19 research outputs found

    ROBUST DECODING OF A 3D-ESCOT BITSTREAM TRANSMITTED OVER A NOISY CHANNEL

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    International audienceIn this paper, we propose a joint source-channel (JSC) decoding scheme for 3D ESCOT-based video coders, such as Vidwav. The embedded bitstream generated by such coders is very sensitive to transmission errors unavoidable on wireless channels. The proposed JSC decoder employs the residual redundancy left in the bitstream by the source coder combined with bit reliability information provided by the channel or channel decoder to correct transmission errors. When considering an AWGN channel, the performance gains are in average 4 dB in terms of PSNR of the reconstructed frames, and 0.7 dB in terms of channel SNR. When considering individual frames, the obtained gain is up to 15 dB in PSNR

    High Quality Audio Coding with MDCTNet

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    We propose a neural audio generative model, MDCTNet, operating in the perceptually weighted domain of an adaptive modified discrete cosine transform (MDCT). The architecture of the model captures correlations in both time and frequency directions with recurrent layers (RNNs). An audio coding system is obtained by training MDCTNet on a diverse set of fullband monophonic audio signals at 48 kHz sampling, conditioned by a perceptual audio encoder. In a subjective listening test with ten excerpts chosen to be balanced across content types, yet stressful for both codecs, the mean performance of the proposed system for 24 kb/s variable bitrate (VBR) is similar to that of Opus at twice the bitrate.Comment: Five pages, five figure

    Bit-error resilient packetization for streaming h.264/avc video

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    Survey of error concealment schemes for real-time audio transmission systems

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    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 “Adaptive Differential Pulse-Code Modulation (ADPCM)”, the “Audio Processing Technology Apt-x100”, the “Extended Adaptive Multi-Rate Wideband (AMR-WB+)”, the “Advanced Audio Coding (AAC)”, the “MPEG-1 Audio Layer II (MP2)”, the “MPEG-1 Audio Layer III (MP3)” and finally the “Adaptive Transform Coder 3 (AC3)” 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

    MPEG-4's BIFS-Anim protocol: using MPEG-4 for streaming of 3D animations

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    This thesis explores issues related to the generation and animation of synthetic objects within the context of MPEG-4. MPEG-4 was designed to provide a standard that will deliver rich multimedia content on many different platforms and networks. MPEG-4 should be viewed as a toolbox rather than as a monolithic standard as each implementer of the standard will pick the necessary tools adequate to their needs, likely to be a small subset of the available tools. The subset of MPEG-4 that will be examined here are the tools relating to the generation of 3D scenes and to the animation of those scenes. A comparison with the most popular 3D standard, Virtual Reality Modeling Language (VRML) will be included. An overview of the MPEG-4 standard will be given, describing the basic concepts. MPEG-4 uses a scene description language called Binary Format for Scene (BIFS) for the composition of scenes, this description language will be described. The potential for the technology used in BIFS to provide low bitrate streaming 3D animations will be analysed and some examples of the possible uses of this technology will be given. A tool for the encoding of streaming 3D animations will be described and results will be shown that MPEG-4 provides a more efficient way of encoding 3D data when compared to VRML. Finally a look will be taken at the future of 3D content on the Internet

    Seamless Multimedia Delivery Within a Heterogeneous Wireless Networks Environment: Are We There Yet?

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    The increasing popularity of live video streaming from mobile devices, such as Facebook Live, Instagram Stories, Snapchat, etc. pressurizes the network operators to increase the capacity of their networks. However, a simple increase in system capacity will not be enough without considering the provisioning of quality of experience (QoE) as the basis for network control, customer loyalty, and retention rate and thus increase in network operators revenue. As QoE is gaining strong momentum especially with increasing users' quality expectations, the focus is now on proposing innovative solutions to enable QoE when delivering video content over heterogeneous wireless networks. In this context, this paper presents an overview of multimedia delivery solutions, identifies the problems and provides a comprehensive classification of related state-of-the-art approaches following three key directions: 1) adaptation; 2) energy efficiency; and 3) multipath content delivery. Discussions, challenges, and open issues on the seamless multimedia provisioning faced by the current and next generation of wireless networks are also provided

    The specification and design of a prototype 2-D MPEG-4 authoring tool

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    The purpose of this project was the specification, design and implementation of a prototype 2-D MPEG-4 authoring tool. A literature study was conducted of the MPEG-4 standard and multimedia authoring tools to determine the specification and design of a prototype 2- D MPEG-4 authoring tool. The specification and design was used as a basis for the implementation of a prototype 2-D MPEG-4 authoring tool that complies with the Complete 2-D Scene Graph Profile. The need for research into MPEG-4 authoring tools arose from the reported lack of knowledge of the MPEG-4 standard and the limited implementations of MPEG-4 authoring tools available to content authors. In order for MPEG-4 to reach its full potential, it will require authoring tools and content players that satisfy the needs of its users. The theoretical component of this dissertation included a literature study of the MPEG-4 standard and an investigation of relevant multimedia authoring systems. MPEG-4 was introduced as a standard that allows for the creation and streaming of interactive multimedia content at variable bit rates over high and low bandwidth connections. The requirements for the prototype 2-D MPEG-4 authoring system were documented and a prototype system satisfying the requirements was designed, implemented and evaluated. The evaluation of the prototype system showed that the system successfully satisfied all its requirements and that it provides the user with an easy to use and intuitive authoring tool. MPEG-4 has the potential to satisfy the increasing demand for innovative multimedia content on low bandwidth networks, including the Internet and mobile networks, as well as the need expressed by users to interact with multimedia content. This dissertation makes an important contribution to the understanding of the MPEG-4 standard, its functionality and the design of a 2-D MPEG-4 Authoring tool. Keywords: MPEG-4; MPEG-4 authoring; Binary Format for Scenes

    Seamless multimedia delivery within a heterogeneous wireless networks environment: are we there yet?

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    The increasing popularity of live video streaming from mobile devices such as Facebook Live, Instagram Stories, Snapchat, etc. pressurises the network operators to increase the capacity of their networks. However, a simple increase in system capacity will not be enough without considering the provisioning of Quality of Experience (QoE) as the basis for network control, customer loyalty and retention rate and thus increase in network operators revenue. As QoE is gaining strong momentum especially with increasing users’ quality expectations, the focus is now on proposing innovative solutions to enable QoE when delivering video content over heterogeneous wireless networks. In this context, this paper presents an overview of multimedia delivery solutions, identifies the problems and provides a comprehensive classification of related state-of-the-art approaches following three key directions: adaptation, energy efficiency and multipath content delivery. Discussions, challenges and open issues on the seamless multimedia provisioning faced by the current and next generation of wireless networks are also provided

    Audio/Video Transmission over IEEE 802.11e Networks: Retry Limit Adaptation and Distortion Estimation

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    The objective of this thesis focuses on the audio and video transmission over wireless networks adopting the family of the IEEE 802.11x standards. In particular, this thesis discusses about the resolution of four issues: the adaptive retransmission, the comparison of video quality indexes for retry limit adaptation purposes, the estimation of the distortion and the joint adaptation of the maximum number of retransmissions of voice and video flows

    Étude de transformĂ©es temps-frĂ©quence pour le codage audio faible retard en haute qualitĂ©

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    In recent years there has been a phenomenal increase in the number of products and applications which make use of audio coding formats. Amongthe most successful audio coding schemes, the MPEG-1 Layer III (mp3), the MPEG-2 Advanced Audio Coding (AAC) or its evolution MPEG-4High Efficiency-Advanced Audio Coding (HE-AAC) can be cited. More recently, perceptual audio coding has been adapted to achieve codingat low-delay such to become suitable for conversational applications. Traditionally, the use of filter bank such as the Modified Discrete CosineTransform (MDCT) is a central component of perceptual audio coding and its adaptation to low delay audio coding has become an important researchtopic. Low delay transforms have been developed in order to retain the performance of standard audio coding while reducing dramatically the associated algorithmic delay.This work presents some elements allowing to better accommodate the delay reduction constraint. Among the contributions, a low delay blockswitching tool which allows the direct transition between long transform and short transform without the insertion of transition window. The sameprinciple has been extended to define new perfect reconstruction conditions for the MDCT with relaxed constraints compared to the original definition.As a consequence, a seamless reconstruction method has been derived to increase the flexibility of transform coding schemes with the possibility toselect a transform for a frame independently from its neighbouring frames. Finally, based on this new approach, a new low delay window design procedure has been derived to obtain an analytic definition for a new family of transforms, permitting high quality with a substantial coding delay reduction. The performance of the proposed transforms has been thoroughly evaluated, an evaluation framework involving an objective measurement of the optimal transform sequence is proposed. It confirms the relevance of the proposed transforms used for audio coding. In addition, the new approaches have been successfully applied to the recent standardisation work items, such as the low delay audio coding developed at MPEG (LD-AAC and ELD-AAC) and they have been evaluated with numerous subjective testing, showing a significant improvement of the quality for transient signals. The new low delay window design has been adopted in G.718, a scalable speech and audio codec standardized in ITU-T and has demonstrated its benefit in terms of delay reduction while maintaining the audio quality of a traditional MDCT.Codage audio Ă  faible retard Ă  l'aide de la dĂ©finition de nouvelles fenĂȘtres pour la transformĂ©e MDCT et l'introduction d'un nouveau schĂ©ma de commutation de fenĂȘtre
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