  1. 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

  2. Audio steganalysis based on "negative resonance phenomenon" caused by steganographic tools

    Institute of Scientific and Technical Information of China (English)

    2006-01-01

    Researching on the impact different steganographic software tools have audio statistical features, revealed the phenomenon that when messages are embedded in a WAV file by using a certain tool, the variation of statistical features in the WAV file which already contains messages embedded by the same tool is abruptly smaller than those in which messages have not been embedded. We call it "negative resonance phenomenon" temporarily. With the phenomenon above and Support Vector Machines (SVMs), we can detect the existence of hidden messages, and also identify the tools used to hide them. As shown by the experimental results, the proposed method can be very effectively used to detect hidden messages embedded by Hide4PGP, Stegowav and S-Tools4.

  3. Cover signal specific steganalysis: the impact of training on the example of two selected audio steganalysis approaches

    Science.gov (United States)

    Kraetzer, Christian; Dittmann, Jana

    2008-02-01

    The main goals of this paper are to show the impact of the basic assumptions for the cover channel characteristics as well as the impact of different training/testing set generation strategies on the statistical detectability of exemplary chosen audio hiding approaches known from steganography and watermarking. Here we have selected exemplary five steganography algorithms and four watermarking algorithms. The channel characteristics for two different chosen audio cover channels (an application specific exemplary scenario of VoIP steganography and universal audio steganography) are formalised and their impact on decisions in the steganalysis process, especially on the strategies applied for training/ testing set generation, are shown. Following the assumptions on the cover channel characteristics either cover dependent or cover independent training and testing can be performed, using either correlated or non-correlated training and test sets. In comparison to previous work, additional frequency domain features are introduced for steganalysis and the performance (in terms of classification accuracy) of Bayesian classifiers and multinomial logistic regression models is compared with the results of SVM classification. We show that the newly implemented frequency domain features increase the classification accuracy achieved in SVM classification. Furthermore it is shown on the example of VoIP steganalysis that channel character specific evaluation performs better than tests without focus on a specific channel (i.e. universal steganalysis). A comparison of test results for cover dependent and independent training and testing shows that the latter performs better for all nine algorithms evaluated here and the used SVM based classifier.

  4. Multi-modal gesture recognition using integrated model of motion, audio and video

    Science.gov (United States)

    Goutsu, Yusuke; Kobayashi, Takaki; Obara, Junya; Kusajima, Ikuo; Takeichi, Kazunari; Takano, Wataru; Nakamura, Yoshihiko

    2015-07-01

    Gesture recognition is used in many practical applications such as human-robot interaction, medical rehabilitation and sign language. With increasing motion sensor development, multiple data sources have become available, which leads to the rise of multi-modal gesture recognition. Since our previous approach to gesture recognition depends on a unimodal system, it is difficult to classify similar motion patterns. In order to solve this problem, a novel approach which integrates motion, audio and video models is proposed by using dataset captured by Kinect. The proposed system can recognize observed gestures by using three models. Recognition results of three models are integrated by using the proposed framework and the output becomes the final result. The motion and audio models are learned by using Hidden Markov Model. Random Forest which is the video classifier is used to learn the video model. In the experiments to test the performances of the proposed system, the motion and audio models most suitable for gesture recognition are chosen by varying feature vectors and learning methods. Additionally, the unimodal and multi-modal models are compared with respect to recognition accuracy. All the experiments are conducted on dataset provided by the competition organizer of MMGRC, which is a workshop for Multi-Modal Gesture Recognition Challenge. The comparison results show that the multi-modal model composed of three models scores the highest recognition rate. This improvement of recognition accuracy means that the complementary relationship among three models improves the accuracy of gesture recognition. The proposed system provides the application technology to understand human actions of daily life more precisely.

  5. Fiber-channel audio video standard for military and commercial aircraft product lines

    Science.gov (United States)

    Keller, Jack E.

    2002-08-01

    Fibre channel is an emerging high-speed digital network technology that combines to make inroads into the avionics arena. The suitability of fibre channel for such applications is largely due to its flexibility in these key areas: Network topologies can be configured in point-to-point, arbitrated loop or switched fabric connections. The physical layer supports either copper or fiber optic implementations with a Bit Error Rate of less than 10-12. Multiple Classes of Service are available. Multiple Upper Level Protocols are supported. Multiple high speed data rates offer open ended growth paths providing speed negotiation within a single network. Current speeds supported by commercially available hardware are 1 and 2 Gbps providing effective data rates of 100 and 200 MBps respectively. Such networks lend themselves well to the transport of digital video and audio data. This paper summarizes an ANSI standard currently in the final approval cycle of the InterNational Committee for Information Technology Standardization (INCITS). This standard defines a flexible mechanism whereby digital video, audio and ancillary data are systematically packaged for transport over a fibre channel network. The basic mechanism, called a container, houses audio and video content functionally grouped as elements of the container called objects. Featured in this paper is a specific container mapping called Simple Parametric Digital Video (SPDV) developed particularly to address digital video in avionics systems. SPDV provides pixel-based video with associated ancillary data typically sourced by various sensors to be processed and/or distributed in the cockpit for presentation via high-resolution displays. Also highlighted in this paper is a streamlined Upper Level Protocol (ULP) called Frame Header Control Procedure (FHCP) targeted for avionics systems where the functionality of a more complex ULP is not required.

  6. Multi-modal Gesture Recognition using Integrated Model of Motion, Audio and Video

    Institute of Scientific and Technical Information of China (English)

    GOUTSU Yusuke; KOBAYASHI Takaki; OBARA Junya; KUSAJIMAIkuo; TAKEICHI Kazunari; TAKANO Wataru; NAKAMURA Yoshihiko

    2015-01-01

    Gesture recognition is used in many practical applications such as human-robot interaction, medical rehabilitation and sign language. With increasing motion sensor development, multiple data sources have become available, which leads to the rise of multi-modal gesture recognition. Since our previous approach to gesture recognition depends on a unimodal system, it is difficult to classify similar motion patterns. In order to solve this problem, a novel approach which integrates motion, audio and video models is proposed by using dataset captured by Kinect. The proposed system can recognize observed gestures by using three models. Recognition results of three models are integrated by using the proposed framework and the output becomes the final result. The motion and audio models are learned by using Hidden Markov Model. Random Forest which is the video classifier is used to learn the video model. In the experiments to test the performances of the proposed system, the motion and audio models most suitable for gesture recognition are chosen by varying feature vectors and learning methods. Additionally, the unimodal and multi-modal models are compared with respect to recognition accuracy. All the experiments are conducted on dataset provided by the competition organizer of MMGRC, which is a workshop for Multi-Modal Gesture Recognition Challenge. The comparison results show that the multi-modal model composed of three models scores the highest recognition rate. This improvement of recognition accuracy means that the complementary relationship among three models improves the accuracy of gesture recognition. The proposed system provides the application technology to understand human actions of daily life more precisely.

  7. Deep Complementary Bottleneck Features for Visual Speech Recognition

    NARCIS (Netherlands)

    Petridis, Stavros; Pantic, Maja

    2016-01-01

    Deep bottleneck features (DBNFs) have been used successfully in the past for acoustic speech recognition from audio. However, research on extracting DBNFs for visual speech recognition is very limited. In this work, we present an approach to extract deep bottleneck visual features based on deep auto

  8. Deep Complementary Bottleneck Features for Visual Speech Recognition

    NARCIS (Netherlands)

    Petridis, Stavros; Pantic, Maja

    Deep bottleneck features (DBNFs) have been used successfully in the past for acoustic speech recognition from audio. However, research on extracting DBNFs for visual speech recognition is very limited. In this work, we present an approach to extract deep bottleneck visual features based on deep

  9. Audio Quality Assurance : An Application of Cross Correlation

    DEFF Research Database (Denmark)

    Jurik, Bolette Ammitzbøll; Nielsen, Jesper Asbjørn Sindahl

    2012-01-01

    We describe algorithms for automated quality assurance on content of audio files in context of preservation actions and access. The algorithms use cross correlation to compare the sound waves. They are used to do overlap analysis in an access scenario, where preserved radio broadcasts are used in...

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

    Science.gov (United States)

    Wormnaes, Siri; Sellaeg, Nina

    2013-01-01

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

  11. 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,...

  12. Reading's SLiCK with New Audio Texts and Strategies.

    Science.gov (United States)

    Boyle, Elizabeth A.; Washburn, Shari Gallin; Rosenberg, Michael S.; Connelly, Vincent J.; Brinckerhoff, Loring C.; Banerjee, Manju

    2002-01-01

    This article discusses challenges for secondary students with disabilities and alternative instructional methods that teachers of students with poor reading skills can use to convey content information effectively and efficiently. The use of audio textbooks on CD-ROMs is emphasized and the SLiCK strategy is explained as a support for the CD-ROM.…

  13. 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.

  14. Market potential for interactive audio-visual media

    NARCIS (Netherlands)

    Leurdijk, A.; Limonard, S.

    2005-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Martin Žvelc

    2011-01-01

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

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

    Science.gov (United States)

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

    2014-05-28

    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.

  17. Output impedance and stability of audio power amplifiers

    NARCIS (Netherlands)

    Schaink, T.

    2006-01-01

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

  18. An Audio-Visual Lecture Course in Russian Culture

    Science.gov (United States)

    Leighton, Lauren G.

    1977-01-01

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

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

    DEFF Research Database (Denmark)

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

    2008-01-01

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

  20. Market potential for interactive audio-visual media

    NARCIS (Netherlands)

    Leurdijk, A.; Limonard, S.

    2005-01-01

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

  1. Audio and Video Reflections to Promote Social Justice

    Science.gov (United States)

    Boske, Christa

    2011-01-01

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

  2. Comparative study of Audio-lingual method and CLT

    Institute of Scientific and Technical Information of China (English)

    2013-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Shuixian Chen

    2010-01-01

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

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

    DEFF Research Database (Denmark)

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

    2013-01-01

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

  5. Making Audio-Visual Teaching Materials for Elementary Science

    OpenAIRE

    永田, 四郎

    1980-01-01

    For the elementary science, some audio-visual teaching materials were made by author and our students. These materials are slides for projector, transparencies and materials for OHP, 8 mm sound films and video tapes. We hope this kind of study will continue.

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

    NARCIS (Netherlands)

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

    2006-01-01

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

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

    Science.gov (United States)

    Szarkowska, Agnieszka

    2013-01-01

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

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

    Science.gov (United States)

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

    2016-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Chen Shuixian

    2010-01-01

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

  10. 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.

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

    Directory of Open Access Journals (Sweden)

    Petr Motlicek

    2013-01-01

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

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

    NARCIS (Netherlands)

    van der Zee, Ronan A.R.; van Tuijl, Adrianus Johannes Maria

    1998-01-01

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

  13. 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...

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

    Energy Technology Data Exchange (ETDEWEB)

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

    1999-07-01

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

  15. Real-time Loudspeaker Distance Estimation with Stereo Audio

    DEFF Research Database (Denmark)

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

    2015-01-01

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

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

    NARCIS (Netherlands)

    van der Zee, Ronan A.R.; van Tuijl, Adrianus Johannes Maria

    1998-01-01

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

  17. SPEECH/MUSIC CLASSIFICATION USING WAVELET BASED FEATURE EXTRACTION TECHNIQUES

    Directory of Open Access Journals (Sweden)

    Thiruvengatanadhan Ramalingam

    2014-01-01

    Full Text Available Audio classification serves as the fundamental step towards the rapid growth in audio data volume. Due to the increasing size of the multimedia sources speech and music classification is one of the most important issues for multimedia information retrieval. In this work a speech/music discrimination system is developed which utilizes the Discrete Wavelet Transform (DWT as the acoustic feature. Multi resolution analysis is the most significant statistical way to extract the features from the input signal and in this study, a method is deployed to model the extracted wavelet feature. Support Vector Machines (SVM are based on the principle of structural risk minimization. SVM is applied to classify audio into their classes namely speech and music, by learning from training data. Then the proposed method extends the application of Gaussian Mixture Models (GMM to estimate the probability density function using maximum likelihood decision methods. The system shows significant results with an accuracy of 94.5%.

  18. Design and Usability Testing of an Audio Platform Game for Players with Visual Impairments

    Science.gov (United States)

    Oren, Michael; Harding, Chris; Bonebright, Terri L.

    2008-01-01

    This article reports on the evaluation of a novel audio platform game that creates a spatial, interactive experience via audio cues. A pilot study with players with visual impairments, and usability testing comparing the visual and audio game versions using both sighted players and players with visual impairments, revealed that all the…

  19. 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...

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

    Science.gov (United States)

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

    2010-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Christoph Dahl

    2010-04-01

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

  2. Spectacular Attractions: Museums, Audio-Visuals and the Ghosts of Memory

    Directory of Open Access Journals (Sweden)

    Mandelli Elisa

    2015-12-01

    Full Text Available In the last decades, moving images have become a common feature not only in art museums, but also in a wide range of institutions devoted to the conservation and transmission of memory. This paper focuses on the role of audio-visuals in the exhibition design of history and memory museums, arguing that they are privileged means to achieve the spectacular effects and the visitors’ emotional and “experiential” engagement that constitute the main objective of contemporary museums. I will discuss this topic through the concept of “cinematic attraction,” claiming that when embedded in displays, films and moving images often produce spectacular mises en scène with immersive effects, creating wonder and astonishment, and involving visitors on an emotional, visceral and physical level. Moreover, I will consider the diffusion of audio-visual witnesses of real or imaginary historical characters, presented in Phantasmagoria-like displays that simulate ghostly and uncanny apparitions, creating an ambiguous and often problematic coexistence of truth and illusion, subjectivity and objectivity, facts and imagination.

  3. 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)…

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

    Energy Technology Data Exchange (ETDEWEB)

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

    2007-07-01

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

  5. 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"--

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

    Directory of Open Access Journals (Sweden)

    M.B.I. Reaz

    2012-04-01

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

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

    Science.gov (United States)

    Barton, Robert J., III; Chodura, Hartmut

    2000-07-01

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

  8. State of the art in digital audio/video communications

    CERN Document Server

    Flückiger, François

    1997-01-01

    This series of 5 lectures will introduce the principles, describe the technology and discuss the latest advances in digital audio and video communications. The first lecture presents the principles of the digitisation process of sounds, still and moving images, and discusses its advantages and drawbacks. Encoding formats are also presented. Lecture 2 and 3 are dedicated to compression principles and techniques : why to compress, how, what are the differences between still and moving images ? Source, entropy, transform, vector quantization compression are explained. Advanced techniques such as psyshoacoustic modelling, wavelet encoding, or object-based compression are also described. Lecture 4 concentrates on the requirements placed by digital audio and video applications on the underlying communications networks (latencies, delay variation, isochronism, multicasting). Lecture 5 describes what can and cannot be done today over existing networks such as the Internet, discusses the evolution of the digital media...

  9. Evaluation of embedded audio feedback on writing assignments.

    Science.gov (United States)

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

    2015-01-01

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

  10. Quantization Audio Watermarking with Optimal Scaling on Wavelet Coefficients

    CERN Document Server

    Chen, S -T; Tu, S -Y

    2011-01-01

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

  11. Adaptive audio watermarking based on SNR in localized regions

    Institute of Scientific and Technical Information of China (English)

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

    2005-01-01

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

  12. Sumber Gagasan Penciptaan Karya Audio Visual Berbasis Konten Lokal

    Directory of Open Access Journals (Sweden)

    Dyah Ayu Wiwid Sintowoko

    2016-01-01

    Full Text Available Creativity can be done by utilizing the environment, natural resources, and human resources. Human Resorces can attempt to pour out their thoughts, find problems, and take chances, and viable solutions to publish. Creativity through the media can inspire, encourage, influence, and able to bring a positive change for the audience. The potential of these natural resources that need to be developed by the community at the same time revive, that the stored wealth needs to be raised (one of them with audio-visual approach. In this case the community (residents as actors, not only as a spectator only. Their awareness that we must lift to play the audio-visual media. Something extraordinary in rural or are citizens can see themselves, relatives, family, and neighbors through the medium of film and television, so that it becomes an individual and collective pride.

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

    Directory of Open Access Journals (Sweden)

    Yin Bin

    2003-01-01

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

  14. Audio-visual interactions in product sound design

    Science.gov (United States)

    Özcan, Elif; van Egmond, René

    2010-02-01

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

  15. A novel fiber audio transmission system for secure communication

    Institute of Scientific and Technical Information of China (English)

    SU Ke; JIA Bo

    2005-01-01

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

  16. Random Numbers Generated from Audio and Video Sources

    Directory of Open Access Journals (Sweden)

    I-Te Chen

    2013-01-01

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

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

    Science.gov (United States)

    Gilchrist, N. H. C.

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

  18. Quality and Distortion Evaluation of Audio Signal by Spectrum

    OpenAIRE

    Er. Niranjan Singh; Dr. Bhupendra Verma

    2012-01-01

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

  19. Audio Signal Generator System Based On State Machines

    Institute of Scientific and Technical Information of China (English)

    王维喜

    2009-01-01

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

  20. Audio Quality Assurance : An Application of Cross Correlation

    DEFF Research Database (Denmark)

    Jurik, Bolette Ammitzbøll; Nielsen, Jesper Asbjørn Sindahl

    2012-01-01

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

  1. Audio-magnetotelluric methods in reconnaissance geothermal exploration

    Science.gov (United States)

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

    1976-01-01

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

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

    DEFF Research Database (Denmark)

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

    2015-01-01

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

  3. Amplitude Modulated Sinusoidal Signal Decomposition for Audio Coding

    DEFF Research Database (Denmark)

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

    2006-01-01

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

  4. From ITU-T G.722.1 to ITU-T G.722.1 Annex C: A New Low-Complexity 14kHz Bandwidth Audio Coding Standard

    Directory of Open Access Journals (Sweden)

    Minjie Xie

    2007-04-01

    Full Text Available This paper describes the low-complexity 14kHz bandwidth audio coding algorithm which has been recently standardized by ITU-T as Recommendation G.722.1 Annex C (“G.722.1C”. The algorithm is an extension to ITU-T Recommendation G.722.1 and a doubled form of the G.722.1 algorithm to permit 14 kHz audio bandwidth using a 32 kHz audio sample rate, at 24, 32, and 48 kbit/s. The G. 722.1C codec features very high audio quality, extremely low computational complexity, and low algorithmic delay compared to other state-of-the-art audio coding algorithms. This codec is suitable for use in video conferencing and teleconferencing, and Internet streaming applications as well as a general-purpose 14 kHz audio codec. Subjective test results from the Characterization phase of G 722.1C are also presented in the paper.

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

    Science.gov (United States)

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

    2003-04-01

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

  6. 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.

  7. Comparison of Linear Prediction Models for Audio Signals

    Directory of Open Access Journals (Sweden)

    van Waterschoot Toon

    2008-01-01

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

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

    Science.gov (United States)

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

    2002-05-01

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

  9. DAB: Multiplex and system support features

    Science.gov (United States)

    Riley, J. L.

    This Report describes the multiplex and system support features of the Eureka 147/DAB digital audio system. It sets out the requirements of all users along the broadcast chain from service providers and broadcaster through to the listener. The contents of the transmission frame are examined drawing the distinction between the main service multiplex and the provision of control information in a separate fast data channel. The concept of the DAB service structure is introduced and the inherent system flexibility for altering the service arrangement is explained. A wide range of service information features builds on those provided in earlier systems, such as RDS (Radio Data System) and is intended to make it easier for a listener to find any required service and to add a further dimension to audio broadcasting. The choices available to users in all of these areas are examined.

  10. An Analysis of Translation Strategies for Professional English for Audio Recording Techniques%录音专业英语翻译策略探究

    Institute of Scientific and Technical Information of China (English)

    黄艺平

    2013-01-01

    随着现代传媒科技的发展,音频技术与艺术广泛的结合,越来越深入人们的生活。录音英语的专业性对中西方的跨文化交流形成了一定的障碍。该文通过对录音专业英语的特点进行梳理,结合尤金·奈达的等效翻译理论,力图探索出行之有效的录音专业英语翻译转化之路。%With the development of modern media technology, audio recording technology is widely integrated into artistic forms, which increasingly influences people’s life. The special characteristics of professional English for audio recording technology has somewhat become an obstacle in the cross-cultural exchange between the west and the east. This paper attempts to find some applica-ble translation strategies for professional English for audio recording techniques by analyzing the linguistic features of professional English for audio recording techniques from the perspective of Eugene A. Nida’s Functional Equivalence Theory.

  11. GENIE TRECVID2011 Multimedia Event Detection: Late Fusion Approaches to Combine Multiple Audio Visual features

    Science.gov (United States)

    2012-03-01

    27, May 2011. [2] Sheng Gao, De- Hong Wang , and Chin-Hui Lee. Automatic image annotation through multi-topic text categorization. In ICASSP, 2006. [3...International Journal of Computer Vision, 42(3):145175, 2001. [9] De- Hong Wang , Sheng Gao, Qi Tian, and Wing-Kin Sung. Discriminative fusion approach for automatic image annotation. In MMSP, 2005. ...Government. References [1] Chih -Chung Chang and Chih -Jen Lin. Libsvm: A library for support vector machines. ACM Trans. Intell. Syst. Technol., 2:27:127

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

    Institute of Scientific and Technical Information of China (English)

    Zhou Ping; Liu Zhuo; Xia Liang

    2008-01-01

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

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

    Science.gov (United States)

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

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

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

    Directory of Open Access Journals (Sweden)

    Gunjan Nehru

    2012-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Fei Wang

    2014-01-01

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

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

    Science.gov (United States)

    Wang, Fei; Xie, Zhaoxin; Chen, Zuo

    2014-01-01

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

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

    OpenAIRE

    Puidokas, Vytenis

    2011-01-01

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

  18. 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....

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

    Science.gov (United States)

    Fletcher, R.

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

  20. 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

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

    Institute of Scientific and Technical Information of China (English)

    涂艳; 刘俊

    2016-01-01

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

  2. Audio-haptic interaction in simulated walking experiences

    DEFF Research Database (Denmark)

    Serafin, Stefania

    2011-01-01

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

  3. Audio Hijack Pro万能录音机

    Institute of Scientific and Technical Information of China (English)

    2004-01-01

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

  4. Lost Audio Packets Steganography: The First Practical Evaluation

    CERN Document Server

    Mazurczyk, Wojciech

    2011-01-01

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

  5. Unsupervised incremental online learning and prediction of musical audio signals

    DEFF Research Database (Denmark)

    Marxer, Richard; Purwins, Hendrik

    2016-01-01

    Guided by the idea that musical human-computer interaction may become more effective, intuitive, and creative when basing its computer part on cognitively more plausible learning principles, we employ unsupervised incremental online learning (i.e. clustering) to build a system that predicts...... the next event in a musical sequence, given as audio input. The flow of the system is as follows: 1) segmentation by onset detection, 2) timbre representation of each segment by Mel frequency cepstrum coefficients, 3) discretization by incremental clustering, yielding a tree of different sound classes (e...

  6. Sinusoidal Analysis-Synthesis of Audio Using Perceptual Criteria

    Directory of Open Access Journals (Sweden)

    Ted Painter

    2003-01-01

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

  7. Digital audio recordings improve the outcomes of patient consultations

    DEFF Research Database (Denmark)

    Wolderslund, Maiken; Kofoed, Poul-Erik; Holst, René

    2017-01-01

    OBJECTIVES: To investigate the effects on patients' outcome of the consultations when provided with: a Digital Audio Recording (DAR) of the consultation and a Question Prompt List (QPL). METHODS: This is a three-armed randomised controlled cluster trial. One group of patients received standard care......, while the other two groups received either the QPL in combination with a recording of their consultation or only the recording. Patients from four outpatient clinics participated: Paediatric, Orthopaedic, Internal Medicine, and Urology. The effects were evaluated by patient-administered questionnaires...

  8. Digital audio recordings improve the outcomes of patient consultations

    DEFF Research Database (Denmark)

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

    2016-01-01

    OBJECTIVES: To investigate the effects on patients' outcome of the consultations when provided with: a Digital Audio Recording (DAR) of the consultation and a Question Prompt List (QPL). METHODS: This is a three-armed randomised controlled cluster trial. One group of patients received standard care......, while the other two groups received either the QPL in combination with a recording of their consultation or only the recording. Patients from four outpatient clinics participated: Paediatric, Orthopaedic, Internal Medicine, and Urology. The effects were evaluated by patient-administered questionnaires...

  9. Inexpensive Audio Activities: Earbud-based Sound Experiments

    Science.gov (United States)

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

    2016-11-01

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

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

    Science.gov (United States)

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

    2010-01-01

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

  11. Utilization of non-linear converters for audio amplification

    DEFF Research Database (Denmark)

    Iversen, Niels Elkjær; Birch, Thomas; Knott, Arnold

    2012-01-01

    Class D amplifiers fits the automotive demands quite well. The traditional buck-based amplifier has reduced both the cost and size of amplifiers. However the buck topology is not without its limitations. The maximum peak AC output voltage produced by the power stage is only equal the supply voltage....... The introduction of non-linear converters for audio amplification defeats this limitation. A Cuk converter, designed to deliver an AC peak output voltage twice the supply voltage, is presented in this paper. A 3V prototype has been developed to prove the concept. The prototype shows that it is possible to achieve...

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

    CERN Document Server

    Moreau, Nicolas

    2013-01-01

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

  13. A listening test system for automotive audio - listeners

    DEFF Research Database (Denmark)

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

    2007-01-01

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

  14. The complete guide to high-end audio

    CERN Document Server

    Harley, Robert

    2015-01-01

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

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

    DEFF Research Database (Denmark)

    Dimitrov, Smilen; Serafin, Stefania

    2011-01-01

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

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

    Science.gov (United States)

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

    2013-07-01

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

  17. The Research on Wavelet Audio Watermark Based on Independent Component Analysis

    Energy Technology Data Exchange (ETDEWEB)

    Ma, X F [Engineering Training Centre, Harbin Engineering University, Harbin, 150001 (China); Jiang, T [Info. and Comm. Engineering College, Harbin Engineering University, Harbin, 150001 (China)

    2006-10-15

    Along with the development of the watermark technique many new scheme were presented and most of them were proved efficient. Some researchers have presented extraction of audio watermark using ICA in spatial domain. In this paper, we present a wavelet audio watermark using ICA. We embedded the image watermark into the wavelet coefficient of the audio signal, and extracted the watermark image using ICA in wavelet domain. We added noise on the watermark audio for analysis and the simulation results show that this watermark scheme we present is efficient and robustness.

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

    Science.gov (United States)

    Buś, Szymon; Jedrzejewski, Konrad

    2015-09-01

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

  19. Efficiency of Switch-Mode Power Audio Amplifiers - Test Signals and Measurement Techniques

    DEFF Research Database (Denmark)

    Iversen, Niels Elkjær; Knott, Arnold; Andersen, Michael A. E.

    2016-01-01

    Switch-mode technology is greatly used for audio amplification. This is mainly due to the great efficiency this technology offers. Normally the efficiency of a switch-mode audio amplifier is measured using a sine wave input. However this paper shows that sine waves represent real audio very poorly....... An alternative signal is proposed for test purposes. The efficiency of a switch-mode power audio amplifier is modelled and measured with both sine wave and the proposed test signal as inputs. The results show that the choice of switching devices with low on resistances are unfairly favored when measuring...

  20. On the relevance of spectral features for instrument classification

    DEFF Research Database (Denmark)

    Nielsen, Andreas Brinch; Sigurdsson, Sigurdur; Hansen, Lars Kai

    2007-01-01

    Automatic knowledge extraction from music signals is a key component for most music organization and music information retrieval systems. In this paper, we consider the problem of instrument modelling and instrument classification from the rough audio data. Existing systems for automatic instrument...... classification operate normally on a relatively large number of features, from which those related to the spectrum of the audio signal are particularly relevant. In this paper, we confront two different models about the spectral characterization of musical instruments. The first assumes a constant envelope...

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

    Science.gov (United States)

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

    2015-01-01

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

  2. Separate mechanisms for audio-tactile pitch and loudness interactions

    Directory of Open Access Journals (Sweden)

    Jeffrey M Yau

    2010-10-01

    Full Text Available A major goal in perceptual neuroscience is to understand how signals from different sensory modalities are combined to produce stable and coherent representations. We previously investigated interactions between audition and touch, motivated by the fact that both modalities are sensitive to environmental oscillations. In our earlier study, we characterized the effect of auditory distractors on tactile frequency and intensity perception. Here, we describe the converse experiments examining the effect of tactile distractors on auditory processing. Because the two studies employ the same psychophysical paradigm, we combined their results for a comprehensive view of how auditory and tactile signals interact and how these interactions depend on the perceptual task. Together, our results show that temporal frequency representations are perceptually linked regardless of the attended modality. In contrast, audio-tactile loudness interactions depend on the attended modality: Tactile distractors influence judgments of auditory intensity, but judgments of tactile intensity are impervious to auditory distraction. Lastly, we show that audio-tactile loudness interactions depend critically on stimulus timing, while pitch interactions do not. These results reveal that auditory and tactile inputs are combined differently depending on the perceptual task. That distinct rules govern the integration of auditory and tactile signals in pitch and loudness perception implies that the two are mediated by separate neural mechanisms. These findings underscore the complexity and specificity of multisensory interactions.

  3. 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; ·     �...

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

    Science.gov (United States)

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

    2014-01-01

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

  5. 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.

  6. Audio Editing Skills%音频剪辑技巧

    Institute of Scientific and Technical Information of China (English)

    范炜; 杨澍彬; 谭忠凯

    2015-01-01

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

  7. Audio watermarking technologies for automatic cue sheet generation systems

    Science.gov (United States)

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

    2001-08-01

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

  8. Noise Adaptive Stream Weighting in Audio-Visual Speech Recognition

    Directory of Open Access Journals (Sweden)

    Martin Heckmann

    2002-11-01

    Full Text Available It has been shown that integration of acoustic and visual information especially in noisy conditions yields improved speech recognition results. This raises the question of how to weight the two modalities in different noise conditions. Throughout this paper we develop a weighting process adaptive to various background noise situations. In the presented recognition system, audio and video data are combined following a Separate Integration (SI architecture. A hybrid Artificial Neural Network/Hidden Markov Model (ANN/HMM system is used for the experiments. The neural networks were in all cases trained on clean data. Firstly, we evaluate the performance of different weighting schemes in a manually controlled recognition task with different types of noise. Next, we compare different criteria to estimate the reliability of the audio stream. Based on this, a mapping between the measurements and the free parameter of the fusion process is derived and its applicability is demonstrated. Finally, the possibilities and limitations of adaptive weighting are compared and discussed.

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

    Directory of Open Access Journals (Sweden)

    Johahn eLeung

    2016-01-01

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

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

    Directory of Open Access Journals (Sweden)

    Anja Kuchenbuch

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

  11. Effectiveness and Comparison of Various Audio Distraction Aids in Management of Anxious Dental Paediatric Patients.

    Science.gov (United States)

    Navit, Saumya; Johri, Nikita; Khan, Suleman Abbas; Singh, Rahul Kumar; Chadha, Dheera; Navit, Pragati; Sharma, Anshul; Bahuguna, Rachana

    2015-12-01

    Dental anxiety is a widespread phenomenon and a concern for paediatric dentistry. The inability of children to deal with threatening dental stimuli often manifests as behaviour management problems. Nowadays, the use of non-aversive behaviour management techniques is more advocated, which are more acceptable to parents, patients and practitioners. Therefore, this present study was conducted to find out which audio aid was the most effective in the managing anxious children. The aim of the present study was to compare the efficacy of audio-distraction aids in reducing the anxiety of paediatric patients while undergoing various stressful and invasive dental procedures. The objectives were to ascertain whether audio distraction is an effective means of anxiety management and which type of audio aid is the most effective. A total number of 150 children, aged between 6 to 12 years, randomly selected amongst the patients who came for their first dental check-up, were placed in five groups of 30 each. These groups were the control group, the instrumental music group, the musical nursery rhymes group, the movie songs group and the audio stories group. The control group was treated under normal set-up & audio group listened to various audio presentations during treatment. Each child had four visits. In each visit, after the procedures was completed, the anxiety levels of the children were measured by the Venham's Picture Test (VPT), Venham's Clinical Rating Scale (VCRS) and pulse rate measurement with the help of pulse oximeter. A significant difference was seen between all the groups for the mean pulse rate, with an increase in subsequent visit. However, no significant difference was seen in the VPT & VCRS scores between all the groups. Audio aids in general reduced anxiety in comparison to the control group, and the most significant reduction in anxiety level was observed in the audio stories group. The conclusion derived from the present study was that audio distraction

  12. Reducing audio stimulus presentation latencies across studies, laboratories, and hardware and operating system configurations.

    Science.gov (United States)

    Babjack, Destiny L; Cernicky, Brandon; Sobotka, Andrew J; Basler, Lee; Struthers, Devon; Kisic, Richard; Barone, Kimberly; Zuccolotto, Anthony P

    2015-09-01

    Using differing computer platforms and audio output devices to deliver audio stimuli often introduces (1) substantial variability across labs and (2) variable time between the intended and actual sound delivery (the sound onset latency). Fast, accurate audio onset latencies are particularly important when audio stimuli need to be delivered precisely as part of studies that depend on accurate timing (e.g., electroencephalographic, event-related potential, or multimodal studies), or in multisite studies in which standardization and strict control over the computer platforms used is not feasible. This research describes the variability introduced by using differing configurations and introduces a novel approach to minimizing audio sound latency and variability. A stimulus presentation and latency assessment approach is presented using E-Prime and Chronos (a new multifunction, USB-based data presentation and collection device). The present approach reliably delivers audio stimuli with low latencies that vary by ≤1 ms, independent of hardware and Windows operating system (OS)/driver combinations. The Chronos audio subsystem adopts a buffering, aborting, querying, and remixing approach to the delivery of audio, to achieve a consistent 1-ms sound onset latency for single-sound delivery, and precise delivery of multiple sounds that achieves standard deviations of 1/10th of a millisecond without the use of advanced scripting. Chronos's sound onset latencies are small, reliable, and consistent across systems. Testing of standard audio delivery devices and configurations highlights the need for careful attention to consistency between labs, experiments, and multiple study sites in their hardware choices, OS selections, and adoption of audio delivery systems designed to sidestep the audio latency variability issue.

  13. Audio Feedback: Richer Language but No Measurable Impact on Student Performance

    Science.gov (United States)

    Chalmers, Charlotte; MacCallum, Janis; Mowat, Elaine; Fulton, Norma

    2014-01-01

    Audio feedback has been shown to be popular and well received by students. However, there is little published work to indicate how effective audio feedback is in improving student performance. Sixty students from a first year science degree agreed to take part in the study; thirty were randomly assigned to receive written feedback on coursework,…

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

    DEFF Research Database (Denmark)

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

    2012-01-01

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

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

    Science.gov (United States)

    Branje, Carmen J.; Fels, Deborah I.

    2012-01-01

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

  16. DOUBLE-BOOST DC-AC CONVERTER WITH SLIDING-MODE CONTROL FOR PORTABLE AUDIO

    DEFF Research Database (Denmark)

    Bolten Maizonave, Gert; Andersen, Michael Andreas E.; Kjærgaard, Claus

    2009-01-01

    The double-boost topology is studied for operation as a dc-ac converter and single stage audio amplifier. A sliding-mode controller is designed in order to achieve fast enough response for the whole audio frequency range. Symmetric, asymmetric and interleaved operation modes are analyzed....

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

    DEFF Research Database (Denmark)

    Dimitrov, Smilen; Serafin, Stefania

    2011-01-01

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

  18. Instruction document on multimedia formats:optimal accessibility of audio, video and images

    NARCIS (Netherlands)

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

    2010-01-01

    We increasingly express ourselves through multimedia. Internet traffic already consists for the most part of audio and video. A variety of formats are used for this purpose, often without due consideration. This document provides a background for choices that can be made for making video and audio a

  19. 47 CFR 73.4275 - Tone clusters; audio attention-getting devices.

    Science.gov (United States)

    2010-10-01

    ... 47 Telecommunication 4 2010-10-01 2010-10-01 false Tone clusters; audio attention-getting devices. 73.4275 Section 73.4275 Telecommunication FEDERAL COMMUNICATIONS COMMISSION (CONTINUED) BROADCAST... clusters; audio attention-getting devices. See Public Notice, FCC 76-610, dated July 2, 1976. 60 FCC 2d...

  20. A Perceptually Reweighted Mixed-Norm Method for Sparse Approximation of Audio Signals

    DEFF Research Database (Denmark)

    Christensen, Mads Græsbøll; Sturm, Bob L.

    2011-01-01

    In this paper, we consider the problem of finding sparse representations of audio signals for coding purposes. In doing so, it is of utmost importance that when only a subset of the present components of an audio signal are extracted, it is the perceptually most important ones. To this end, we pr...

  1. 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...

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

    Science.gov (United States)

    Fawcett, Hannah; Oldfield, Jeremy

    2016-01-01

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

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

    Science.gov (United States)

    Fawcett, Hannah; Oldfield, Jeremy

    2016-01-01

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

  4. 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)...

  5. A Management Review and Analysis of Purdue University Libraries and Audio-Visual Center.

    Science.gov (United States)

    Baaske, Jan; And Others

    A management review and analysis was conducted by the staff of the libraries and audio-visual center of Purdue University. Not only were the study team and the eight task forces drawn from all levels of the libraries and audio-visual center staff, but a systematic effort was sustained through inquiries, draft reports and open meetings to involve…

  6. 47 CFR Figure 2 to Subpart N of... - Typical Audio Wave

    Science.gov (United States)

    2010-10-01

    ... 47 Telecommunication 1 2010-10-01 2010-10-01 false Typical Audio Wave 2 Figure 2 to Subpart N of Part 2 Telecommunication FEDERAL COMMUNICATIONS COMMISSION GENERAL FREQUENCY ALLOCATIONS AND RADIO... Audio Wave EC03JN91.006...

  7. Audio-video decision support for patients: the documentary genre as a basis for decision aids

    NARCIS (Netherlands)

    Volandes, A.E.; Barry, M.J.; Wood, F.; Elwyn, G.

    2013-01-01

    Objective Decision support tools are increasingly using audio-visual materials. However, disagreement exists about the use of audio-visual materials as they may be subjective and biased. Methods This is a literature review of the major texts for documentary film studies to extrapolate issues of obje

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

    NARCIS (Netherlands)

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

    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 benchmark

  9. 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…

  10. A Preliminary Investigation into the Search Behaviour of Users in a Collection of Digitized Broadcast Audio

    DEFF Research Database (Denmark)

    Lund, Haakon; Skov, Mette; Larsen, Birger;

    2014-01-01

    An increasing number of large digitized audio-visual collections within digital humanities have recently been made available for users. Often access to digitized audio-visual collections is hampered by little and inconsistent metadata. This paper presents the preliminary findings from a study of ...

  11. An Analysis of Certain Elements of an Audio-Tape Approach to Instruction.

    Science.gov (United States)

    Bell, Ronald Ernest

    This study was designed to determine the association between selected variables and an audio-tape approach to instruction. Fifty sophomore students enrolled in a physical anthropology course at Shoreline Community College (Washington) participated in an experimental instructional program that consisted of thirty-two audio-tapes and three optional…

  12. 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

  13. 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...

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

    Institute of Scientific and Technical Information of China (English)

    李浚

    2011-01-01

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

  15. 多格式音频感知哈希算法%Perceptual Hashing Algorithm for Multi-Format Audio

    Institute of Scientific and Technical Information of China (English)

    张秋余; 省鹏飞; 黄羿博; 董瑞洪; 杨仲平

    2016-01-01

    提出一种基于双树复小波变换的多格式音频感知哈希算法,解决了现有音频认证算法音频格式单一、算法不通用、效率低的问题.首先对预处理后的音频信号进行全局双树复小波变换,获得信号的实小波和复小波系数,对它们分别分帧,帧数相同;对实小波系数计算每帧信号Teager能量算子的模值,作为实小波系数的帧间特征,接着对每帧信号再分帧,提取再分帧帧信号的短时能量作为实小波系数的帧内特征;对复小波系数求取每帧信号的熵值作为复小波系数的帧间特征;最后对上述特征分别进行哈希构造,生成感知哈希序列.实验结果表明,该算法对5种不同格式的音频都具有强鲁棒性,且区分性好,效率高,并能实现小范围篡改检测.%A novel multi-format audio perceptual hashing algorithm based on dual tree complex wavelet transform ( DT-CWT ) was proposed. It solves the problems of the existing audio authentication algo-rithms, including that audio files are kept in a single format, and algorithms are not generic and low effi-ciency. The proposed algorithm first applies the global DT-CWT to the audio signal after pre-processing conducts to obtain the real and complex wavelet coefficients. Next, the coefficients are partitioned in some frames respectively, and the frame numbers are same. For the real wavelet coefficients, the module values of teager energy operator in every frame are computed to serve as its inter-frame feature. And then short-time energy of the new signal, which is generated to frame the frame signal, is computed to serve as its intra-frame feature. For the complex wavelet coefficients, entropy values are obtained in every frame to serve as its inter-frame feature. Finally, the above features are to conduct a hashing structure process to produce the perceptual hashing sequence. Experiments show that the proposed algorithm has good robust-ness and discrimination for audio

  16. Implementasi Teknik Watermarking menggunakan FFT dan Spread Spectrum Watermark pada Data Audio Digital

    Directory of Open Access Journals (Sweden)

    HANNAN HARAHAP

    2016-02-01

    Full Text Available ABSTRAK Penggunaan teknologi dan internet yang berkembang dengan pesat menyebabkan banyak pemalsuan dan penyebaran yang tidak sah terhadap data digital. Oleh karena itu, sangat diperlukan suatu teknologi yang dapat melindungi hak cipta data multimedia seperti audio. Teknik yang sering digunakan dalam perlindungan hak cipta adalah watermarking karena teknik ini memiliki tiga kriteria utama dalam keamanan data, yaitu robustness, imperceptibility, dan safety. Untuk itu, pada penelitian ini dirancang suatu skema yang dapat melindungi hak cipta data audio. Metode yang digunakan adalah Fast Fourier Transform, yang mengubah data audio asli ke dalam domain frekuensi sebelum dilakukan proses penyisipan watermark dan proses ekstraksi watermark. Watermark disebar pada komponen yang paling signifikan dari spektrum magnitude audio host. Teknik watermarking pada penelitian ini dapat menghasilkan Signal-to-Noise Ratio di atas 20 dB dan Bit Error Rate di bawah 5%. Kata kunci: Audio watermarking, Copyright Protection, Fast Fourier Transform, Spektrum magnitude ABSTRACT The use of technology and internet has grown rapidly that causes a lot of forgery and illegal proliferation of digital data. It needs a technology that can protect the copyright of multimedia data such as audio. The most common technique in copyright protection is watermarking because it has three main criteria in data security: robustness, imperceptibility, and safety. This research created a scheme that can protect a copyright of audio data. The method that we used is Fast Fourier Transform. This method changes the original audio data into frequency domain before the embedding and extraction process. The watermark is spread into the most significant component of the magnitude spectrum of audio host. This technique obtains Signal-to-Noise Ratio above 20 dB and Bit Error Rate below 5%. Keywords: Audio watermarking, Copyright Protection, Fast Fourier Transform, Magnitude spectrum

  17. Audio-visual speech in noise perception in dyslexia.

    Science.gov (United States)

    van Laarhoven, Thijs; Keetels, Mirjam; Schakel, Lemmy; Vroomen, Jean

    2016-12-18

    Individuals with developmental dyslexia (DD) may experience, besides reading problems, other speech-related processing deficits. Here, we examined the influence of visual articulatory information (lip-read speech) at various levels of background noise on auditory word recognition in children and adults with DD. We found that children with a documented history of DD have deficits in their ability to gain benefit from lip-read information that disambiguates noise-masked speech. We show with another group of adult individuals with DD that these deficits persist into adulthood. These deficits could not be attributed to impairments in unisensory auditory word recognition. Rather, the results indicate a specific deficit in audio-visual speech processing and suggest that impaired multisensory integration might be an important aspect of DD. © 2016 John Wiley & Sons Ltd.

  18. Haptic and Visual feedback in 3D Audio Mixing Interfaces

    DEFF Research Database (Denmark)

    Gelineck, Steven; Overholt, Daniel

    2015-01-01

    in order to augment the perception of the 3D space. We compare different interaction paradigms implemented using these interfaces, aiming to increase speed and accuracy and reduce the need for constant visual feedback. While the LEAP Motion relies upon visual perception and proprioception, users can forego......This paper describes the implementation and informal evaluation of a user interface that explores haptic feedback for 3D audio mixing. The implementation compares different approaches using either the LEAP Motion for mid-air hand gesture control, or the Novint Falcon for active haptic feed- back...... visual feedback with interfaces such as the Novint Falcon and rely primarily on haptic cues, allowing more focus on the spatial sound elements. Results of the evaluation support this claim, as users preferred the interaction paradigm using the Falcon with no visual feedback. Furthermore, users disliked...

  19. Audio collection in the SASA Institute of Musicology

    Directory of Open Access Journals (Sweden)

    Lajić-Mihajlović Danka

    2010-01-01

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

  20. Audio-haptic-virtual Mona Lisa (The Blind and Painting

    Directory of Open Access Journals (Sweden)

    Aksinja Kermauner

    2014-03-01

    Full Text Available The purpose of the article is to explore in what ways the visual arts (with emphasis on painting can be brought closer to the blind in the postmodern society, in which sight is perceived to be the chief sense and in which most information is based on images. The basic methods of presenting a work of art involve the remaining senses, mostly those of hearing and touch. It is of course not enough just to deliver a factual description of a painting or to transform it into tactile graphics – more complex techniques such as audio-description, method of associations, participating in role-playing, all with the aim of a holistic experience of the work of art, must be sought instead. In the world of virtual reality, additional equipment for the blind (e.g., data gloves provides new opportunities.

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

    Science.gov (United States)

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

    2006-08-01

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

  2. Class D audio amplifiers for high voltage capacitive transducers

    DEFF Research Database (Denmark)

    Nielsen, Dennis

    voltage capacitive transducers can be constructed with THD+N below 0.1 % and peak efficiency above 80 %. However the complexity of the amplifier combined with the current high cost of components, makes the technology of DEAP based loudspeaker unfeasible. Suggestions to future work in the pursuit...... 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....... Due to the similarities between the electrostatic loudspeaker and the DEAP transducer, the state-of-the-art has a special focus on amplifiers for electrostatic loudspeakers. Amplifiers for other type of capacitive transducers like piezoelectric ones are also considered. Finally the current state...

  3. Real Time Recognition Of Speakers From Internet Audio Stream

    Directory of Open Access Journals (Sweden)

    Weychan Radoslaw

    2015-09-01

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

  4. Real-time Covert Communications Channel for Audio Signals

    Directory of Open Access Journals (Sweden)

    Ashraf Seleym

    2012-09-01

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

  5. 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)

  6. Vertigo with sudden hearing loss: audio-vestibular characteristics.

    Science.gov (United States)

    Pogson, Jacob M; Taylor, Rachael L; Young, Allison S; McGarvie, Leigh A; Flanagan, Sean; Halmagyi, G Michael; Welgampola, Miriam S

    2016-10-01

    Acute vertigo with sudden sensorineural hearing loss (SSNHL) is a rare clinical emergency. Here, we report the audio-vestibular test profiles of 27 subjects who presented with these symptoms. The vestibular test battery consisted of a three-dimensional video head impulse test (vHIT) of semicircular canal function and recording ocular and cervical vestibular-evoked myogenic potentials (oVEMP, cVEMP) to test otolith dysfunction. Unlike vestibular neuritis, where the horizontal and anterior canals with utricular function are more frequently impaired, 74 % of subjects with vertigo and SSNHL demonstrated impairment of the posterior canal gain (0.45 ± 0.20). Only 41 % showed impairment of the horizontal canal gains (0.78 ± 0.27) and 30 % of the anterior canal gains (0.79 ± 0.26), while 38 % of oVEMPs [asymmetry ratio (AR) = 41.0 ± 41.3 %] and 33 % of cVEMPs (AR = 47.3 ± 41.2 %) were significantly asymmetrical. Twenty-three subjects were diagnosed with labyrinthitis/labyrinthine infarction in the absence of evidence for an underlying pathology. Four subjects had a definitive diagnosis [Ramsay Hunt Syndrome, vestibular schwannoma, anterior inferior cerebellar artery (AICA) infarction, and traction injury]. Ischemia involving the common-cochlear or vestibulo-cochlear branches of the labyrinthine artery could be the simplest explanation for vertigo with SSNHL. Audio-vestibular tests did not provide easy separation between ischaemic and non-ischaemic causes of vertigo with SSNHL.

  7. The audio-visual revolution: do we really need it?

    Science.gov (United States)

    Townsend, I

    1979-03-01

    In the United Kingdom, The audio-visual revolution has steadily gained converts in the nursing profession. Nurse tutor courses now contain information on the techniques of educational technology and schools of nursing increasingly own (or wish to own) many of the sophisticated electronic aids to teaching that abound. This is taking place at a time of hitherto inexperienced crisis and change. Funds have been or are being made available to buy audio-visual equipment. But its purchase and use relies on satisfying personal whim, prejudice or educational fashion, not on considerations of educational efficiency. In the rush of enthusiasm, the overwhelmed teacher (everywhere; the phenomenon is not confined to nursing) forgets to ask the searching, critical questions: 'Why should we use this aid?','How effective is it?','And, at what?'. Influential writers in this profession have repeatedly called for a more responsible attitude towards published research work of other fields. In an attempt to discover what is known about the answers to this group of questions, an eclectic look at media research is taken and the widespread dissatisfaction existing amongst international educational technologists is noted. The paper isolates out of the literature several causative factors responsible for the present state of affairs. Findings from the field of educational television are cited as representative of an aid which has had a considerable amount of time and research directed at it. The concluding part of the paper shows the decisions to be taken in using or not using educational media as being more complicated than might at first appear.

  8. Something for Everyone? An Evaluation of the Use of Audio-Visual Resources in Geographical Learning in the UK.

    Science.gov (United States)

    McKendrick, John H.; Bowden, Annabel

    1999-01-01

    Reports from a survey of geographers that canvassed experiences using audio-visual resources to support teaching. Suggests that geographical learning has embraced audio-visual resources and that they are employed effectively. Concludes that integration of audio-visual resources into mainstream curriculum is essential to ensure effective and…

  9. 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)

  10. APPLICATION OF PARTIAL LEAST SQUARES REGRESSION FOR AUDIO-VISUAL SPEECH PROCESSING AND MODELING

    Directory of Open Access Journals (Sweden)

    A. L. Oleinik

    2015-09-01

    Full Text Available Subject of Research. The paper deals with the problem of lip region image reconstruction from speech signal by means of Partial Least Squares regression. Such problems arise in connection with development of audio-visual speech processing methods. Audio-visual speech consists of acoustic and visual components (called modalities. Applications of audio-visual speech processing methods include joint modeling of voice and lips’ movement dynamics, synchronization of audio and video streams, emotion recognition, liveness detection. Method. Partial Least Squares regression was applied to solve the posed problem. This method extracts components of initial data with high covariance. These components are used to build regression model. Advantage of this approach lies in the possibility of achieving two goals: identification of latent interrelations between initial data components (e.g. speech signal and lip region image and approximation of initial data component as a function of another one. Main Results. Experimental research on reconstruction of lip region images from speech signal was carried out on VidTIMIT audio-visual speech database. Results of the experiment showed that Partial Least Squares regression is capable of solving reconstruction problem. Practical Significance. Obtained findings give the possibility to assert that Partial Least Squares regression is successfully applicable for solution of vast variety of audio-visual speech processing problems: from synchronization of audio and video streams to liveness detection.

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

    Directory of Open Access Journals (Sweden)

    Jussi Kuutti

    2014-02-01

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

  12. Impact of audio narrated animation on students' understanding and learning environment based on gender

    Science.gov (United States)

    Nasrudin, Ajeng Ratih; Setiawan, Wawan; Sanjaya, Yayan

    2017-05-01

    This study is titled the impact of audio narrated animation on students' understanding in learning humanrespiratory system based on gender. This study was conducted in eight grade of junior high school. This study aims to investigate the difference of students' understanding and learning environment at boys and girls classes in learning human respiratory system using audio narrated animation. Research method that is used is quasy experiment with matching pre-test post-test comparison group design. The procedures of study are: (1) preliminary study and learning habituation using audio narrated animation; (2) implementation of learning using audio narrated animation and taking data; (3) analysis and discussion. The result of analysis shows that there is significant difference on students' understanding and learning environment at boys and girls classes in learning human respiratory system using audio narrated animation, both in general and specifically in achieving learning indicators. The discussion related to the impact of audio narrated animation, gender characteristics, and constructivist learning environment. It can be concluded that there is significant difference of students' understanding at boys and girls classes in learning human respiratory system using audio narrated animation. Additionally, based on interpretation of students' respond, there is the difference increment of agreement level in learning environment.

  13. Broadcast News Story Segmentation Using Conditional Random Fields and Multimodal Features

    Science.gov (United States)

    Wang, Xiaoxuan; Xie, Lei; Lu, Mimi; Ma, Bin; Chng, Eng Siong; Li, Haizhou

    In this paper, we propose integration of multimodal features using conditional random fields (CRFs) for the segmentation of broadcast news stories. We study story boundary cues from lexical, audio and video modalities, where lexical features consist of lexical similarity, chain strength and overall cohesiveness; acoustic features involve pause duration, pitch, speaker change and audio event type; and visual features contain shot boundaries, anchor faces and news title captions. These features are extracted in a sequence of boundary candidate positions in the broadcast news. A linear-chain CRF is used to detect each candidate as boundary/non-boundary tags based on the multimodal features. Important interlabel relations and contextual feature information are effectively captured by the sequential learning framework of CRFs. Story segmentation experiments show that the CRF approach outperforms other popular classifiers, including decision trees (DTs), Bayesian networks (BNs), naive Bayesian classifiers (NBs), multilayer perception (MLP), support vector machines (SVMs) and maximum entropy (ME) classifiers.

  14. The impact of information fusion in steganalysis on the example of audio steganalysis

    Science.gov (United States)

    Kraetzer, Christian; Dittmann, Jana

    2009-02-01

    Information fusion tries to determine the best set of experts in a given problem domain and devise an appropriate function that can optimally combine the decisions of the individual experts. Only few systematic approaches to information fusion exist so far in the signal processing field of steganalysis. Under the basic assumption that steganalysis can be seen as a statistical pattern recognition process like biometrics, a state of the art five level information fusion model known from biometrics is transferred to steganalysis as well as statistical detectability evaluations for watermarking algorithms and its applicability is evaluated in practical testing. The primary test goal for these evaluations is to measure the impact of fusion on the classification accuracy. Therefore a match and decision level fusion are performed here for three selected data hiding algorithms (one steganography and two watermarking), two feature extractors and five different classifiers. For the test heterogeneous audio test sets are used for content independent training and testing. The secondary test goal of this work is to consider the impact of the key selection assumption on the accuracy of the classification in steganalysis. The results show for the test cases an increase of the classification accuracy for two of the three tested algorithms by match level fusions, no gain by decision level fusion and a considerably small impact of the key selection assumption on the statistical detectability.

  15. One-Dimensional Haptic Rendering Using Audio Speaker with Displacement Determined by Inductance

    Directory of Open Access Journals (Sweden)

    Avin Khera

    2016-03-01

    Full Text Available We report overall design considerations and preliminary results for a new haptic rendering device based on an audio loudspeaker. Our application models tissue properties during microsurgery. For example, the device could respond to the tip of a tool by simulating a particular tissue, displaying a desired compressibility and viscosity, giving way as the tissue is disrupted, or exhibiting independent motion, such as that caused by pulsations in blood pressure. Although limited to one degree of freedom and with a relatively small range of displacement compared to other available haptic rendering devices, our design exhibits high bandwidth, low friction, low hysteresis, and low mass. These features are consistent with modeling interactions with delicate tissues during microsurgery. In addition, our haptic rendering device is designed to be simple and inexpensive to manufacture, in part through an innovative method of measuring displacement by existing variations in the speaker’s inductance as the voice coil moves over the permanent magnet. Low latency and jitter are achieved by running the real-time simulation models on a dedicated microprocessor, while maintaining bidirectional communication with a standard laptop computer for user controls and data logging.

  16. Audio Classification in Speech and Music: A Comparison between a Statistical and a Neural Approach

    Directory of Open Access Journals (Sweden)

    Bugatti Alessandro

    2002-01-01

    Full Text Available We focus the attention on the problem of audio classification in speech and music for multimedia applications. In particular, we present a comparison between two different techniques for speech/music discrimination. The first method is based on Zero crossing rate and Bayesian classification. It is very simple from a computational point of view, and gives good results in case of pure music or speech. The simulation results show that some performance degradation arises when the music segment contains also some speech superimposed on music, or strong rhythmic components. To overcome these problems, we propose a second method, that uses more features, and is based on neural networks (specifically a multi-layer Perceptron. In this case we obtain better performance, at the expense of a limited growth in the computational complexity. In practice, the proposed neural network is simple to be implemented if a suitable polynomial is used as the activation function, and a real-time implementation is possible even if low-cost embedded systems are used.

  17. 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.

  18. Steganography: Applying and Evaluating Two Algorithms for Embedding Audio Data in an Image

    Directory of Open Access Journals (Sweden)

    Khaled Nasser ElSayed

    2015-03-01

    Full Text Available Information transmission is increasing with grow of using WEB. So, information security has become very important. Security of data and information is the major task for scientists and political and military people. One of the most secure methods is embedding data (steganography in different media like text, audio, digital images. this paper present two experiments in steganography of digital audio data file. It applies empirically, two algorithms in steganography in images through random insertion of digital audio data using bytes and pixels in image files. Finally, it evaluates both experiments, in order to enhance security of transmitted data.

  19. 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...

  20. 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...... 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...