This article focuses on the compressed representations of the pictures

The quality of real-time voice communication over best-effort networks is mainly determined by the delay and loss characteristics observed along the network path. Excessive playout buffering at the receiver is prohibitive and significantly delayed packets have to be discarded and considered as late loss. We propose to improve the tradeoff among delay, late loss rate, and speech quality using multi-stream transmission of real-time voice over the Internet, where multiple redundant descriptions of the voice stream are sent over independent network paths. Scheduling the playout of the received voice packets is based on a novel multi-stream adaptive playout scheduling technique that uses a Lagrangian cost function to trade delay versus loss. Experiments over the Internet suggest largely uncorrelated packet erasure and delay jitter characteristics for different network paths which leads to a noticeable path diversity gain. We observe significant reductions in mean end-to-end latency and loss rates as well as improved speech quality when compared to FEC protected single-path transmission at the same data rate. In addition to our Internet measurements, we analyze the performance of the proposed multi-path voice communication scheme using the ns network simulator for different network topologies, including shared network links.

This paper examines the effectiveness of combining multiple description coding (MDC) and multiple path transport (MPT) for video and image transmission in a multihop mobile radio network. The video and image information is encoded nonhierarchically into multiple descriptions with the following objectives. The received picture quality should be acceptable, even if only one description is received and every additional received description contributes to enhanced picture quality. Typical applications will need a higher bandwidth/higher reliability connection than that provided by a single link in current mobile networks. For supporting these applications, a mobile node may need to set up and use multiple paths to the desired destination, either simply because of the lack of raw bandwidth on a single channel or because of its poor error characteristics, which reduce its effective throughput. In the context of this work, the principal reasons for considering such an architecture are providing high bandwidth and more robust end-to-end connections. We describe a protocol architecture that addresses this need and, with the help of simulations, we demonstrate the feasibility of this system and compare the performance of the MDC-MPT scheme to a system using layered coding and asymmetrical paths for the base and enhancement layers.

We propose multi-stream transmission of real-time voice over best-effort packet networks such as today's Internet, where multiple redundant descriptions of the voice stream are sent over independent network paths. At the receiver, multi-stream adaptive playout scheduling is employed to improve the tradeoff among delay, late loss rate, and speech quality. Experiments over the Internet suggest largely uncorrelated statistical characteristics, such as erasure probability and delay jitter, for different network paths, which leads to a noticeable path diversity gain. We have obtained significant reductions in mean end-to-end latency and loss rates compared to FEC protected single-path transmission at the same data rate. The speech quality perceived by the receiver is evaluated using the recently standardized objective quality measure PESQ. In our experiments, we observe gains of more than 0.4 PESQ score for voice transmission with packet path diversity.

This paper explores packet loss recovery for automatic speech recognition (ASR) in spoken dialog systems, assuming an architecture in which a lightweight client communicates with a remote ASR server. Speech is transmitted with source and channel codes optimized for the ASR application, i.e., to minimize word error rate. Unequal amounts of forward error correction, depending on the data's effect on ASR performance, are assigned to protect against packet loss. Experiments with simulated packet loss in a range of loss conditions are conducted on the DARPA Communicator (air travel information) task. Results show that the approach provides robust ASR performance which degrades gracefully as packet loss rates increase. Transmitting at 5.2 Kbps with up to 200 ms added delay, leads to only a 7% relative degradation in word error rate even under extremely adverse network conditions.

Most existing speech and audio coders were developed to meet a single purpose of delivering the best quality possible under fixed constraints in bit-rate, computational complexity, and algorithmic delay. Recent expansion in network communications demands the additional capability to cope with the packet-lossy nature associated with these networks. The problem lies within the research area of multiple description coding (MDC). In this report we investigate its essence, state its inherent limitations and tradeoffs, and propose a novel method to design efficient MDC systems for speech.

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 used carriers on the mobile link depending on the packet error rate of the packet-switched network, and the downlink quality of the wireless link in a very flexible way. The proposed system is based on the Adaptive Multi-Rate (AMR) speech codec and a systematic convolutional code. Packeting of the speech data is done in such a way, that a packet-loss can be assumed as puncturing of the convolutional code. Thus, the only information that must be transmitted to the decoder is a bad frame indication, defining whether a certain packet out of the set of transmitted packets has been received or not. In a coding experiment the performance of the proposed system is analyzed and compared to a reference system using the G.711 codec. The proposed system outperforms the reference system over a wide range of signal to noise ratios of the wireless link and the packet error rates of the packet-switched network.

With the increasing dominance of the transform coding technique virtually in every image and video coding schemes proposed up to date, e#cient transform coding system implementation has become an important research topic. This thesis addresses two system issues that may arise in practice: #i# e#cient architecture designs for the discrete wavelet transform; and #ii# e#cient transform coding for robust communication over erasure channels. The #rst contribution of the thesis is to develop an overlap-state technique for e#cientmultilevel wavelet decompositions when memory and delay constraints have to be strictly observed. In this case, the wavelet transform can be computed in a block-by-block fashion, i.e., the input data is segmented into blocks and each block is processed separately, either sequentially or in parallel. The proposed technique enables e#cient data exchange between consecutive data blocks such that the required memory bu#er size and#or communication overhead can be signi#...

Packet loss is a serious problem that severely affects the quality of multimedia streaming over error-prone networks. To reduce the variability of packet loss and delay, packets can be transmitted over different network paths (path diversity), after being coded by error-concealment source coding methods like Multiple Description Coding (MDC) or Layered Coding (LC). Researches in this area lead to a common belief that MDC is better than LC when the network conditions (packet loss rate, bandwidth) are grave [1, 2]. However, in this paper, we show that the decision of which packets to send over which paths can greatly affect the performance of LC and MDC, therefore the quality of the streams received. Particularly, using our analytical framework and polynomial algorithms for finding optimal packet allocations, we show that LC outperforms MDC under various critical network conditions.

A practical approach for the design of multiple-description scalar quantization of speech is presented that conforms to standard G.711 PCM. The method chiefly consists of an index assignment algorithm that enables the side decoders to exhibit SNR characteristics comparable to those of the standard logarithmic quantizer. With two-channel transmission of multiple descriptions, an increase in robustness to lossy channels is obtained without violation of the standard coding method. The method found is suitable for the design of multiple descriptions of any given scalar quantizer, e. g. one within a complex speech coder.