Playout buffers are used in VoIP systems to compensate for network delay jitter by making a trade-off between delay and loss. In this work we propose a playout buffer algorithm that makes the trade-off based on maximization of conversational speech quality, aiming to keep the computational complexity lowest possible. We model the network delay using a Pareto distribution and show that it is a g...

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 i...

In media transmission over packet networks, one of the most challenging issues is the avoidance of receiver buffer underflows. Among the approaches proposed to solve this problem, the source rate adaptation is promising, due to the availability of multi-rate encoders. In order to succeed, rate-adaptive approaches should take into account not only the network throughput, but also the playout buf...

In coping with best-effort service, many VoIP applications employ adaptive playout strategies. Objective methods of speech quality assessment such as the ITU-T Recommendation P.862 (also known as Perceptual Evaluation of Speech Quality PESQ) typically do not capture distortion due to playout adjustments as they match up short segments prior to analysis. Similarly, the ITU-T E-Model does not cap...

Monte Carlo Tree Search (MCTS) is the state of the art algorithm for General Game Playing (GGP). We propose to learn a playout policy online so as to improve MCTS for GGP. We also propose to learn a policy not only using the moves but also according to the features of the moves. We test the resulting algorithms named Playout Policy Adaptation (PPA) and Playout Policy Adaptation with move Featur...

Perceived conversational speech quality is a key quality of service (QoS) metric for voice over IP (VoIP) applications. Speech quality is mainly affected by network impairments, such as delay, jitter and packet loss. Playout buffer algorithms are used to compensate for jitter, based on a tradeoff between delay and loss, but can have a significant effect on perceived quality. The main aim in thi...

Streaming media systems buffer media data at the client to improve reconstruction quality at the cost of latency. Reconstruction quality may be further improved with rate-distortion optimized packet transmission. In this paper we show how the tradeoff between buffering latency and reconstruction quality can be improved with adaptive media playout (AMP) the client-controlled manipulation of the ...

The MPEG-4 Enhanced Low Delay AAC (AAC-ELD) codec extends the application area of the Advanced Audio Coding (AAC) family towards high quality conversational services. Through the support of the full audio bandwidth at low delay and low bit rate, it offers excellent support for enhanced VoIP applications. In this paper we provide a brief overview of the AAC-ELD codec and describe how its codec s...