3:30, MMEDIA-L1.1
A STUDY ON THE APPLICATION OF AN AMR SPEECH CODEC TO VOIP
J. SEO, S. WOO, K. BAE
Degradation of speech quality caused by packet loss of voice traffic is still one of critical technical barriers of VoIP system. We propose a new VoIP system that can adapt transmission bit rate flexibly to network conditions to reduce packet loss. In order to determine the transmission bit rate depending upon the network conditions on a frame basis, we use the timestamp parameter in the RTP of H.323 protocol. Experimental results demonstrate that the proposed system is very promising to reduce packet loss that leads to improvement of speech quality.

3:50, MMEDIA-L1.2
JOINT SOURCE-CHANNEL CODING FOR SCALABLE VIDEO USING MODELS OF RATE-DISTORTION FUNCTIONS
L. KONDI, A. KATSAGGELOS
A joint source-channel coding scheme for scalable video is developed in this paper. An SNR scalable video coder is used and Unequal Error Protection (UEP) is allowed for each scalable layer. Our problem is to allocate the available bit rate across scalable layers and, within each layer, between source and channel coding, while minimizing the end-to-end distortion of the received video sequence. The resulting optimization algorithm we propose utilizes universal rate-distortion characteristic plots. These plots show the contribution of each layer to the total distortion as a function of the source rate of the layer and the residual bit error rate (the error rate that remains after the use of channel coding). Models for these plots are proposed in order to reduce the computational complexity of the solution. Experimental results demonstrate the effectiveness of the proposed approach.

4:10, MMEDIA-L1.3
ACCURATE ESATIMATION OF R-D CHARACTERISTICS FOR RATE CONTROL IN REAL-TIME VIDEO ENCODING
J. BAI, C. FENG, Q. LIAO, X. LIN, X. ZHUANG
In real-time video communications, the rate control strategies must be utilized to satisfy the end-to-end delay and prevent the encoding buffer from over/underflow. In other words, to acquire the best possible video quality with a minimal quality variation in the playback video, an accurate Rate-Distortion (R-D) model of the video source is critical in optimizing the bit allocation for video coding. In the paper, an Exponential Functions based R-D model is proposed for Intra-coded frames in video coding. Numerous experiments have consistently shown that the proposed model outperforms other popular R-D models in terms of both the estimation accuracy and computation complexity, making it suitable for rate control in real-time video coding.

4:30, MMEDIA-L1.4
OPTIMAL RATE ALLOCATION FOR VIDEO CODING AND TRANSMISSION OVER WIRELESS CHANNELS
J. SONG, K. LIU
An integrated framework of optimal rate allocation for video coding is presented in the case of transmission over wireless channels without feedback channels. For a fixed channel bit rate and finite number of channel coding rates, the proposed scheme can find the optimal source and channel coding pair and corresponding robust video coding scheme such that the expected end-to-end distortion of video signals can be minimized. With the assumption that encoder has the stochastic channel information, the proposed scheme takes into account robust video coding, channel coding and packetization, error concealment techniques altogether. Simulation results show the accuracy and optimality of the proposed method.

4:50, MMEDIA-L1.5
MOTION ADAPTIVE ERROR RESILIENT ENCODING FOR MPEG-4
S. WORRALL, A. SADKA, P. SWEENEY, A. KONDOZ
Optimising delivery of video codecs such as MPEG-4 is vital to ensure that acceptable quality can be offered to 3G network customers. The error prone nature of mobile channels means that the video codec that is employed must be fairly robust. In the past, such research has required alteration to existing standards. Given the imminent implementation of 3G technologies, changing the standards is not a convenient option. This paper presents a technique for altering MPEG-4 encoding parameters to increase the error robustness of the output bitstream. It exploits the fact that frames with a high degree motion are often more sensitive to error than those with low amounts of motion. Using a simple model, the video packet length and number of AIR blocks are varied according to the amount of motion in a frame. Simulations using a GPRS channel are presented to confirm the benefits of the proposed scheme.

5:10, MMEDIA-L1.6
SUBBAND BASED MPEG AUDIO MIXING FOR INTERNET STREAMING APPLICATIONS
P. DE SMET, T. STICHELE
In this paper we investigate a special-purpose application of MPEG-1 layer II audio streaming. First, we discuss how two or more already coded MPEG-audio bitstreams can be manipulated and mixed within the coded subband domain by using an appropriate algorithm for combining and recalculating the bit allocation. Next, we will show that very significant speedups can be obtained compared with mixing in the temporal domain. We will also illustrate that, at the same time, the system can act as a bit rate scaling device, using one of the bit allocation-combination algorithms which we implemented. The results obtained with these different algorithms are briefly discussed and compared. Finally, we report on how the developed software was integrated into an internet audio streaming system, which is now able to allow simple, yet efficient, real-time mixing of several MPEG-audio coded signals.