Abstract: An echo reduction method stores a received audio information stream. A sound detection flag is activated following detection of locally generated sound. Output based on the received audio information stream is muted in response to the activating the sound detection flag. Rendering status of the received audio information stream is saved, in response to the activating the sound detection flag, to reduce loss of audio information. At least a portion of the stored received audio information stream is rendered following inactivation of the sound detection flag.

Abstract: An echo reduction method stores a received audio information stream. A sound detection flag is activated following detection of locally generated sound. Output based on the received audio information stream is muted in response to the activating the sound detection flag. Rendering status of the received audio information stream is saved, in response to the activating the sound detection flag, to reduce loss of audio information. At least a portion of the stored received audio information stream is rendered following inactivation of the sound detection flag.

Abstract: A method for conversation like rendering of a stored audio information stream determines a first location in the stored audio information stream. The first location represents a point in time when the sound detection flag became active. The method next moves from the first location to a second location in the stored audio information stream. The second location is selected based upon a criterion to make playback of the stored audio information stream appear like actual conversation. Finally, the stored audio information stream is rendered starting with audio information stored at the second location.

Abstract: An echo reduction method stores a received audio information stream. A sound detection flag is activated following detection of locally generated sound. Output based on the received audio information stream is muted in response to the activating the sound detection flag. Rendering status of the received audio information stream is saved, in response to the activating the sound detection flag, to reduce loss of audio information. At least a portion of the stored received audio information stream is rendered following inactivation of the sound detection flag.

Abstract: An audio communications system has an acoustic echo cancellation (AEC) module. The AEC module receives a digital signal sent to a loudspeaker and a digital signal received from a microphone. The signal received from the microphone contains an echo of the signal played through the loudspeaker. The loudspeaker signal is processed by an audio generation module (AGM) that models substantially nonlinear distortions that can occur while producing the signal played through the loudspeaker. The AGM includes a modeling path comprised of one or more distortion modules. Each distortion module receives digital samples as input, modifies the samples to model a form of distortion, and outputs the modified samples. The output of the AGM is provided to an acoustic echo estimation (AEE) module that uses adaptive algorithms to compensate for substantially linear changes in the echo characteristics of the environment in which the loudspeaker and microphone are located.

Abstract: A communications system and method performs centralized signal processing on received audio signals. A plurality of terminals are coupled to a processing switch via links. The terminals can be, for example, dedicated speakerphones, desktop handsets, or personal computers with audio capabilities. The links can be wired or wireless, can carry analog or digital signals, and can be shared with other users in a network. The switch receives the audio data from the terminals, processes the data according to desired acoustical procedures, creates one or more output mixes, and provides the output mixes to the appropriate terminals. The operation of the processing switch is controlled by a communications support module (CSM) which can receive, process, and send data to/from multiple terminals simultaneously. The CSM receives audio signals from the terminals.

Abstract: Disclosed is a method and system for detecting processor congestion during decompression of a stream of video and audio data. The system and method includes a processor receiving and decoding a first frame of audio data in accordance with an audio decode software algorithm. The processor generates a first audio time stamp ATS1 indicating the time at which the processor finishes decoding the first frame of audio data. Subsequently, the processor receives and decodes a second frame of audio data in accordance with the same audio decode software algorithm and generates a second audio time stamp ATS2 indicating the time at which the processor finishes decoding the second audio data frame. The first audio time stamp ATS1 is added to a predetermined amount of time T, the result of which is compared with ATS2. T, in one embodiment, is the time it takes a speaker to generate audio corresponding to a decoded frame of audio data.

Abstract: Disclosed is a method and system for reducing audio artifacts and/or avoiding invalid reference memory in a compressed video decoder due to processor congestion. The system and method includes decoding compressed frames of video data, and decoding compressed frames of audio data. The system and method determines whether audio data can be generated without audio artifacts. If it is determined that audio can be generated without audio artifacts, then images corresponding to the decoded frames of video data, respectively, are subsequently displayed. If however it is determined that the audio can not be generated without artifacts due to processor congestion, images are displayed and redisplayed while processor power is shifted to decoding frames of audio data.

Abstract: A method and associated apparatus for controlling an acoustic canceler (“AEC”) are disclosed. Prior to passing audio signals to the AEC, a distortion detector is used to determine if the signals are distorted. If so, the AEC does not adapt is filter coefficients to the distorted signals. This technique improves the AEC's ability to adapt its filter coefficients to subsequent undistorted signals. For example, near-end or far-end audio signals above a predetermined threshold value are detected by a distortion detector which disables adaptive filter control logic so that distorted signals do not result in generation of erroneous filter coefficients.

Abstract: A video teleconferencing system and method transfers video teleconferencing signals from a sender to a receiver. The sender determines decision information based on internal or external factors. The sender may or may not generate a video teleconferencing signal depending on the content of the decision information. If generated, the video teleconferencing signal is encoded at the sender and sent to the receiver. Each sender includes at least one memory module for storing the decoded signal, each memory module is one group. The sender updates its memory module with a copy of each sent video teleconferencing signal. The receiver decodes the signal and presents the signal to the user of the receiver. The receiver stores a copy of the signal in a memory module identified with each specific sender.