Markus Kaindl

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When transmitting speech signals, residual redundancy is still left in the signal after source coding, due to limited complexity of the coding algorithms and delay constraints. This redundancy expresses in correlations inside one frame as well as in a time correlation of subsequent speech frames. The method of iterative channel and source decoding applied(More)
A very flexible transmission system for voice over mobile internet is proposed. With mobile internet a combination of a lossy packet-switched network and a wireless link of a mobile network is assumed. The new system allows to adaptively change the speech codec rate, and, thus, the payload of a packet, the number of transmitted packets and the number of(More)
This paper describes an adaptive multi-rate wideband (AMR-WB) speech codcc proposcd for the GSM system and also for the evolving Third Generation (3G) mobile speech services. The speech codec is based on SB-CELP (Splitband-Code-Excited Linear Prediction) with five modes operating bit rates from 24kbit/s down to S.lkbit/s. The respective channel coding(More)
A very flexible transmission system for multimedia over mobile in-ternet is proposed. For a mobile internet scenario a combination of a lossy packet-switched network and a mobile network is assumed. Conventional solutions are based on a separation of both networks. The new system is able to manage both packet-losses and bit errors in a single decoding step(More)
We will present an algorithm for enhanced AMR mode switching for VoIP over GERAN that takes into account abstract criteria like frame delay and packet loss rate at the application layer, as well as segmentation overhead at the data link layer. With the proposed algorithm in a wireless terminal, dynamic cross-layer optimization between the VoIP application(More)
— A very flexible transmission system for multimedia over mobile internet is proposed. For a mobile internet scenario a combination of a lossy packet data network and a mobile network is assumed. Conventional solutions are based on a separation of both networks. The new system is able to manage both packet-losses and bit errors in a single decoding step(More)
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