摘要:
An apparatus for mixing audio signals in a voice-over-IP teleconferencing environment comprises a preprocessor, a mixing controller, and a mixing processor. The preprocessor is divided into a media parameter estimator and a media preprocessor. The media parameter estimator estimates signal parameters such as signal-to-noise ratios, energy levels, and voice activity (i.e., the presence or absence of voice in the signal), which are used to control how different channels are mixed. The media preprocessor employs signal processing algorithms such as silence suppression, automatic gain control, and noise reduction, so that the quality of the incoming voice streams is optimized. Based on a function of the estimated signal parameters, the mixing controller specifies a particular mixing strategy and the mixing processor mixes the preprocessed voice streams according the strategy provided by the controller.
摘要:
A differential microphone array (DMA) is provided that includes a number (M) of microphone sensors for converting a sound to a number of electrical signals and a processor that is configured to apply linearly-constrained minimum variance filters on the electrical signals over a time window to calculate frequency responses of the electrical signals over a plurality of subbands and sum the frequency responses of the electrical signals for each subband to calculate an estimated frequency spectrum of the sound.
摘要:
A differential microphone array includes a number (M) of microphone sensors for converting sound to a number of electrical signals, and a processor, operably coupled to the microphone sensors, to specify a target differential order (N) for the differential microphone array, and wherein M>N+1, specify a steering matrix D comprising N+1 steering vectors, calculate a respective one of a plurality of linearly specify a steering matrix D comprising N+1 steering vectors-constrained minimum variance filters based on the steering matrix, apply the respective one of the plurality of linearly-constrained minimum variance filters to a respective one of the electrical signals to calculate a respective frequency response of the electrical signals, wherein the respective frequency response comprises a plurality of components associated with a plurality of subbands, and sum the frequency responses of the electrical signals with respect to each subband to calculate an estimated frequency spectrum of the sound.
摘要:
A method and apparatus for performing active speaker selection in teleconferencing applications illustratively comprises a microphone array module, a speaker recognition system, a user interface, and a speech signal selection module. The microphone array module separates the speech signal from each active speaker from those of other active speakers, providing a plurality of individual speaker's speech signals. The speaker recognition system identifies each currently active speaker using conventional speaker recognition/identification techniques. These identities are then transmitted to a remote teleconferencing location for display to remote participants via a user interface. The remote participants may then select one of the identified speakers, and the speech signal selection module then selects for transmission the speech signal associated with the selected identified speaker, thereby enabling the participants at the remote location to listen to the selected speaker and neglect the speech from other active speakers.
摘要:
An apparatus for mixing audio signals in a voice-over-IP teleconferencing environment comprises a preprocessor, a mixing controller, and a mixing processor. The preprocessor is divided into a media parameter estimator and a media preprocessor. The media parameter estimator estimates signal parameters such as signal-to-noise ratios, energy levels, and voice activity (i.e., the presence or absence of voice in the signal), which are used to control how different channels are mixed. The media preprocessor employs signal processing algorithms such as silence suppression, automatic gain control, and noise reduction, so that the quality of the incoming voice streams is optimized. Based on a function of the estimated signal parameters, the mixing controller specifies a particular mixing strategy and the mixing processor mixes the preprocessed voice streams according the strategy provided by the controller.
摘要:
A signal detection technique for multiple-input multiple-output (MIMO) communications systems embodied in a method and apparatus for detecting a plurality of transmitted signals with use of a plurality of receiving antennas. An iterative procedure decodes one of a plurality of transmitted signals at each iteration using an intermediate matrix at each iteration to determine the transmitted signal to be decoded. The intermediate matrix for each successive iteration is advantageously computed in a recursive manner with use of a Schur complement operation performed based on the inverse of a modified version of the intermediate matrix used in the previous iteration.
摘要:
A differential microphone array (DMA) is provided that includes a number (M) of microphone sensors for converting a sound to a number of electrical signals and a processor that is configured to apply linearly-constrained minimum variance filters on the electrical signals over a time window to calculate frequency responses of the electrical signals over a plurality of subbands and sum the frequency responses of the electrical signals for each subband to calculate an estimated frequency spectrum of the sound.
摘要:
A method and apparatus for performing active speaker selection in teleconferencing applications illustratively comprises a microphone array module, a speaker recognition system, a user interface, and a speech signal selection module. The microphone array module separates the speech signal from each active speaker from those of other active speakers, providing a plurality of individual speaker's speech signals. The speaker recognition system identifies each currently active speaker using conventional speaker recognition/identification techniques. These identities are then transmitted to a remote teleconferencing location for display to remote participants via a user interface. The remote participants may then select one of the identified speakers, and the speech signal selection module then selects for transmission the speech signal associated with the selected identified speaker, thereby enabling the participants at the remote location to listen to the selected speaker and neglect the speech from other active speakers.