24 research outputs found
An investigation into tools and protocols for commercial audio web-site creation
This thesis presents a feasibility study of a Web-based digital music library and purchasing system. It investigates the current status of the enabling technologies for developing such a system. An analysis of various Internet audio codecs, streaming audio protocols, Internet credit card payment security methods, and ways for accessing remote Web databases is presented. The objective of the analysis is to determine the viability and the economic benefits of using these technologies when developing systems that facilitate music distribution over the Internet. A prototype of a distributed digital music library and purchasing system named WAPS (for Web-based Audio Purchasing System) was developed and implemented in the Java programming language. In this thesis both the physical and the logical component elements of WAPS are explored in depth so as to provide an insight into the inherent problems of creating such a system, as well as the overriding benefits derived from the creation of such a system
Scalable and perceptual audio compression
This thesis deals with scalable perceptual audio compression. Two scalable perceptual solutions as well as a scalable to lossless solution are proposed and investigated. One of the scalable perceptual solutions is built around sinusoidal modelling of the audio signal whilst the other is built on a transform coding paradigm. The scalable coders are shown to scale both in a waveform matching manner as well as a psychoacoustic manner. In order to measure the psychoacoustic scalability of the systems investigated in this thesis, the similarity between the original signal\u27s psychoacoustic parameters and that of the synthesized signal are compared. The psychoacoustic parameters used are loudness, sharpness, tonahty and roughness. This analysis technique is a novel method used in this thesis and it allows an insight into the perceptual distortion that has been introduced by any coder analyzed in this manner
Very low bit rate parametric audio coding
[no abstract
Survey of error concealment schemes for real-time audio transmission systems
This thesis presents an overview of the main strategies employed for error detection and error concealment in different real-time transmission systems for digital audio. The âAdaptive Differential Pulse-Code Modulation (ADPCM)â, the âAudio Processing Technology Apt-x100â, the âExtended Adaptive Multi-Rate Wideband (AMR-WB+)â, the âAdvanced Audio Coding (AAC)â, the âMPEG-1 Audio Layer II (MP2)â, the âMPEG-1 Audio Layer III (MP3)â and finally the âAdaptive Transform Coder 3 (AC3)â are considered. As an example of error management, a simulation of the AMR-WB+ codec is included. The simulation allows an evaluation of the mechanisms included in the codec definition and enables also an evaluation of the different bit error sensitivities of the encoded audio payload.IngenierĂa TĂ©cnica en TelemĂĄtic
Audio Coding Based on Integer Transforms
Die Audiocodierung hat sich in den letzten Jahren zu einem sehr
populÀren Forschungs- und Anwendungsgebiet entwickelt. Insbesondere
gehörangepasste Verfahren zur Audiocodierung, wie etwa MPEG-1 Layer-3
(MP3) oder MPEG-2 Advanced Audio Coding (AAC), werden hÀufig zur
effizienten Speicherung und Ăbertragung von Audiosignalen verwendet. FĂŒr
professionelle Anwendungen, wie etwa die Archivierung und Ăbertragung im
Studiobereich, ist hingegen eher eine verlustlose Audiocodierung angebracht.
Die bisherigen AnsĂ€tze fĂŒr gehörangepasste und verlustlose
Audiocodierung sind technisch völlig verschieden. Moderne
gehörangepasste Audiocoder basieren meist auf FilterbÀnken, wie etwa der
ĂŒberlappenden orthogonalen Transformation "Modifizierte Diskrete
Cosinus-Transformation" (MDCT). Verlustlose Audiocoder hingegen
verwenden meist prÀdiktive Codierung zur Redundanzreduktion. Nur wenige
AnsÀtze zur transformationsbasierten verlustlosen Audiocodierung wurden
bisher versucht.
Diese Arbeit prÀsentiert einen neuen Ansatz hierzu, der das
Lifting-Schema auf die in der gehörangepassten Audiocodierung
verwendeten ĂŒberlappenden Transformationen anwendet. Dies ermöglicht
eine invertierbare Integer-Approximation der ursprĂŒnglichen
Transformation, z.B. die IntMDCT als Integer-Approximation der MDCT. Die
selbe Technik kann auch fĂŒr FilterbĂ€nke mit niedriger Systemverzögerung
angewandt werden. Weiterhin ermöglichen ein neuer, mehrdimensionaler
Lifting-Ansatz und eine Technik zur Spektralformung von
Quantisierungsfehlern eine Verbesserung der Approximation der
ursprĂŒnglichen Transformation.
Basierend auf diesen neuen Integer-Transformationen werden in dieser
Arbeit neue Verfahren zur Audiocodierung vorgestellt. Die Verfahren
umfassen verlustlose Audiocodierung, eine skalierbare verlustlose
Erweiterung eines gehörangepassten Audiocoders und einen integrierten
Ansatz zur fein skalierbaren gehörangepassten und verlustlosen
Audiocodierung. SchlieĂlich wird mit Hilfe der Integer-Transformationen
ein neuer Ansatz zur unhörbaren Einbettung von Daten mit hohen
Datenraten in unkomprimierte Audiosignale vorgestellt.In recent years audio coding has become a very popular field for
research and applications. Especially perceptual audio coding schemes,
such as MPEG-1 Layer-3 (MP3) and MPEG-2 Advanced Audio Coding (AAC), are
widely used for efficient storage and transmission of music
signals. Nevertheless, for professional applications, such as archiving
and transmission in studio environments, lossless audio coding schemes
are considered more appropriate.
Traditionally, the technical approaches used in perceptual and lossless
audio coding have been separate worlds. In perceptual audio coding, the
use of filter banks, such as the lapped orthogonal transform "Modified
Discrete Cosine Transform" (MDCT), has been the approach of choice being
used by many state of the art coding schemes. On the other hand,
lossless audio coding schemes mostly employ predictive coding of
waveforms to remove redundancy. Only few attempts have been made so far
to use transform coding for the purpose of lossless audio coding.
This work presents a new approach of applying the lifting scheme to
lapped transforms used in perceptual audio coding. This allows for an
invertible integer-to-integer approximation of the original transform,
e.g. the IntMDCT as an integer approximation of the MDCT. The same
technique can also be applied to low-delay filter banks. A generalized,
multi-dimensional lifting approach and a noise-shaping technique are
introduced, allowing to further optimize the accuracy of the
approximation to the original transform.
Based on these new integer transforms, this work presents new audio
coding schemes and applications. The audio coding applications cover
lossless audio coding, scalable lossless enhancement of a perceptual
audio coder and fine-grain scalable perceptual and lossless audio
coding. Finally an approach to data hiding with high data rates in
uncompressed audio signals based on integer transforms is described
Frequency-warped autoregressive modeling and filtering
This thesis consists of an introduction and nine articles. The articles are related to the application of frequency-warping techniques to audio signal processing, and in particular, predictive coding of wideband audio signals. The introduction reviews the literature and summarizes the results of the articles.
Frequency-warping, or simply warping techniques are based on a modification of a conventional signal processing system so that the inherent frequency representation in the system is changed. It is demonstrated that this may be done for basically all traditional signal processing algorithms. In audio applications it is beneficial to modify the system so that the new frequency representation is close to that of human hearing. One of the articles is a tutorial paper on the use of warping techniques in audio applications.
Majority of the articles studies warped linear prediction, WLP, and its use in wideband audio coding. It is proposed that warped linear prediction would be particularly attractive method for low-delay wideband audio coding. Warping techniques are also applied to various modifications of classical linear predictive coding techniques. This was made possible partly by the introduction of a class of new implementation techniques for recursive filters in one of the articles. The proposed implementation algorithm for recursive filters having delay-free loops is a generic technique. This inspired to write an article which introduces a generalized warped linear predictive coding scheme. One example of the generalized approach is a linear predictive algorithm using almost logarithmic frequency representation.reviewe
Audio Coding Based on Integer Transforms
Die Audiocodierung hat sich in den letzten Jahren zu einem sehr
populÀren Forschungs- und Anwendungsgebiet entwickelt. Insbesondere
gehörangepasste Verfahren zur Audiocodierung, wie etwa MPEG-1 Layer-3
(MP3) oder MPEG-2 Advanced Audio Coding (AAC), werden hÀufig zur
effizienten Speicherung und Ăbertragung von Audiosignalen verwendet. FĂŒr
professionelle Anwendungen, wie etwa die Archivierung und Ăbertragung im
Studiobereich, ist hingegen eher eine verlustlose Audiocodierung angebracht.
Die bisherigen AnsĂ€tze fĂŒr gehörangepasste und verlustlose
Audiocodierung sind technisch völlig verschieden. Moderne
gehörangepasste Audiocoder basieren meist auf FilterbÀnken, wie etwa der
ĂŒberlappenden orthogonalen Transformation "Modifizierte Diskrete
Cosinus-Transformation" (MDCT). Verlustlose Audiocoder hingegen
verwenden meist prÀdiktive Codierung zur Redundanzreduktion. Nur wenige
AnsÀtze zur transformationsbasierten verlustlosen Audiocodierung wurden
bisher versucht.
Diese Arbeit prÀsentiert einen neuen Ansatz hierzu, der das
Lifting-Schema auf die in der gehörangepassten Audiocodierung
verwendeten ĂŒberlappenden Transformationen anwendet. Dies ermöglicht
eine invertierbare Integer-Approximation der ursprĂŒnglichen
Transformation, z.B. die IntMDCT als Integer-Approximation der MDCT. Die
selbe Technik kann auch fĂŒr FilterbĂ€nke mit niedriger Systemverzögerung
angewandt werden. Weiterhin ermöglichen ein neuer, mehrdimensionaler
Lifting-Ansatz und eine Technik zur Spektralformung von
Quantisierungsfehlern eine Verbesserung der Approximation der
ursprĂŒnglichen Transformation.
Basierend auf diesen neuen Integer-Transformationen werden in dieser
Arbeit neue Verfahren zur Audiocodierung vorgestellt. Die Verfahren
umfassen verlustlose Audiocodierung, eine skalierbare verlustlose
Erweiterung eines gehörangepassten Audiocoders und einen integrierten
Ansatz zur fein skalierbaren gehörangepassten und verlustlosen
Audiocodierung. SchlieĂlich wird mit Hilfe der Integer-Transformationen
ein neuer Ansatz zur unhörbaren Einbettung von Daten mit hohen
Datenraten in unkomprimierte Audiosignale vorgestellt.In recent years audio coding has become a very popular field for
research and applications. Especially perceptual audio coding schemes,
such as MPEG-1 Layer-3 (MP3) and MPEG-2 Advanced Audio Coding (AAC), are
widely used for efficient storage and transmission of music
signals. Nevertheless, for professional applications, such as archiving
and transmission in studio environments, lossless audio coding schemes
are considered more appropriate.
Traditionally, the technical approaches used in perceptual and lossless
audio coding have been separate worlds. In perceptual audio coding, the
use of filter banks, such as the lapped orthogonal transform "Modified
Discrete Cosine Transform" (MDCT), has been the approach of choice being
used by many state of the art coding schemes. On the other hand,
lossless audio coding schemes mostly employ predictive coding of
waveforms to remove redundancy. Only few attempts have been made so far
to use transform coding for the purpose of lossless audio coding.
This work presents a new approach of applying the lifting scheme to
lapped transforms used in perceptual audio coding. This allows for an
invertible integer-to-integer approximation of the original transform,
e.g. the IntMDCT as an integer approximation of the MDCT. The same
technique can also be applied to low-delay filter banks. A generalized,
multi-dimensional lifting approach and a noise-shaping technique are
introduced, allowing to further optimize the accuracy of the
approximation to the original transform.
Based on these new integer transforms, this work presents new audio
coding schemes and applications. The audio coding applications cover
lossless audio coding, scalable lossless enhancement of a perceptual
audio coder and fine-grain scalable perceptual and lossless audio
coding. Finally an approach to data hiding with high data rates in
uncompressed audio signals based on integer transforms is described
Biorthogonality in lapped transforms : a study in high-quality audio compression
Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1996.Includes bibliographical references (leaves 76-82).by Shiufun Cheung.Ph.D
Bi & tri dimensional scene description and composition in the MPEG-4 standard
MPEG-4 is a new ISO/IEC standard being developed by MPEG (Moving Picture Experts Group). The standard is to be released in November 1998 and version 1 will be an International Standard in January 1999 The MPEG-4 standard addresses the new demands that arise in a world in which more and more audio-visual material is exchanged in digital form MPEG-4 addresses the coding of objects of various types. Not only traditional video and audio frames, but also natural video and audio objects as well as textures, text, 2- and 3-dimensional graphic primitives, and synthetic music
and sound effects.
Using MPEG-4 to reconstruct an audio-visual scene at a terminal, it is hence no longer sufficient to encode the raw audio-visual data and transmit it, as MPEG-2 does m order to synchronize video and audio. In MPEG-4, all objects are multiplexed together at the encoder and transported to the terminal Once de-multiplexed, these objects are composed at the terminal to construct and present to the end user a meaningful audio-visual scene. The placement of these elementary audio-visual objects in space and time is described in the scene description of a scene. While the action of putting these objects together in the same representation space is the composition of audio-visual objects.
My research was concerned with the scene description and composition of the audio-visual objects that are defined in an audio-visual scene Scene descriptions are coded independently irom sticams related to primitive audio-visual objects. The set of parameters belonging to the scene description are differentiated from the parameters that are used to improve the coding efficiency of an object. While the independent coding of different objects may achieve a higher compression rate, it also brings the ability to manipulate content at the terminal. This allows the modification of the scene description parameters without having to decode the primitive audio-visual objects themselves. This approach allows the development of a syntax that describes the spatio-temporal relationships of audio-visual scene objects. The behaviours of objects and their response to user inputs can thus also be represented in the scene description, allowing richer audio-visual content to be delivered as an MPEG-4 stream