Skype Limited

Microsoft Technology Licensing, LLC

Karsten Vandborg Sorensen

Koen Bernard Vos

Jon Bergenheim

Soren Skak Jensen

Per Åhgren

Per Ahgren

Stefan Strommer

Julian Spittka

Henrik Astrom

Antonio Varanda

Tiago Loureiro

Michael Bevin

Henrik Aström

Jesus De Vicente Peña

Jesus De Vicente Pena

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.

Audio signal processing

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.

Echo cancellation

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.

Controlling multi-party communications

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.

Gain control for an audio signal

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.

Noise suppression

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.

Processing audio signals

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.

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.

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.

Processing signals

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 &#x125;(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 &#x125;(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.

Echo suppression

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.

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.

A method of estimating noise in data containing voice information and noise includes receiving the data as a sequence of input values; transforming the data by applying a first non linear mapping to the input values wherein the derivative function of the mapping decreases in magnitude as the input values increase in magnitude smoothing the transformed data; and transforming the smoothed transformed data by applying a second non linear mapping that is opposite to the first non linear mapping, to determine an estimate of the noise in the inputted data.

Method of estimating noise levels in a communication system

According to an embodiment, a method of reducing noise in a signal received at a processing stage of an acoustic system includes, at the processing stage identifying at least one frequency which causes a system gain of the acoustic system to be above an average system gain of the acoustic system; providing a noise attenuation factor for reducing noise in the signal for the at least one frequency, the noise attenuation factor for the at least one frequency based on the system gain for that frequency; and applying the noise attenuation factor to a component of the signal at that frequency.

A delay between a first signal and a second signal is estimated. The first signal and second signals are received and for each of a plurality of candidate delays between the signals, a correlation value is determined. Based on the correlation values, one of the candidate delays is selected to be used as an estimate of the delay between the first and second signals.

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Stockholm, Sweden

Karsten Vandborg Sorensen