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There is an increasing interest in multipoint conferencing but service providers face the challenge of complexity when scaling to thousands of users. This problem can be resolved by a scalable architecture based on central media routers, which allows for low complexity server components and therefore low operational cost. In this paper we describe an audio routing approach using advantages offered by the AAC-ELD bit stream structure. By estimating the signal energy in the bit stream domain, we can detect active speakers at low complexity, which results in substantial bitrate and complexity reduction. Subjective tests with mediated conversations among four participants are conducted to compare the perceived audio quality of the audio router to conventional mixing in a conference bridge. The results show a reduction of complexity by one order of magnitude while maintaining the same subjective audio quality when forwarding the two most active speakers on a 10 ms framing.

Low-cost video conferencing systems have provided an existence proof for the value of video communication in a home setting. At the same time, current systems have a number of fundamental limitations that inhibit more general social interactions among multiple groups of participants. In our work, we describe the development, implementation and evaluation of a domestic video conferencing system that is geared to providing true 'togetherness' among conference participants. We show that such interactions require sophisticated support for high-quality audiovisual presentation, and processing support for person identification and localisation. In this paper, we describe user requirements for effective interpersonal interaction. We then report on a system that implements these requirements. We conclude with a systems and user evaluation of this work. We present results that show that participants in a video conference can be made feel as 'together' as collocated players of a board game.

Hierarchical B-Frames (HBF) has emerged as an efficient video coding tool in recent years. As shown in the literature, this approach results in excellent PSNR gains of >;1 dB. However these PSNR gains are not sufficiently assessed in a scientific manner by subjective tests. Hence in this paper, we evaluate HBF coding pattern subjectively by using the MUSHRA test methodology. While MUSHRA is well established in audio coding research, its application to video is a novelty of this paper. We compare HBF with simple IPP coding pattern at either same PSNR or same bit rate. Our results indicate that, HBF gains are clearly subjectively perceptible. Hence, it can be shown that PSNR gains also correlate with a subjective gain. Interestingly, even at same PSNR, HBF is found to be subjectively superior to simple IPP coding.

In this paper we present a bit allocation approach based on motion vector analysis for improved rate-distortion performance. Bit allocation is done at the macroblock level such that the macroblock with high priority is coded finely and the one with low priority is coded coarsely. In order to calculate macroblock priorities, first reference counts for each pixel are determined through motion vector analysis. A reference count of a pixel is defined as total number of pixels in the remaining GOP which use that pixel as a reference. Then macroblock wise reference counts is obtained by summing the pixel wise reference counts, which are then scaled and prioritized. Based upon the priority value the given fixed quant of each frame is modulated at the macroblock level. The algorithm is applied to H.264/AVC encoding and PSNR gains of up to 1.4 dB are achieved

In this paper we propose a rate-distortion optimized multipass algorithm for video encoding. The problem of dependent quantization in a hybrid coding framework is addressed by constructing a trellis from the exponentially growing search tree which is then searched using a Lagrangian cost function. To avoid exponential growth, states with similarly distorted reference frames are merged based on the assumption that similar rate-distortion behavior results for future frames. The goal is to select an optimal quantizer for each frame such that the overall R-D performance is optimized. The algorithm is applied to MPEG-4 SP and typical average PSNR gains of 0.3 dB are achieved. A comparison with the DivX 5.2 codec shows PSNR gains of up to 2 dB.

The quality of service limitation of today's Internet is a major challenge for real-time voice communications. Excessive delay, packet loss, and high delay jitter all impair the communication quality. A new receiver-based playout scheduling scheme is proposed to improve the tradeoff between buffering delay and late loss for real-time voice communication over IP networks. In this scheme the network delay is estimated from past statistics and the playout time of the voice packets is adaptively adjusted. In contrast to previous work, the adjustment is not only performed between talkspurts, but also within talkspurts in a highly dynamic way. Proper reconstruction of continuous playout speech is achieved by scaling individual voice packets using a time-scale modification technique based on the Waveform Similarity Overlap-Add (WSOLA) algorithm. Results of subjective listening tests show that this operation does not impair audio quality, since the adaptation process requires infrequent scaling of the voice packets and low playout jitter is perceptually tolerable. The same time-scale modification technique is also used to conceal packet loss at very low delay, i.e., one packet time. Simulation results based on Internet measurements show that the tradeoff between buffering delay and late loss can be improved significantly. The overall audio quality is investigated based on subjective listening tests, showing typical gains of 1 on a 5-point scale of the Mean Opinion Score.

A new receiver-based playout scheduling scheme is proposed, which estimates the network delay from past statistics and adaptively adjusts the playout time of the voice packets. In contrast to previous work, the adjustment is not only performed in between talkspurts, but also within the talkspurts in a highly dynamic way. Proper reconstruction of continuous output speech is achieved by scaling individual voice packets using a time-scale modification technique which modifies the rate of playout while preserving voice pitch. Subjective listening tests show that this operation does not impair the audio quality. Simulation results based on Internet measurement indicate that buffering delay and loss rate can be significantly reduced by adaptive scheduling.

Network variations in video streaming require sufficient data to be prebuffered at the client prior to playout. This receiver buffer prevents the display process from starvation in case of network congestion. Prebuffering, however, is also responsible for the major part of the delay between requesting a media stream and playing it at the receiver. We show how adaptive media playout can be employed to reduce the delay introduced by the receiver buffer while preserving the same resilience against buffer underflow as in non-adaptive media playout. For adaptive media playout we adjust the playout speed of the media packets depending on the condition of the channel and the current client buffer fullness. We employ a two-state Markov channel model to analyze the buffer underflow-delay trade-off for our adaptive playout strategy and show that for typical parameters the average end-to-end delay can be reduced by 1 to 2 seconds.

An analytical model of a video transmission system is used to study the impact of its parameters. The optimal trade-off between error resilience using INTRA coding and the channel code rate is determined for given channel parameters by minimizing the expected mean square error at the decoder. It is shown that the results strongly depend on the burstiness of the channel.

Long-term memory prediction extends the spatial displacement vector utilized in hybrid video coding by a variable time delay permitting the use of more than one reference frame for motion compensation. This extension provides improved rate-distortion performance. However, motion compensation in combination with transmission errors leads to temporal error propagation that occurs when the reference frames at encoder and decoder differ. In this paper, we present a framework that incorporates an error estimate into rate-constrained motion estimation and mode decision. Experimental results with a Rayleigh fading channel show that long-term memory motion compensation significantly outperforms single-frame prediction.

A theoretical analysis of the overall mean squared error (MSE) in hybrid video coding is presented for the case of error prone transmission. Our model covers the complete transmission system including the rate-distortion performance of the video encoder, forward error correction, interleaving, and the effect of error concealment and interframe error propagation at the video decoder. The channel model used is a 2-state Markov model describing burst errors on the symbol level. Reed-Solomon codes are used for forward error correction. Extensive simulation results using an H.263 video codec are provided for verification. Using the model, the optimal tradeoff between INTRA and INTER coding as well as the optimal channel code rate can be determined for given channel parameters by minimizing the expected MSE at the decoder. The main focus of this paper is to show the accuracy of the derived analytical model and its applicability to the analysis and optimization of an entire video transmission system.

Long-term memory prediction extends the spatial displacement vector utilized in hybrid video coding by a variable time delay, permitting the use of more than one reference frame for motion compensation. This extension leads to improved rate-distortion performance. However, motion compensation in combination with transmission errors leads to temporal error propagation that occurs when the reference frames at the coder and decoder differ. In this paper, we present a framework that incorporates an estimated error into rate-constrained motion estimation and mode decision. Experimental results with a Rayleigh fading channel show that long-term memory prediction significantly outperforms the single-frame prediction H.263-based anchor. When a feedback channel is available, the decoder can inform the encoder about successful or unsuccessful transmission events by sending positive (ACK) or negative (NACK) acknowledgments. This information is utilized for updating the error estimates at the encoder. Similar concepts, such as the ACK and NACK mode known from the H.263 standard, are unified into a general framework providing superior transmission performance.

A theoretical analysis of the overall mean squared error (MSE) in hybrid video coding is presented for the case of error prone transmission. The derived model for interframe error propagation includes the effects of INTRA coding and spatial loop filtering and corresponds to simulation results very accurately. For a given target bit rate, only four parameters are necessary to describe the overall distortion behavior of the decoder. Using the model, the optimal trade-off between INTRA and INTER coding can be determined for a given packet loss probability by minimizing the expected MSE at the decoder.

We review feedback-based low bit-rate video coding techniques for robust transmission in mobile multimedia networks. For error control on the source coding level, each decoder has to make provisions for error detection, resynchronization, and error concealment, and we review techniques suitable for that purpose. Further, techniques are discussed for intelligent processing of acknowledgment information by the coding control to adapt the source coder to the channel. We review and compare error tracking, error confinement, and reference picture selection techniques for channel-adaptive source coding. For comparison of these techniques, a system for transmitting low bit-rate video over a wireless channel is presented and the performance is evaluated for a range of transmission conditions. We also show how feedback-based source coding can be employed in conjunction with precompressed video stored on a media server. The techniques discussed are applicable to a wide variety of interframe video schemes, including various video coding standards. Several of the techniques have been incorporated into the H.263 video compression standard, and this standard is used as an example throughout.

For interactive video storage applications that retrieve video data over low bit-rate networks, precoded bit-streams have to be transmitted to the receiver with low latency. Besides random access, fast forward and reverse, the scalability of the bit-stream is an important requirement for the above application. In this paper, we investigate a standard compatible approach that avoids the overhead involved in the transmission of periodic I-frames. We propose to store multiple bit-streams of different rates at the video server, which can be selected for transmission. Especially encoded P-frames are used to switch between the stored bit-streams. During normal-speed playback only P-frames are transmitted, providing increased efficiency. The approach is evaluated in the framework of the H.263 video compression standard. Based on the Salesman test sequence encoded at 12.5 fps in QCIF format, the resulting frame jitter is investigated when an error control protocol is used to transmit the bit-streams over a wireless DECT channel. Compared to periodic I-frames at an interval of 800 ms, a gain of 2 dB PSNR can be observed at the same frame jitter.

In this paper we address the problem of robust video transmission in error prone environments. The approach is compatible with the ITU-T video coding standard H.263. Fading situations in mobile networks are tolerated and the image quality degradation due to spatio-temporal error propagation is minimized utilizing a feedback channel between transmitter and receiver carrying acknowledgment information. In a first step, corrupted group of blocks (GOB's) are concealed to avoid annoying artifacts caused by decoding of an erroneous bit stream. The GOB and the corresponding frame number are reported to the transmitter via the back channel. The encoder evaluates the negative acknowledgments and reconstructs the spatial and temporal error propagation. A low complexity algorithm for real-time reconstruction of spatio-temporal error propagation is described in detail. Rapid error recovery is achieved by INTRA refreshing image regions (macroblocks) bearing visible distortion. The feedback channel method does not introduce additional delay and is particularly relevant for real-time conversational services in mobile networks. Experimental results with bursty bit error sequences simulating a Digital European Cordless Telephony (DECT) channel are presented with different combinations of forward error correction (FEC), automatic repeat on request (ARQ), and the proposed error compensation technique, Compared to the case where FEC and ARQ are used for error correction, a gain of up to 3 dB peak signal-to-noise ratio (PSNR) is observed if error compensation is employed additionally.