Mitel Networks Corporation

Zarlink Semiconductor Limited

Mitel Corporation

Mitel Knowledge Corporation

Franck Beaucoup

Michael Tetelbaum

Robert Geoffrey Wood

In a system having an adaptive filter block for receiving a reference signal and adapting to an input signal, and a block for detecting corruption of the input signal by an interference signal and in response limiting adaptation of the adaptive filter block, the improvement comprising delaying and applying the input signal to the adaptive filter block while applying the input signal to the block for detecting corruption without delay.

Delayed adaptation structure for improved double-talk immunity in echo cancellation devices

A method and apparatus is set forth for accelerating the total acoustic echo cancellation convergence time in a microphone array full-duplex conferencing system with beamformer. The invention is based on sequentially switching the beamformer to untrained sectors during periods of far-end speech activity and performing real-time cancellation of far-end echo signals from the near-end signals to consecutively adapt and store filter coefficients for the echo canceller corresponding to each sector.

Method of accelerating the training of an acoustic echo canceller in a full-duplex beamforming-based audio conferencing system

A howling control structure for a full duplex communication system. The structure is implemented as part of an acoustic echo canceller having a conventional transversal adaptive filter. A second transversal adaptive filter, shorter than the conventional filter, that adapts even in the absence of speech in its reference signal, is provided. The short filter adapts quickly and provides enough echo cancellation to prevent howling from occurring, even if the echo path is changed significantly during silence periods.

Anti-howling structure

A method and system for rapid adaptive coupling equalization in beamforming-based communication systems, particularly sector-based beamforming systems, provides smooth transitions for AEC when the look direction of the communication system changes and when the acoustic environment varies with time. The coefficients of inactive beamformers are modified in real-time, using adaptive beamforming techniques based on the real-time loudspeaker-coupling signal, in order to force the outputs of inactive beamformers to have the same response to the loudspeaker coupling signal as the active beamformer does.

Adaptive coupling equalization in beamforming-based communication systems

A method is set forth for reducing the total acoustic echo cancellation convergence time for all look directions in a microphone array based full-duplex system. The method is based on capturing the loudspeaker signal due to the first far-end speech bursts when the conferencing system is first used, as well as the corresponding loudspeaker feedback signals in the individual microphones. The captured signals are then used for consecutive adaptation of the acoustic echo canceller on all echo paths corresponding to all look directions of the beamformer, thereby training the AEC. This training process can be executed concurrently with normal phone operation, for example, as a background process that utilizes available processing cycles.

Method to reduce training time of an acoustic echo canceller in a full-duplex beamforming-based audio conferencing system

A method to reduce acoustic coupling in an audio conferencing system having a loudspeaker and a plurality of microphones, comprising detecting the presence of one of either a source of near-end signal or a source of far-end signal source relative to the audio conferencing system, and in the event of detecting a source of near-end signal then processing the near-end signal with high directivity towards the near-end source, and in the event of detecting a source of far-end signal or double-talk then processing the far-end signal with high coupling characteristics between the loudspeaker and microphones.

Method to reduce acoustic coupling in audio conferencing systems

A method of designing a beamformer, and a beamformer made in accordance with this method, characterized by a uniform speakerphone response condition, resulting in optimal beamforming directivity under a uniform coupling constraint. According to the present invention, a finite number of individual beamformers are constrained to have the same response to a loudspeaker signal (as well as the same gain in their respective look directions) without specifying the exact value of their response to this signal. This results in beamformer weights that are optimal in the minimum variance sense and satisfy the uniform coupling constraint. The minimum variance condition combines all beamformer weights at once, and the uniform coupling constraint is expressed as a finite number of linear constraints on the weights of the individual beamformers, without specifying an arbitrary, a priori value for the actual value of the uniform response.

Method for optimal microphone array design under uniform acoustic coupling constraints

A method of improving convergence of an echo canceller in a full duplex speakerphone, wherein the echo canceller includes LEC (Line Echo Canceller) and AEC (Acoustic Echo Canceller) portions, comprising the steps of capturing coefficients during operation, storing the coefficients, and utilizing the stored coefficients as default values during start-up of a subsequent call. According to the method of the present invention, the amount of echo cancellation necessary to provoke a save of the coefficients decreases with time when it is not achieved by the system despite the presence of a reference signal (i.e. speakerphone signal).

Method of capturing constant echo path information in a full duplex speakerphone

A method of locating a talker in a reverberant environment comprises receiving multiple audio signals from a microphone array that include direct path audio signal and reverberation signal components. The direct path audio signal components of the multiple audio signals are detected and are used to weight the multiple audio signals. A position estimate based on the weighted audio signals is then calculated. Periods of speech activity are detected and a final position estimate is generated during the periods of speech activity.

Robust talker localization in reverberant environment

Apparatus for controlling noise characteristic estimation in a conferencing system, comprising a noise characteristic estimator for estimating a noise characteristic of a signal of interest transmitted in a first direction through the conferencing system, and a first voice activity detector for detecting audio signal activity in a signal transmitted through the conferencing system in a direction opposite to the signal of interest and in response disabling the noise characteristic estimator.

Method and apparatus of controlling noise level calculations in a conferencing system

A parallel GSC structure is provided by which an adaptive process is performed by a plurality of beamformers in parallel in such a way that they present a common response to the equalization signal that varies over time in an optimal manner with respect to the statistics of the steering vectors.

Parallel GCS structure for adaptive beamforming under equalization constraints

A method is set forth of controlling an acoustic echo canceller at the output of a beamformer in an audio conferencing device. Information is saved to, and retrieved from, memory that characterizes each of a finite number of look directions, or regions of focus, covering the entire spatial span of the conferencing device. Each time a change occurs from a first look direction to a second look direction, information relating to the workspace captured by the acoustic echo canceller is saved for the first look direction, and previously saved information for the second look direction is retrieved from memory. The acoustic echo cancellation then takes place for the second look direction with the retrieved information.

Method of acoustic echo cancellation in full-duplex hands free audio conferencing with spatial directivity

A Comfort Noise Generation (CNG) system is provided for use in open systems where there is no predefined protocol for transmission of Silence Insertion Descriptor (SID) information from transmitter to receiver. The receiver enters an underrun condition in response to periods of silence, and in response generates comfort noise. According to the present invention, the computation of the level and spectral characteristics of the background of the speech signal is done within the receiver, thereby overcoming the lack of a protocol to transmit the SID information during silence periods. These characteristics are computed as a gain parameter and a set of Linear Prediction Coding (LPC) parameters which are applied to a filter which filters flat-spectrum noise in order to generate noise that sounds like the background noise of the speech signal.

Comfort noise generation for open discontinuous transmission systems

A method is provided for generating comfort noise in a packetized voice communication system having a transmitter and a receiver. The receiver is provided with a buffer for storing voice packets. The buffer is chosen to be of a predetermine size such that, upon halting the transmitter as a result of silence detection, the buffer is filled with actual silence samples from the transmitter. A comparator compares an output TDM sample pointer with a start of silence pointer of the buffer. In the event that the pointers are the same, silence is flagged and a random number generator loads numbers into the TDM sample pointer for outputting a random sequence of the silence packets to the telephone receiver.

Method to generate telephone comfort noise during silence in a packetized voice communication system

A method and apparatus for detecting silence in voice packets. A packet energy calculator calculates a smoothed energy value for each packet of voice data to be transmitted. A noise level detector adaptively calculates noise values during periods of said silence. A silent packet detector compares the energy value to the noise value and if it is less than the noise value and less than a predetermined silence ceiling value then silence is indicated. Also, if the energy value is less than a predetermined silence noise value then silence is also indicated.

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