Abstract: Network communication speech handling systems are provided herein. In one example, a method of processing audio signals by a network communications handling node is provided. The method includes processing an audio signal to determine a pitch cycle property associated with the audio signal, determining transfer times for encoded segments of the audio signal based at least in part on the pitch cycle property, and transferring packets comprising one or more encoded segments for delivery to a target node in accordance with the transfer time.

Abstract: Techniques for real-time audio communication including supplying an initial audio signal to an audio signal processor configured to process the initial audio signal and output a processed audio signal to an audio output means, obtaining a loopback audio signal corresponding to the processed audio signal, generating a plurality of audio features for the initial audio signal and the loopback audio signal, generating a similarity measure estimating a similarity of the initial audio signal to the first loopback audio signal based on at least the plurality of audio features, adjusting an adaptation rate for an audio signal processing operation based on at least the similarity measure, and controlling, based on at least the adjusted adaptation rate, an echo cancellation process for cancelling an estimated echo from a near-end audio signal received via an audio input means in proximity to the audio output means.

Abstract: Network communication speech handling systems are provided herein. In one example, a method of processing audio signals by a network communications handling node is provided. The method includes processing an audio signal to determine a pitch cycle property associated with the audio signal, determining transfer times for encoded segments of the audio signal based at least in part on the pitch cycle property, and transferring packets comprising one or more encoded segments for delivery to a target node in accordance with the transfer time.

Abstract: Network communication speech handling systems are provided herein. In one example, a method of processing audio signals by a network communications handling node is provided. The method includes receiving an incoming excitation signal transferred by a sending endpoint, the incoming excitation signal spanning a first bandwidth portion of audio captured by the sending endpoint. The method also includes identifying a supplemental excitation signal spanning a second bandwidth portion that is generated at least in part based on parameters that accompany the incoming excitation signal, determining a normalized version of the supplemental excitation signal based at least on energy properties of the incoming excitation signal, and merging the incoming excitation signal and the normalized version of the supplemental excitation signal by at least synthesizing an output speech signal having a resultant bandwidth spanning the first bandwidth portion and the second bandwidth portion.

Abstract: Disclosed is an audio signal processing device comprising an input for receiving a noisy audio signal, a variable gain component and a noise suppression component. The noisy audio signal has a desired audio component and a noise component. The variable gain component and the noise suppression component are respectively configured to apply a gain and a noise suppression procedure to the audio signal, thereby generating a gain adjusted noise reduced audio signal. The aggressiveness of the noise suppression procedure is rapidly changed responsive to a change in the applied gain. That change is a change from a current value by an amount substantially matching the change in applied gain to a new value. The aggressiveness is then gradually returned to the current value.

Abstract: A method of echo cancellation in an acoustic system, comprising a first user device and at least one further user device, the method comprising, at the first user device: detecting an audio signal received from the at least one further user device over a communications network; supplying the audio signal when received to an audio signal processing module of the first user device, wherein the audio signal processing module processes the audio signal and outputs a processed audio signal to audio output means of the first user device; and controlling an echo cancellation process for cancelling echo from an audio signal received via audio input means of the first user device based on a loopback signal provided by the processed audio signal and selectively based on the received audio signal when detected.

Abstract: A method and system for modifying an audio signal, the method comprising: receiving the audio signal at signal processing means; analysing the received audio signal to identify characteristic signal components in the audio signal; determining that another signal is input to the signal processing means, the input signal resulting from an activity which generates noise in the audio signal; and selectively applying an adjusted gain to the audio signal based on the determination that the input signal is input to the signal processing means, wherein the adjusted gain is generated in dependence upon the signal strength of the identified signal components.

Abstract: A first user terminal, host terminal, method and program. The first terminal comprises: a transceiver for communicating with a plurality of other user terminals over a communication network; and communications processing apparatus, coupled to the transceiver, and arranged to participate in a call with a selected number of the other user terminals via the transceiver and communication network, the call including transmission of a voice signal from the first user terminal. The communications processing apparatus is operable in a mode whereby it temporarily discontinues transmission of the voice signal in response to detecting less than a predetermined level of activity on said voice signal, and the communications processing apparatus is further configured to selectively enable that mode in dependence on the selected number of other user terminals in the call.

Abstract: A method and computing system for suppressing noise in an audio signal, comprising: receiving the audio signal at signal processing means; determining that another signal is input to the signal processing means, the input signal resulting from an activity which generates noise in the audio signal; and selectively suppressing noise in the audio signal in dependence on the determination that the input signal is input to the signal processing means to thereby suppress the generated noise in the audio signal.

Abstract: A method of processing audio signals during a communication session between a user device and a remote node, includes receiving a plurality of audio signals at audio input means at the user device including at least one primary audio signal and unwanted signals and receiving direction of arrival information of the audio signals at a noise suppression means. Known direction of arrival information representative of at least some of said unwanted signals is provided to the noise suppression means and the audio signals are processed at the noise suppression means to treat as noise, portions of the signal identified as unwanted dependent on a comparison between the direction of arrival information of the audio signals and the known direction of arrival information.

Abstract: In an embodiment, a method of processing audio signals at a device includes receiving audio signals at a plurality of microphones of the device; processing at least one of the audio signals received by the plurality of microphones to generate a first characteristic; a beamformer applying beamformer coefficients to the received audio signals, thereby generating a beamformer output; processing the beamformer output to generate a second characteristic. An echo canceller is applied to the beamformer output, thereby suppressing, from the beamformer output, an echo resulting from audio signals output from an audio output. An operating parameter of the echo canceller is determined, using a relationship between the first and second characteristics.

Abstract: Disclosed is an audio signal processing device comprising an input for receiving a noisy audio signal, a variable gain component and a noise suppression component. The noisy audio signal has a desired audio component and a noise component. The variable gain component and the noise suppression component are respectively configured to apply a gain and a noise suppression procedure to the audio signal, thereby generating a gain adjusted noise reduced audio signal. The aggressiveness of the noise suppression procedure is rapidly changed responsive to a change in the applied gain. That change is a change from a current value by an amount substantially matching the change in applied gain to a new value. The aggressiveness is then gradually returned to the current value.

Abstract: Beamformer coefficients may include a plurality of sets of theoretical statistical data for theoretical signals. Each theoretical signal may have its own particular attributes. The statistical data may be used in computing beamformer coefficients for application by a beamformer to signals received at a device. Signals are received at an input of the device. A respective plurality of weights is determined, for the theoretical statistical data sets, based on an analysis of the extent to which the signals have the particular attributes of the theoretical signals. The theoretical are retrieved, and a statistical data set is calculated for the signals by performing a weighted sum of the theoretical statistical data sets using the determined respective plurality of weights. Beamformer coefficients are computed based on the calculated statistical data set for the signals, which are used by a beamformer to the signals for generating a beamformer output.

Abstract: A method, device and computer program product for processing audio signals, the method including determining beamformer filter coefficients to be applied to the audio signals; applying the beamformer filter coefficients to the audio signals; outputting the filtered audio signals from an audio output comprising a plurality of speakers coupled with the device; and receiving at a microphone coupled with the device, the filtered audio signals output from the audio output. The filter coefficients are determined such that the filtered audio signals are suppressed when received at the microphone.

Abstract: Audio signals are processed for use in a communication event. A data store may be queried to obtain an indication of an echo direction, which relates to a direction from which audio signals output from the audio output are likely to be received at a microphone array (plurality of microphones) of a device. Beamformer coefficients of an adaptive beamformer of the device are determined in dependence upon the received indication of the echo direction. Audio signals are received at the microphone array. The adaptive beamformer applies the determined beamformer coefficients to the received audio signals, thereby generating a beamformer output for use in the communication event. The beamformer coefficients are determined such that echo suppression is applied to audio signals received at the microphone array from the indicated echo direction.

Abstract: Method, user device and computer program product for suppressing echo. An audio signal is output from a speaker. A microphone receives an audio signal, wherein the received audio signal includes an echo resulting from the outputted audio signal. A Finite Impulse Response filter estimate ?(n) is dynamically adapted in the time domain based on the outputted audio signal and the received audio signal to model an echo path h(n) related to the echo in the received audio signal. The filter estimate ?(n) is used in an estimate of the echo power in the received audio signal, and the estimated echo power is used to apply echo suppression to the received audio signal, thereby suppressing the echo in the received audio signal.

Abstract: Method, device and computer program product for processing signals at the device. Signals are received, over a range of angles, at a plurality of sensors of the device, the received signals including an interfering signal received from an interfering source location. An interference delay pattern between receipt of signals at the sensors corresponding to receipt of a signal from the interfering source location is determined. A plurality of regularization signals having a delay pattern matching the determined interference delay pattern are generated. The generated regularization signals are used to determine beamformer coefficients to be applied by a beamformer, and the beamformer applies the determined beamformer coefficients to the signals received by the plurality of sensors, thereby generating a beamformer output.

Abstract: A method of echo cancellation in an acoustic system, comprising a first user device and at least one further user device, the method comprising, at the first user device: detecting an audio signal received from the at least one further user device over a communications network; supplying the audio signal when received to an audio signal processing module of the first user device, wherein the audio signal processing module processes the audio signal and outputs a processed audio signal to audio output means of the first user device; and controlling an echo cancellation process for cancelling echo from an audio signal received via audio input means of the first user device based on a loopback signal provided by the processed audio signal and selectively based on the received audio signal when detected.

Abstract: Signals are received, over a range of angles, at an input of a device. The signals include a primary signal with a principal direction of arrival and an interfering signal with a respective interfering direction of arrival at the input. Measurements are determined for the received signals over the range of angles. Each measurement relates to a particular angle and indicating the energy of the received signals which are received from the particular angle. For each angle over the range of angles, a value is removed from the measurement for that angle, the value being based on the minimum of: (i) the energy of the measurement for that angle, and (ii) the energy of a corresponding measurement for a corresponding angle mirrored around the principal direction of arrival, whereby the remaining values of the plurality of measurements are indicative of said at least one interfering direction of arrival.

Abstract: A method and computing system for suppressing noise in an audio signal, comprising: receiving the audio signal at signal processing means; determining that another signal is input to the signal processing means, the input signal resulting from an activity which generates noise in the audio signal; and selectively suppressing noise in the audio signal in dependence on the determination that the input signal is input to the signal processing means to thereby suppress the generated noise in the audio signal.