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  FlexCode Public Seminar, 16.06.2008, 13.30-15.00,
Nokia's building Hermia 5, Auditorium A156, Visiokatu 1, Tampere, Finland
 
 

Title: Turbo Source Coding
Speaker: Laurent Schmalen
Abstract:
During the last decade, it has been found that channel codes employing iterative decoders like, e.g., Turbo or LDPC codes, allow near-Shannon limit transmission of binary data with moderate computational complexity. It can be shown that these codes can also be effectively used for lossless or near-lossless source coding. In our presentation we will first review the basics of iteratively decodable codes and show then how these codes can be used for source coding. Afterwards, the concept of soft decision source decoding is presented which enables the exploitation of the statistical properties of the source in order to generate extrinsic information necessary for the use in iterative receivers. It is finally shown how the soft decision source decoder can be utilized in an iterative framework in order to realize near-lossless source coding. It is furthermore shown that the entropy of the source can be closely reached and that the system automatically incorporated channel coding and becomes very flexible in varying channel conditions.

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Title: Flexible Multiple Description Coding of Audio
Speaker: Janusz Klejsa
Abstract:
Multiple-description coding (MDC) addresses the ubiquitous problem of packet loss in packet networks by exploiting diversity. The capacity and packet loss rate of modern communication networks is highly variable, both in time and between networks. Existing MDC methods generally do not reflect this variability: the design of an MDC for a particular rate and packet loss environment requires off-line optimization and is, therefore, not suitable for a flexible coder. Moreover the application of MDC to an audio coding scheme that employs signal modeling introduces yet another problem of finding the optimal rate allocation between the model and the descriptions. This talk will focus on the application of a flexible MDC in a scalable audio coder. Our objective is a coding scheme that adapts in an on-line fashion to a specified rate and packet-loss probability. This includes finding the optimal rate allocation between the model and the descriptions. The goal for the MDC is to react to changing channel conditions by adapting the level of redundancy. The rate allocation problems are treated analytically, leading to a flexible multiple-description audio coder.

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