Provided are a videoconferencing server capable of providing multiscreen videoconferencing by using multiple videoconferencing terminals, and a method therefor. The videoconferencing server of the present disclosure can be implemented in such a manner that multiple conventional videoconferencing terminals (physical terminals) having one or two displays are logically grouped to operate as a logical terminal which operates as if it were one videoconferencing point. Through distribution of videos provided to multiple physical terminals constituting a logical terminal, the videoconferencing server can perform processing as if the logical terminal supported a multiscreen, and the videoconferencing server provides an echo cancellation function in the logical terminal.

First and second computer systems and respective first and second microphones thereof receive respective portions of a same audio input signal. Audio buffers received respectively from the first and second computer system include data encoded from respective microphone inputs of the first and the second computer systems. The received audio buffers are synchronized and corrected for gain differences between the received audio buffers to produce corrected audio buffers. The corrected audio buffers are mixed into an output buffer. The synchronization reduces echo when the output buffer is played at a remote peer computer system.

The present disclosure provides a smart audio device, including a front chip provided therein with a plurality of voice algorithm modules; and a main control chip signally connected with the front chip and configured to call the voice algorithm modules in the front chip according to a user request in a multi-thread mode. The smart audio device is low in cost and power consumption, has long service life, and can improve user experience. The present disclosure further provides a calling method for audio device, an electronic device and a computer readable medium.

A system and method for reducing audio feedback, and in particular howling, during a communication session involving two or more computing devices. The system determines that a first device is likely to join a meeting in which another, second device is already joined. In response, the system can cause the second device to broadcast a signal that, if detected by the first device, will indicate that the two devices are in auditory range of one another. As a result, the system can preemptively mute the audio components of the first device to prevent an audio stream of the first device from interfering with the audio already streaming via the second device.

Echo cancellation for a two-way audio communication includes receiving, at an AEC system from microphone(s), an audio signal based on, at least in part, near-end signals and reproduced far-end signals. Loudspeaker(s) reproduced the far-end signals. The AEC system is operated, at least in part, with filter(s) so as to update estimates of coefficients of an acoustic channel from the loudspeaker(s) to the microphone(s). Control parameter(s) affecting an operation of the AEC system that is/are configurable and is/are set to value(s), from a range of values, is/are determined, based on estimating an accuracy of the estimates of the coefficients of the acoustic channel and a characteristic of the near-end signals. The AEC system controls the filter(s) with different values of the control parameter(s) at different times.

Method, apparatus, and program code embodied in computer-readable media, for providing enhanced echo suppression in a conferencing system having at least one microphone and at least one speaker. At least one microphone input signal is received, and at least one speaker input signal is provided. At least one processor has at least one primary echo-suppressor and at least one secondary echo-suppressor. The at least one primary echo-suppressor receives (i) the microphone input signal(s) and (ii) the speaker input signal(s). The at least one primary echo-suppressor provides at least one echo-suppressed microphone signal. The at least one secondary echo-suppressor receives the at least one echo-suppressed microphone signal and provides an output signal. The at least one processor provides the at least one echo-suppressed microphone signal to the at least one secondary echo-suppressor without providing the at least one speaker input signal directly to the at least one secondary echo-suppressor.

Systems, methods, and devices are disclosed for detecting an active speaker in a two-way conference. Real time audio in one or more sub band domains are analyzed according to an echo cancellor model. Based on the analyzed real time audio, one or more audio metrics are determined from output from an acoustic echo cancellation linear filter. The one or more audio metrics are weighted based on a priority, and a speaker status is determined based on the weighted one or more audio metrics being analyzed according to an active speaker detection model. For an active speaker status, one or more residual echo or noise is removed from the real time audio based on the one or more audio metrics.

Systems, methods, and computer program products for provisioning a temporary disposable number are described. A user can be provided with a pool of available temporary disposable numbers that have a limited shelf life. The user can select one of the available temporary disposable numbers while submitting a permanent phone number associated with a communications device (e.g., mobile phone, home phone, business phone, etc.). Prior to activating the selected temporary disposable number, the temporary disposable number is linked to the permanent phone number. After activation, when an incoming call to the temporary disposable number is received, the permanent phone number is identified to be associated with the temporary disposable number being called. The incoming call is then forwarded to the communications device on which the permanent phone number is established.

An endpoint receives audio from a remote endpoint. A first signal corresponding to the audio is received at an adaptive filter, and a filtered signal is generated. First audio is emitted at a loudspeaker based on the first signal. A microphone collects second audio which is based on the first audio. The microphone signal emits a signal based on the second audio. The filtered signal is subtracted from microphone signal to generate an adapted signal, the adapted signal having an energy level. The adapted signal is then transmitted to a double-talk detector, the double-talk detector configured to allow transmission of the adapted signal to the remote endpoint when the energy level of the adapted signal exceeds an energy threshold. The degree of cross-correlation between the first signal and the adapted signal is determined (iteratively). If the cross-correlation exceeds a cross-correlation threshold, the energy threshold of the double-talk detector is raised.

A system includes participant nodes that unicast a voice stream to a master node and receive a multicast voice stream from the master node. The multicast voice stream may be a mixed voice stream mixing the voice streams of the participants. In this case, each participant performs echo cancellation on the multicast voice stream. Alternatively, the multicast voice stream may be a multiplexed voice stream multiplexing the voice streams of all participants. In this case, each participant de-multiplexes the multicast voice stream and generates a mixed voice stream that does not include that participant. Alternatively, the multicast voice stream received by each participant may be a multiplexed digital voice stream multiplexing multiple mixed streams including one mixed stream that does not include the voice stream of that participant. In this case, each participant de-multiplexes the multicast voice stream to obtain the mixed stream that does not include that participant.