Abstract: A system configured to improve double-talk detection. The system inputs microphone signals into an adaptive filter and determines whether double talk is present based on how the adaptive filter adapts to the microphone signals. For example, when an audible sound is detected, the adaptive filter updates the filter coefficients that correspond to a time difference of arrival of the audible sound. Thus, the device may detect single-talk conditions (e.g., a single peak in the filter coefficients) or double-talk conditions (e.g., two peaks in the filter coefficients). In addition, the device may track a location of the local speech or remote speech over time based on the time difference of arrival.

Abstract: Techniques for improving acoustic echo cancellation are described. Energy levels of audio data received from a microphone and representing near-end audio and reference audio data representing far-end audio are determined. If near-end audio is detected but far-end audio is not detected, a controller turns of or bypasses an acoustic echo cancellation system until far-end audio is again detected, thereby decreasing or eliminating distortion of the near-end audio by the acoustic echo cancellation system.

Abstract: A system configured to improve audio processing by adaptively selecting target signals based on current system conditions. For example, a device may select a target signal based on a highest signal quality metric when only the local speech is present (e.g., during near-end single-talk conditions), as this maximizes an amount of energy included in the output audio signal. In contrast, the device may select the target signal based on a lowest signal quality metric when only the remote speech is present (e.g., during far-end single-talk conditions), as this minimizes an amount of energy included in the output audio signal. In addition, the device may track positions of the local speech and the remote speech over time, enabling the device to accurately select the target signal when both local speech and remote speech is present (e.g., during double-talk conditions).

Abstract: A method and device are described for implementing instant communication. Information indicating that a first user is logged into an instant communication interface is obtained. On-line users are displayed to the first user in response to the login of the first user. A first user selects an object to follow from the on-line users and, information about the selected object to follow is received in response to the first user selecting the object to follow from the on-line users. The first user may be set into a first instant communication room in which the selected object to follow resides, for performing instant communication. The device includes: an obtaining module, a displaying module and a processing module. In this manner communication between strangers is facilitated and the efficiency of making friends is enhanced.

Abstract: A system configured to improve double-talk detection. The system detects when double-talk is present in a voice conversation using two or more speaker models. The system extracts feature data from microphone audio data and compares the feature data to each speaker model. For example, the system may generate a first distance score indicating a likelihood that the feature data corresponds to a far-end speaker model and a second distance score indicating a likelihood that the feature data corresponds to a universal speaker model. The system may determine current system conditions based on the distance scores and may change settings to improve speech quality during the voice conversation. For example, during far-end single-talk the system may aggressively reduce an echo signal, whereas during near-end single-talk and double-talk the system may apply minimal echo cancellation to improve a quality of the local speech.

Abstract: Techniques for improving acoustic echo cancellation to attenuate an echo signal generated by a wireless loudspeaker are described. A relative position of the wireless loudspeaker is determined when sound from the wireless loudspeaker is determined to be the dominant sound (e.g., during output of a response to a user command or query or during speech by a far-side party during a two-way audio communication). A beam corresponding to the relative position is thereby selected for echo cancellation.

Abstract: A device that includes a first beam selection component that selects a portion of the multiple beams prior to performing adaptive noise cancellation, in addition to a second beam selection component that selects a beam after adaptive noise cancellation is performed. The device only performs adaptive noise cancellation on the portion of the multiple beams that are selected, reducing a complexity of performing adaptive noise cancellation. In addition, as the first beam selection component selects from the multiple beams before adaptive noise cancellation, the quality of the signals are not reduced and/or desired signals are not suppressed, resulting in the first beam selection component increasing a likelihood that the second beam selection component selects the desired beam. The first beam selection component may use a different algorithm and/or selection criteria than the second beam selection component in order to select beams associated with desired signals such as speech.

Abstract: A method and device are described for implementing instant communication. Information indicating that a first user is logged into an instant communication interface is obtained. On-line users are displayed to the first user in response to the login of the first user. A first user selects an object to follow from the on-line users and, information about the selected object to follow is received in response to the first user selecting the object to follow from the on-line users. The first user may be set into a first instant communication room in which the selected object to follow resides, for performing instant communication. The device includes: an obtaining module, a displaying module and a processing module. In this manner communication between strangers is facilitated and the efficiency of making friends is enhanced.

Abstract: A method and device are described for implementing instant communication. Information indicating that a first user is logged into an instant communication interface is obtained. On-line users are displayed to the first user in response to the login of the first user. A first user selects an object to follow from the on-line users and, information about the selected object to follow is received in response to the first user selecting the object to follow from the on-line users. The first user may be set into a first instant communication room in which the selected object to follow resides, for performing instant communication. The device includes: an obtaining module, a displaying module and a processing module. In this manner communication between strangers is facilitated and the efficiency of making friends is enhanced.

Abstract: Techniques are described herein that use sensors (e.g., microphones) for noise reduction in a mobile communication device. For example, one technique enables a first sensor that is initially configured to be a speech sensor to be used as a noise reference sensor. This technique also enables a second sensor that is initially configured to be a noise reference sensor to be used as a speech sensor. Another technique enables a primary sensor and/or a secondary sensor in a handset of a mobile communication device to be used as a speech sensor while a sensor in a headset of the mobile communication device is used as a noise reference sensor, or vice versa. In yet another technique, a secondary sensor in a mobile communication device is configured to be a directional sensor.

Abstract: There are provided a method, apparatus, system and computer readable storage medium of adding an instant message contact. The method includes: receiving a joining-in request for a group from a client where a user locates, the joining-in request comprising a group identifier; acquiring audio and/or video information of other user in the group, based on the group identifier; and transmitting the audio and/or video information of the other user to the client where the user locates, for adding by the client where the user locates the instant message contact based on the audio and/or video information of the other user. Thereby, the mutual trust between unknown users is improved, and the success ratio of adding an unknown user as a contact and the ratio of turning into a contact from an unknown user are increased.

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this tact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract: Systems and methods are provided for cross-terminal pushing of applications. For example, installation information of an application corresponding to an application-push event in a browser opened in a first terminal is acquired upon detection of the application-push event being triggered; the first terminal sends a push instruction to a server, the push instruction carrying the installation information of the application; the server sends the installation information of the application to one or more second terminals related to the first terminal; the second terminals send the installation information of the application; and the second terminals install the application based on at least information associated with the installation information of the application.

Abstract: Techniques are described herein that use sensors (e.g., microphones) for noise reduction in a mobile communication device. For example, one technique enables a first sensor that is initially configured to be a speech sensor to be used as a noise reference sensor. This technique also enables a second sensor that is initially configured to be a noise reference sensor to be used as a speech sensor. Another technique enables a primary sensor and/or a secondary sensor in a handset of a mobile communication device to be used as a speech sensor while a sensor in a headset of the mobile communication device is used as a noise reference sensor, or vice versa. In yet another technique, a secondary sensor in a mobile communication device is configured to be a directional sensor.

Abstract: Techniques are described herein that use sensors (e.g., microphones) for noise reduction in a mobile communication device. For example, one technique enables a first sensor that is initially configured to be a speech sensor to be used as a noise reference sensor. This technique also enables a second sensor that is initially configured to be a noise reference sensor to be used as a speech sensor. Another technique enables a primary sensor and/or a secondary sensor in a handset of a mobile communication device to be used as a speech sensor while a sensor in a headset of the mobile communication device is used as a noise reference sensor, or vice versa. In yet another technique, a secondary sensor in a mobile communication device is configured to be a directional sensor.

Abstract: Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage.

Abstract: Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract: Techniques are described herein that suppress noise using multiple sensors (e.g., microphones) of a communication device. Noise modeling (e.g., estimation of noise basis vectors and noise weighting vectors) is performed with respect to a noise signal during operation of a communication device to provide a noise model. The noise model includes noise basis vectors and noise coefficients that represent noise provided by audio sources other than a user of the communication device. Speech modeling (e.g., estimation of speech basis vectors and speech weighting) is performed to provide a speech model. The speech model includes speech basis vectors and speech coefficients that represent speech of the user. A noisy speech signal is processed using the noise basis vectors, the noise coefficients, the speech basis vectors, and the speech coefficients to provide a clean speech signal.

Abstract: A method and device are described for implementing instant communication. Information indicating that a first user is logged into an instant communication interface is obtained. On-line users are displayed to the first user in response to the login of the first user. A first user selects an object to follow from the on-line users and, information about the selected object to follow is received in response to the first user selecting the object to follow from the on-line users. The first user may be set into a first instant communication room in which the selected object to follow resides, for performing instant communication. The device includes: an obtaining module, a displaying module and a processing module. In this manner communication between strangers is facilitated and the efficiency of making friends is enhanced.