Abstract: A mobile communication device and a method of mitigating wind noise in a speech signal generated at a microphone of the mobile communication device. A primary microphone of the mobile communication device receives speech and wind noise and generates a first speech signal based on the speech and the wind noise that is received. A secondary microphone of the mobile communication device receives attenuated speech and generates a second speech signal based on the attenuated speech that is received. The secondary microphone is shielded from the wind noise when the primary microphone receives speech and the wind noise. A processor of the mobile communication device processes the first speech signal to replace a low frequency portion of the first speech signal with the second speech signal in order to mitigate the wind noise in the first speech signal.

Abstract: A mobile communication device and a method of mitigating wind noise in a speech signal generated at a microphone of the mobile communication device. A primary microphone of the mobile communication device receives speech and wind noise and generates a first speech signal based on the speech and the wind noise that is received. A secondary microphone of the mobile communication device receives attenuated speech and generates a second speech signal based on the attenuated speech that is received. The secondary microphone is shielded from the wind noise when the primary microphone receives speech and the wind noise. A processer of the mobile communication device processes the first speech signal to replace a low frequency portion of the first speech signal with the second speech signal in order to mitigate the wind noise in the first speech signal.

Abstract: A method, system and apparatus for loudspeaker excursion domain processing are provided. At a device comprising: a processor, a loudspeaker comprising a voice coil, one or more devices for determining loudspeaker voltage and current as a function of time, t, and a memory storing a Bl product for the loudspeaker, loudspeaker currents I(t) and corresponding voltages V(t) are received at the processor from the one or more devices. A current-from-voltage transfer function HIV(?) is derived from the loudspeaker currents I(t) and voltages V(t), as a function of frequency, ?. A Fourier space excursion-from-voltage transfer function HXV(?) is determined, whose form is constrained by parameters HIV(?), Bl, Rvc, and Lvc, where: Rvc comprises a resistance of the voice coil; and Lvc comprises an inductance of the voice coil. Filter coefficients are determined using HXV(?), which are used in a filter applied to an input signal for the loudspeaker.

Abstract: A method, system and apparatus for loudspeaker excursion domain processing are provided. At a device comprising: a processor, a loudspeaker comprising a voice coil, one or more devices for determining loudspeaker voltage and current as a function of time, t, and a memory storing a Bl product for the loudspeaker, loudspeaker currents I(t) and corresponding voltages V(t) are received at the processor from the one or more devices. A current-from-voltage transfer function HIV(?) is derived from the loudspeaker currents I(t) and voltages V(t), as a function of frequency, ?. A Fourier space excursion-from-voltage transfer function HXV(?) is determined, whos e form is constrained by parameters HIV(?), Bl, Rvc, and Lvc, where: Rvc comprises a resistance of the voice coil; and Lvc comprises an inductance of the voice coil. Filter coefficients are determined using HXV(?), which are used in a filter applied to an input signal for the loudspeaker.

Abstract: Techniques for use in improving audio quality with use of an acoustic leak compensation (ALC) system in a mobile device are described. The mobile device includes a receiver and a microphone which is acoustically coupled to the receiver. A change in a signal power of signals received at the microphone is detected. In response to the detecting, a probe signal is enabled, and a frequency response between the receiver and the microphone is estimated using the probe signal as an input. Filter coefficients of a filter are calculated based on the estimated frequency response, and the calculated filter coefficients are applied to the filter. The filter type may be selected from a plurality of filter types based on an estimated signal-to-noise ratio (SNR) of the microphone signal.

Abstract: Audio input may be received at one or more microphones of a mobile device. Based on the audio input, a change in an acoustic environment of the device, such as a change in a direction of arrival of the audio input or a degradation in a quality of acoustic echo cancellation being performed upon the audio input, may be detected. A determination may be made that the detected change coincides with a non-acoustic physical event detected using an auxiliary sensor at the mobile device. The event may for example be device motion, a new proximate object, a change in a proximity of an object, a new heat source or a change in a heat level from a known heat source. Based on the determining, a signal processor, possibly comprising an audio beamformer or echo canceller, may be recalibrated, e.g. the audio beamformer or echo canceller may be caused to reconverge.

Abstract: A telephonic device having a speakerphone function has a loudspeaker and a plurality of microphones. The plurality of microphones are coupled to a plurality of beamformers which produce a first beamform having a spatial null in the direction of the loudspeaker and a second beamform having a spatial null in the direction of near-end signals. The two beamforms are then processed to facilitate echo reduction. By comparing the beamform powers, a state of double-talk can be determined and the determination can be used to enhance echo reduction associated with speakerphone functionality.

Abstract: A telephonic device having a speakerphone function has a loudspeaker and a plurality of microphones. The plurality of microphones are coupled to a plurality of beamformers which produce a first beamform having a spatial null in the direction of the loudspeaker and a second beamform having a spatial null in the direction of near-end signals. The two beamforms are then processed to facilitate echo reduction. By comparing the beamform powers, a state of double-talk can be determined and the determination can be used to enhance echo reduction associated with speakerphone functionality.

Abstract: Techniques for use in improving audio quality with use of an acoustic leak compensation (ALC) system in a mobile device are described. The mobile device includes a receiver and a microphone which is acoustically coupled to the receiver. A change in a signal power of signals received at the microphone is detected. In response to the detecting, a probe signal is enabled, and a frequency response between the receiver and the microphone is estimated using the probe signal as an input. Filter coefficients of a filter are calculated based on the estimated frequency response, and the calculated filter coefficients are applied to the filter. The filter type may be selected from a plurality of filter types based on an estimated signal-to-noise ratio (SNR) of the microphone signal.

Abstract: Audio input may be received at one or more microphones of a mobile device. Based on the audio input, a change in an acoustic environment of the device, such as a change in a direction of arrival of the audio input or a degradation in a quality of acoustic echo cancellation being performed upon the audio input, may be detected. A determination may be made that the detected change coincides with a non-acoustic physical event detected using an auxiliary sensor at the mobile device. The event may for example be device motion, a new proximate object, a change in a proximity of an object, a new heat source or a change in a heat level from a known heat source. Based on the determining, a signal processor, possibly comprising an audio beamformer or echo canceller, may be recalibrated, e.g. the audio beamformer or echo canceller may be caused to reconverge.