In packet networks including the Internet and commercial 3G wireless bearers, the network states that a streaming media application experiences are not known a priori and exhibit time-varying characteristics. For such dynamic environments, network-adaptive techniques are essential to efficiently deliver video data. In this paper, we propose a frame-based optimal scheduling algorithm which incorporates a MAP (Maximum A Posteriori) framework for adapting to varying network loss rate. The optimal transmission schedule is determined such that effective frame-rate is maximized at playback. Also, for multiple packets per frame, frame-based selection of delivery order greatly reduces computational complexity for a server scheduler when compared with packet-based scheduling techniques. In addition, by dynamically estimating instantaneous packet loss probability, the proposed scheduler performs network-adaptive transmission for streaming video over time-varying packet networks. Simulation results for test video sequence show that the proposed scheduling algorithm outperforms conventional ARQ-based schemes from a view point of reconstructed video quality as well as playable frame-rate. In particular, the proposed scheduling algorithm exhibits significant improvements of frame-rate over highly lossy channels.

This paper proposes a packet loss recovery method using packets arrived behind the playout time for CELP (Code Excited Liner Prediction) decoding. The proposed method recovers synchronization of the filter states between encoding and decoding in the period following packet loss. The recovery is performed by replacing the degraded filter states with the ones calculated from the late arrival packet in decoding. When the proposed method is applied to the AMR (Adaptive Multi-Rate) speech decoder, it improves the segmental SNR (Signal-to-Noise Ratio) by 0.2 to 1.8 dB at packet loss rates of 2 to 10 % in case that all the packet losses occur due to their late arrival. PESQ (Perceptual Evaluation of Speech Quality) results also show that the proposed method slightly improves the speech quality. The subjective test results show that five-grade mean opinion scores are improved by 0.35 and 0.28 at a packet loss rate of 5 % at speech coding bitrates of 7.95 and 12.2 kbit/s, respectively.