The Experts below are selected from a list of 59193 Experts worldwide ranked by ideXlab platform
Bernd Girod - One of the best experts on this subject based on the ideXlab platform.
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adaptive playout scheduling and loss concealment for Voice Communication over ip networks
IEEE Transactions on Multimedia, 2003Co-Authors: Yi Liang, N Farber, Bernd GirodAbstract: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.
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real time Voice Communication over the internet using packet path diversity
ACM Multimedia, 2001Co-Authors: Yi Liang, Eckehard Steinbach, Bernd GirodAbstract:The quality of real-time Voice Communication over best-effort networks is mainly determined by the delay and loss characteristics observed along the network path. Excessive playout buffering at the receiver is prohibitive and significantly delayed packets have to be discarded and considered as late loss. We propose to improve the tradeoff among delay, late loss rate, and speech quality using multi-stream transmission of real-time Voice over the Internet, where multiple redundant descriptions of the Voice stream are sent over independent network paths. Scheduling the playout of the received Voice packets is based on a novel multi-stream adaptive playout scheduling technique that uses a Lagrangian cost function to trade delay versus loss. Experiments over the Internet suggest largely uncorrelated packet erasure and delay jitter characteristics for different network paths which leads to a noticeable path diversity gain. We observe significant reductions in mean end-to-end latency and loss rates as well as improved speech quality when compared to FEC protected single-path transmission at the same data rate. In addition to our Internet measurements, we analyze the performance of the proposed multi-path Voice Communication scheme using the ns network simulator for different network topologies, including shared network links.
Peter Vary - One of the best experts on this subject based on the ideXlab platform.
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frequency domain adaptive kalman filter for acoustic echo control in hands free telephones
Signal Processing, 2006Co-Authors: Gerald Enzner, Peter VaryAbstract:Acoustic echo canceler and postfilter for residual echo suppression are two essential building blocks of a hands-free Voice Communication system. Based on the Kalman filter theory, we derive a simple and advanced algorithm for the optimum joint statistical adaptation of both filter coefficients in time-varying and noisy acoustic environments. The Kalmar filter utilizes a stochastic state-space model of the acoustic echo path which is formulated entirely in the frequency-domain. The resulting adaptive algorithm is computationally efficient and inherently robust, i.e., the adaptation does not require additional regularization or control mechanisms.
Yi Liang - One of the best experts on this subject based on the ideXlab platform.
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adaptive playout scheduling and loss concealment for Voice Communication over ip networks
IEEE Transactions on Multimedia, 2003Co-Authors: Yi Liang, N Farber, Bernd GirodAbstract: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.
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real time Voice Communication over the internet using packet path diversity
ACM Multimedia, 2001Co-Authors: Yi Liang, Eckehard Steinbach, Bernd GirodAbstract:The quality of real-time Voice Communication over best-effort networks is mainly determined by the delay and loss characteristics observed along the network path. Excessive playout buffering at the receiver is prohibitive and significantly delayed packets have to be discarded and considered as late loss. We propose to improve the tradeoff among delay, late loss rate, and speech quality using multi-stream transmission of real-time Voice over the Internet, where multiple redundant descriptions of the Voice stream are sent over independent network paths. Scheduling the playout of the received Voice packets is based on a novel multi-stream adaptive playout scheduling technique that uses a Lagrangian cost function to trade delay versus loss. Experiments over the Internet suggest largely uncorrelated packet erasure and delay jitter characteristics for different network paths which leads to a noticeable path diversity gain. We observe significant reductions in mean end-to-end latency and loss rates as well as improved speech quality when compared to FEC protected single-path transmission at the same data rate. In addition to our Internet measurements, we analyze the performance of the proposed multi-path Voice Communication scheme using the ns network simulator for different network topologies, including shared network links.
Guoliang Xing - One of the best experts on this subject based on the ideXlab platform.
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asm adaptive Voice stream multicast over low power wireless networks
IEEE Transactions on Parallel and Distributed Systems, 2012Co-Authors: Guoliang XingAbstract:Low-power Wireless Networks (LWNs) have become increasingly available for mission-critical applications such as security surveillance and disaster response. In particular, emerging low-power wireless audio platforms provide an economical solution for ad hoc Voice Communication in emergency scenarios. In this paper, we develop a system called Adaptive Stream Multicast (ASM) for Voice Communication over multihop LWNs. ASM is composed of several novel components specially designed to deliver robust Voice quality for multiple sinks in dynamic environments: 1) an empirical model to automatically evaluate the Voice quality perceived at sinks based on current network condition; 2) a feedback-based Forward Error Correction (FEC) scheme where the source can adapt its coding redundancy ratio dynamically in response to the Voice quality variation at sinks; 3) a Tree-based Opportunistic Routing (TOR) protocol that fully exploits the broadcast opportunities on a tree based on novel forwarder selection and coordination rules; and 4) a distributed admission control algorithm that ensures the Voice quality guarantees when admitting new Voice streams. ASM has been implemented on a low-power hardware platform and extensively evaluated through experiments on a test bed of 18 nodes. The experiment results show that ASM can achieve satisfactory multicast Voice quality in dynamic environments while incurring low-Communication overhead.
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adaptive Voice stream multicast over low power wireless networks
Real-Time Systems Symposium, 2010Co-Authors: Guoliang XingAbstract:Low-power Wireless Networks (LWNs) have become increasingly available for mission-critical applications such as security surveillance and disaster response. In particular, emerging low-power wireless audio platforms provide an economical solution for ad hoc Voice Communication in emergency scenarios. In this paper, we develop a system called Adaptive Stream Multicast (ASM) for Voice Communication over multi-hop LWNs. ASM is composed of several novel components specially designed to deliver robust Voice quality for multiple sinks in dynamic environments: 1) an empirical model to automatically evaluate the Voice quality perceived at sinks based on current network condition, 2) a feedback-based Forward Error Correction scheme where the source can adapt its coding redundancy ratio dynamically in response to the Voice quality variation at sinks, 3) a Tree-based Opportunistic Routing (TOR) protocol that fully exploits the broadcast opportunities on a tree based on novel forwarder selection and coordination rules, and 4) a distributed admission control algorithm that ensures the Voice quality guarantees when admitting new Voice streams. ASM has been implemented on a low-power hardware platform and extensively evaluated through experiments on a testbed of 18 nodes.
Gerald Enzner - One of the best experts on this subject based on the ideXlab platform.
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frequency domain adaptive kalman filter for acoustic echo control in hands free telephones
Signal Processing, 2006Co-Authors: Gerald Enzner, Peter VaryAbstract:Acoustic echo canceler and postfilter for residual echo suppression are two essential building blocks of a hands-free Voice Communication system. Based on the Kalman filter theory, we derive a simple and advanced algorithm for the optimum joint statistical adaptation of both filter coefficients in time-varying and noisy acoustic environments. The Kalmar filter utilizes a stochastic state-space model of the acoustic echo path which is formulated entirely in the frequency-domain. The resulting adaptive algorithm is computationally efficient and inherently robust, i.e., the adaptation does not require additional regularization or control mechanisms.
