Abstract: 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.

Abstract: 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.

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 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.

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: A method of processing an audio signal transmitted from a remote transmitter and received at a local receiver of an acoustic system, includes at the receiver receiving with the audio signal an indication of remote transmitter gain, determining an overall system gain of the acoustic system from the remote transmitter gain and a local receiver gain and selectively applying a system gain reduction step to the audio signal if it is determined that the overall system gain exceeds a threshold.

Abstract: A method, system and program for encoding and decoding speech according to a source-filter model whereby speech is modelled to comprise a source signal filtered by a time-varying filter. The method comprises: receiving a speech signal comprising successive frames, for each of a plurality of frames of the speech signal, deriving a first line spectral frequency vector for a first portion of the frame, and a second line spectral frequency vector for a second portion of the frame, and determining a transmit line spectral frequency vector and an interpolation factor based on the first and second line spectral frequency vectors, and on the transmit line spectral frequency vector for a preceding one of the frames.

Abstract: A method and apparatus for transmitting an audio signal over a communication channel comprising encoding the audio signal with an encoder 204 using a first sampling rate, filtering the audio signal using a first cut off frequency, the first cut off frequency being chosen in dependence upon the first sampling rate, and transmitting the encoded and filtered audio signal over the communication channel. The presence of a condition in which the sampling rate of the encoder 204 is to be switched to a second sampling rate at a switching time is determined and if the condition has been determined to be present, the cut off frequency used in the filtering step is gradually changed from the first cut off frequency to a second cut off frequency, the second cut off frequency being chosen in dependence upon the second sampling rate, such that the audio bandwidth of the transmitted signal changes gradually when the sampling rate is switched to the second sampling rate.

Abstract: 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.

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: 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: 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 statistical data sets 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: Mobile device, method and computer program product for processing signals at the mobile device. The signals are received at a plurality of signal sensors of the mobile device. Motion of the mobile device is sensed and the received signals are processed using beamforming means at the mobile device, in dependence upon their direction of arrival at the plurality of signal sensors and in dependence upon the sensed motion of the mobile device.

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: 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: 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 of transmitting a combined audio signal to at least one of a plurality of participants in a communication event comprising; receiving an audio signal from each of said plurality of participants together with audio activity information associated with each of said received audio signals; analysing a measure of audio activity for each received audio signal based on the audio activity information associated with each of said received signals, wherein said measure of audio activity allows audio signals comprising audio activity to be compared; selecting a set of audio signals from said received audio signals based on the analysed measure of audio activity for each signal; decoding said set of audio signals; and combining said set of audio signals to generate said combined audio signal to be transmitted to said at least one of said plurality of participants.

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: A method, user device and computer program product for processing audio signals during a communication session between a user device and a remote node. The method comprising: receiving a plurality of audio signals at audio input means at the user device including at least one primary audio signal and unwanted signals; receiving direction of arrival information of the audio signals at a gain control means; providing to the gain control means known direction of arrival information representative of at least some of said unwanted signals; processing the audio signals at the gain control means by applying a level of gain to generate a gain controlled signal for transmission to the remote node, wherein the level of gain applied is dependent on a comparison between the direction of arrival information of the audio signals and the known direction of arrival information.

Abstract: A method of processing an audio signal transmitted from a remote transmitter and received at a local receiver of an acoustic system, includes at the receiver receiving with the audio signal an indication of remote transmitter gain, determining an overall system gain of the acoustic system from the remote transmitter gain and a local receiver gain and selectively applying a system gain reduction step to the audio signal if it is determined that the overall system gain exceeds a threshold.