Methods and systems are provided for participant-individualized audio volume control and host-customized audio volume control of streaming audio for a plurality of participants who are each receiving the streaming audio from a host within a videoconferencing platform, and who are also simultaneously engaged in remote audio communications with each other within the same videoconferencing platform.

An operation method of voice communication system with echo-cancellation, comprising following steps of: capturing audio-signal by a first sound capturing portion of a first transceiver device to output a first residual-echo signal to an audio-signal-processing portion of the first transceiver device for echo cancellation; outputting a first echo-cancelled signal by the audio-signal-processing portion to a first communication portion and to a second communication portion of a second transceiver device through a connection and to a second sound generating portion for generating audio-signal; capturing audio-signal by a second sound capturing portion to output a second residual-echo signal to the second communication portion and to the first communication portion of the first transceiver device through the connection and to the audio-signal-processing portion for echo cancellation; outputting a second echo-cancelled signal to a first sound generating portion of the first transceiver device for generating audio-signal.

Techniques to dynamically engage echo cancellation are described. In one embodiment, an apparatus may comprise a streaming component operative to establish a audio connection between the first client device and a second client device via the network interface controller; and receive a far-end audio stream at the first client device from the second client device via the audio connection; an audio capture component operative to capture a near-end audio stream at the first client device; and an echo processing component operative to compare the near-end audio stream and the far-end audio stream to determine whether a far-end echo is present in the near-end audio stream; and use an echo-cancellation module at the first client device where the far-end echo is present in the near-end audio stream. Other embodiments are described and claimed.

An apparatus and/or method discloses a video conference with enhanced audio quality using high-fidelity audio sharing (“HAS”). In one embodiment, a network connection between a first user equipment (“UE”) and a second UE is established via a communication network for providing an interactive real-time meeting. After sending a first calibration audio signal from the first UE to the second UE, a second calibration audio signal is retuned from the second UE to the first UE according to the first calibration audio signal. Upon identifying a far end audio (“FEA”) delay based on the first calibration audio signal and the second calibration audio signal, a first mixed audio data containing the first shared audio data and first FEA data is fetched from an audio buffer. The first FEA data is subsequently removed or extracted from the mixed audio data in response to the FEA delay.

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.

A method, a system, and a server for reducing noise in a workspace are disclosed. The workspace may have a plurality of terminals connected to a server via a communication network. The method may include detecting a noise level, which is above a predetermined threshold value, in or adjacent to at least one of the terminals. The noise level in or adjacent to a terminal may be detected by monitoring noise emitted from or in the vicinity of that terminal by a microphone associated with that terminal. The method may also include identifying the terminal at which the noise level exceeds the predetermined threshold. Further, the method may include initiating a measure for reducing the noise level.

Provided are an audio processing device, an audio processing method, an information processing device, and a computer program that perform echo cancellation corresponding to double talk. The audio processing device includes an estimation unit that estimates a filter representing a transmission characteristic from a speaker where a reference signal is output to a microphone in which the reference signal sneaks, an adjustment unit that adjusts a step size on the basis of a filter update coefficient estimated by the estimation unit, and an update unit that updates the filter according to the update coefficient and the step size. The adjustment unit adjusts the step size on the basis of a ratio of power of the filter update coefficient to maximum power of the filter.

A double-talk state detection method includes: calculating an energy ratio between a first energy of an error signal in each sub-band of M sub-bands and a second energy of a filtered signal in the same sub-band as the error signal, thereby obtaining M energy ratios, where the error signal is a difference between an input signal collected by a microphone and the filtered signal, the filtered signal is a signal obtained after performing filtering process on a reference signal, and M is a positive integer; performing a first smoothing processing on the M energy ratios to obtain M first energy smoothing ratios, and performing a second smoothing processing on the M first energy smoothing ratios to obtain M second energy smoothing ratios; performing double-talk state detection based on the M first energy smoothing ratios and the M second energy smoothing ratios to determine a state of the input signal.

A position detection method may include obtaining voice signals during a voice call by at least two voice collecting devices; obtaining position energy information of the voice signals; and identifying a position of the terminal device relative to a user during the voice call, from predefined positions based on the position energy information.

An apparatus and/or method discloses a video conference with enhanced audio quality using high-fidelity audio sharing (“HAS”). In one embodiment, a network connection between a first user equipment (“UE”) and a second UE is established via a communication network for providing an interactive real-time meeting. After sending a first calibration audio signal from the first UE to the second UE, a second calibration audio signal is retuned from the second UE to the first UE according to the first calibration audio signal. Upon identifying a far end audio (“FEA”) delay based on the first calibration audio signal and the second calibration audio signal, a first mixed audio data containing the first shared audio data and first FEA data is fetched from an audio buffer. The first FEA data is subsequently removed or extracted from the mixed audio data in response to the FEA delay.