Mitchell Trott

Path diversity for enhanced media streaming

Media streaming over best effort packet networks such as the Internet is quite challenging because of the dynamic and unpredictable available bandwidth, loss rate, and delay. Recently, streaming over multiple paths to provide path diversity has emerged as an approach to help overcome these problems. This article provides an overview of the benefits and use of path diversity for media streaming. The different approaches to media coding and streaming over multiple paths are examined, together with architectures for achieving path diversity between single or multiple senders and a single receiver. Important examples include using the distributed servers in a content delivery network to provide path diversity to a requesting client, using multiple 802.11 wireless access points to provide path diversity to a mobile client, and using relays to provide low-latency media communication. The design, analysis, and operation of media streaming systems that use path diversity are considered, with emphasis on the accurate performance models needed to select the best paths or best servers.

Rateless Coding for Gaussian Channels

A rateless code-i.e., a rate-compatible family of codes-has the property that codewords of the higher rate codes are prefixes of those of the lower rate ones. A perfect family of such codes is one in which each of the codes in the family is capacity-achieving. We show by construction that perfect rateless codes with low-complexity decoding algorithms exist for additive white Gaussian noise channels. Our construction involves the use of layered encoding and successive decoding, together with repetition using time-varying layer weights. As an illustration of our framework, we design a practical three-rate code family. We further construct rich sets of near-perfect rateless codes within our architecture that require either significantly fewer layers or lower complexity than their perfect counterparts. Variations of the basic construction are also developed, including one for time-varying channels in which there is no a priori stochastic model.

The Road to Immersive Communication

Communication has seen enormous advances over the past 100 years including radio, television, mobile phones, video conferencing, and Internet-based voice and video calling. Still, remote communication remains less natural and more fatiguing than face-to-face. The vision of immersive communication is to enable natural experiences and interactions with remote people and environments in ways that suspend disbelief in being there. This paper briefly describes the current state-of-the-art of immersive communication, provides a vision of the future and the associated benefits, and considers the technical challenges in achieving that vision. The attributes of immersive communication are described, together with the frontiers of video and audio for achieving them. We emphasize that the success of these systems must be judged by their impact on the people who use them. Recent high-quality video conferencing systems are beginning to deliver a natural experience-when all participants are in custom-designed studios. Ongoing research aims to extend the experience to a broader range of environments. Augmented reality has the potential to make remote communication even better than being physically present. Future natural and effective immersive experiences will be created by drawing upon intertwined research areas including multimedia signal processing, computer vision, graphics, networking, sensors, displays and sound reproduction systems, haptics, and perceptual modeling and psychophysics.

Delay-Optimal Burst Erasure Code Construction

Delay-optimal burst erasure correction codes have the shortest possible decoding delay required to correct all bursts of a given length with a fixed redundancy. We provide a simple construction that yields linear convolutional delay-optimal burst erasure codes for any choice of coding parameters while keeping the finite field small.

Rateless space-time coding

Rateless codes are good codes of infinite length that have the property that prefixes of such codes are themselves good codes. This makes them attractive for applications in which the channel quality is uncertain, where systems transmit as much of a codeword as necessary for decoding to be possible. In particular, rateless codes are potentially attractive for wireless communication. In a recent work, a rateless coding scheme was proposed for the AWGN channel, based on layering, repetition and random dithering. We extend this scheme to multiple-input single-output (MISO) Gaussian channels. We show that the rate loss associated with orthogonal design space-time codes may be alleviated by layering and dithering, very similar to the rateless approach for the AWGN channel. We then combine the two schemes and arrive at a close-to-capacity rateless code for MISO channels. The required complexity depends on the fraction of capacity that is targeted, is linear in the capacity of the channel and does not depend on the number of transmit antennas. Furthermore, the coding scheme uses only one base AWGN code

Rateless Coding and Perfect Rate-Compatible Codes for Gaussian Channels

A rateless code, or a rate-compatible family of codes, has the property that the higher rate codes have codewords that are prefixes of those of the lower rate ones. A perfect family of such codes is one in which each of the codes in the family is capacity-achieving. We show by construction that perfect rateless codes with low-complexity decoding algorithms exist for additive white Gaussian noise channels. As an illustration of our framework, we design a practical three-rate code family. We further demonstrate that a rich set of perfect or near-perfect rateless codes may be found via numerical optimization

Zero-Delay Sequential Transmission of Markov Sources Over Burst Erasure Channels

A setup involving zero-delay sequential transmission of a vector Markov source over a burst erasure channel is studied. A sequence of source vectors is compressed in a causal fashion at the encoder, and the resulting output is transmitted over a burst erasure channel. The destination is required to reconstruct each source vector with zero-delay, but those source sequences that are observed either during the burst erasure, or in the interval of length W following the burst erasure need not be reconstructed. The minimum achievable compression rate is called the rate-recovery function. We assume that each source vector is independent identically distributed (i.i.d.) across the spatial dimension and is sampled from a stationary, first-order Markov process across the temporal dimension. For discrete sources, the case of lossless recovery is considered, and upper and lower bounds on the rate-recovery function are established. Both these bounds can be expressed as the rate for predictive coding, plus a term that decreases at least inversely with the recovery window length W. For Gauss-Markov sources and a quadratic distortion measure, upper and lower bounds on the minimum rate are established when W = 0. These bounds are shown to coincide in the high resolution limit. Finally, another setup involving i.i.d. Gaussian sources is studied and the raterecovery function is completely characterized in this case.

A system approach to acoustic echo cancellation in robust hands-free teleconferencing

This paper presents a system approach to the acoustic echo cancellation (AEC) problem in a noisy acoustic environment. We propose a method that makes use of the estimated near-end signal from a postfilter to further improve the AEC system performance. The cancellation performance is enhanced especially during strong near-end interference (e.g., double talk). Simulation results show that our stereophonic AEC based on the system approach with postfilter integration outperforms the one using the original robust AEC system by itself without postfilter integration. The improved performance is noted especially during double talk, where simulation results show that the true echo return loss enhancement can be boosted by as much as 10 dB.

A universal approach to queuing with distortion control

An efficient buffer-management algorithm is developed for queues that handle distortion-tolerant data under finite memory limitations. We avoid overflows and realize significant performance gains through the use of multiresolution source codes. These codes enable us to reduce the fidelity of signal descriptions in a controlled progressive manner. The proposed approach is universal, i.e., it works without knowledge of queue arrival and departure statistics. More strongly, we show that its performance is sample-path optimal, i.e., it achieves an average distortion equal to the best achievable by any algorithm, including those designed with full noncausal knowledge of queue arrival and service times.

Costs and benefits of fading for streaming media over wireless

In this tutorial article we discuss some salient aspects of the wireless channel, and the challenges and opportunities they pose for streaming media. Streaming media has a number of fundamental characteristics that differentiate it from generic data for which most wireless data transport protocols are designed. This means that solutions designed for media distribution in wired contexts do not directly translate into appropriate wireless solutions: designers must jointly optimize for both media and wireless characteristics. Through a number of examples we discuss how channel, source, and application-dependent concepts such as multiuser diversity, opportunistic communications, loss tolerance, and differing application-dependent delay constraints can be exploited to pose new and useful constructs for streaming media over wireless