This paper proposes a new algorithm for predicting audio packet playout delay for voice conferencing applications that use silence suppression. The proposed algorithm uses a hidden Markov model (HMM) to predict the playout delay. Several existing algorithms are reviewed to show that the HMM technique is based on a combination of various desirable features of other algorithms. Voice over Internet protocol (VoIP) applications produce packets at a deterministic rate but various queuing delays are added to the packets by the network causing packet interarrival jitter. Playout delay prediction techniques schedule audio packets for playout and attempt to make a reasonable compromise between the number of lost packets, the one-way delay and the delay variation since these criteria cannot be optimized simultaneously. In particular, this paper will show that the proposed HMM technique makes a good compromise between the mean end-to-end delay, end-to-end delay standard deviation and average packet loss rate.

This paper proposes an acoustic echo cancellation structure for hands-free synthetic surround sound applications, such as multiple participant conferencing, and virtual reality applications. Voice over Internet protocol (VoIP) and other virtual reality applications can benefit from the addition of 3D spatial audio generated by more than two loudspeakers. When full-duplex audio is present in a system, however, acoustic echo cancellation is required to eliminate the feedback echo path. The acoustic echo cancellation structure proposed by this paper is based on the acoustic echo canceller per spatial region allocation scheme previously introduced by the authors for two channel synthetic stereo. This paper shows that the spatial region allocation scheme is extensible to any number of channels which makes it extremely versatile and flexible, especially for surround sound audio. Microsoft Direct X 7, a commonly used application programmer interface (API), was used in our simulations to generate the 3D spatial audio on a PC.

This paper proposes a novel acoustic echo cancellation structure intended for multiple participant, full-duplex, hands-free voice over Internet protocol (VoIP) conferencing. A synthetic stereo image is generated through the use of spatialization functions, which are used to assist the near-end user in distinguishing between far-end talkers. The proposed synthetic stereo structure, which uses a single echo canceler per spatial region, is compared to structures with a single echo canceler per channel and to stereophonic echo cancelers. The proposed structure does not suffer from correlation of the reference signals, which can cause system misconvergence in structures that use a single canceler per channel. Correlation of the reference signals is eliminated since monaural signals are transmitted at the far-end and static spatialization functions are used at the near-end to allocate each participant to a spatial region.

This paper proposes a synthetic stereo acoustic echo cancellation structure with microphone array beamforming for multiple participant voice over Internet protocol (VoIP) conferencing. The proposed acoustic front-end combines a synthetic stereo structure, acoustic echo canceler and microphone array beamformer. Synthetic stereo is used to add spatial information to signals from multiple participants that facilitates distinguishing between participants. Microphone array beamforming is then combined with the single canceler per spatial region echo cancellation structure to reduce noise and reverberation for near-end speech pickup and to cancel echo.

This paper proposes a theoretical model and an ITU-T H.323 compliant empirical measurement method for the acoustic round trip delay (ARTD) of a voice over Internet protocol (VoIP) conference. The empirical measurement method is able to compute the complete acoustic round trip delay that the user perceives. The acoustic round trip delay is experimentally measured by slicing the real-time signal path and inserting test hooks on the Carleton University VoIP platform. Current VoIP systems cannot measure the acoustic round trip delay. An acoustic domain verification test is used to check the ARTD empirical measurements for accuracy.