Gn Audio A/s

Other

Gn Netcom A/s

Mads Dyrholm

Martin Rung

Allan Mejlgren Von Bülow

A headset adapted to transmitting an outgoing audio signal (Tx) and comprising a voice microphone and at least one ambient microphone. The headset comprises an ambient noise reduction block (ANR), which is adapted to reduce the level of ambient noise in the outgoing audio signal (Tx). The ambient noise reduction block (ANR) includes the measuring the levels of the voice microphone signal (X) and ambient microphone signal (Y) to estimate a characteristic constant level drop between the voice microphone signal (X) and the ambient microphone signal (Y), which is characteristic for the headset user talking. A time-varying filter passes the voice microphone signal (X), when the level difference is larger than characteristic constant level drop (CLD) and attenuates the voice microphone signal (X), if the level difference is below characteristic constant level drop.

Headset with ambient noise reduction system

The present invention relates to a microphone apparatus () with a main beamformer () that provides a main output audio signal (S) as a beamformed signal by applying a main weight vector (B) to a main input vector (M). A main beamformer controller () repeatedly determines a main steering vector (d) and adaptively determines the main weight vector (B) in dependence on the main steering vector (d) and the main input vector (M) to increase the relative amount of voice sound (V) from the user () in the main output audio signal (S). The microphone apparatus () further comprises an auxiliary beamformer () that provides an auxiliary beamformer signal (S) as a beamformed signal by applying an auxiliary weight vector (B) to an auxiliary input vector (M) that is a subset of the main input vector (M), and an auxiliary beamformer controller () that adaptively determines the auxiliary weight vector (B) to increase the relative amount of voice sound (V) from the user () in the auxiliary beamformer signal (S). The main beamformer controller () determines the main steering vector (d) in dependence on the auxiliary weight vector (B). This may enable the main beamformer controller () to utilize information derived independently of the steering vector (d) and may thus improve stability and/or accuracy of the estimation of the steering vector (d), and may further reduce the computation load for the main beamformer controller ().

Microphone apparatus and headset

The present invention relates to a microphone apparatus () with a main beamformer (F, BF) that provides a directional audio output (S) by combining microphone signals (X, Y) from multiple microphones (). The quality of beamformed microphone signals normally depends on the individual microphones having equal sensitivity characteristics across the used frequency range. The invention enables automatic adaptation of the main beamformer (F, BF) to variations in microphone sensitivity and to changes in the alignment of the microphone apparatus () with respect to the user's mouth (). This is achieved by having the microphone apparatus (): estimate a suppression filter (Z) for an optimum voice-suppression beamformer (Z, BZ) based on the microphone signals (X, Y); estimate a candidate filter (W) for a candidate beamformer (W, BW) as the complex conjugate of the suppression filter (Z); estimate the performance of the candidate beamformer (W, BW); and replace a main filter (F) in the main beamformer (F, BF) with the candidate filter (W) if the candidate beamformer (W, BW) is estimated to perform better than the current main beamformer (F, BF). The invention may be used to enhance speech quality and intelligibility in headsetsand other audio devices that pick up user voice.

The present invention relates to a desktop speakerphone constructed to make the desktop speakerphone less space-consuming while having improved audio qualities. The desktop speakerphone preferably has two microphone clustersmounted at the upper side of the housingcloser towards respective longitudinal endsof the latter, so that each microphone clustercan receive voice sound Afrom one or more of the users. Each microphone clusterpreferably comprises three pressure microphones. Furthermore, within each microphone cluster, the second and third sound inletsare preferably arranged symmetrically on opposite sides of the respective median plane. Within each microphone cluster, the relative arrangement of the three sound inletsdefines a respective microphone axisfor each of the microphone pairs. The microphone axisof the first microphone pairextends through the first and the second sound inlet, while the microphone axisof the second microphone pairextends through the first and the third sound inlet. The microphones are connected to multiple array processors, each configured to provide an array signal in dependence on two or more of the multiple microphone signals. First and second array processor configured to provide array signals in dependence on microphone signals.

Desktop speakerphone

The present invention relates to a desktop speakerphone array processor for microphones which produces high quality sound. The microphone array has a front direction defined by the line of sight from a microphone inlet of the second microphone towards a microphone inlet of the first microphone, the array processor being connected to receive a front microphone signal from the first microphone, a rear microphone signal from the second microphone and an audio output signal representing a speaker sound emitted from a sound driver arranged near the first and second microphones and in the rearwards hemisphere with respect to the front direction of the microphone array, the array processor being configured to provide a first array signal having a first directivity pattern with a main lobe oriented in the front direction of the microphone array in dependence on the front microphone signal, the rear microphone signal and the audio output signal. The array process has a filter configured to filter the rear microphone signal using a first set of filter coefficients and a subtractor configured to subtract the filtered signal from the front microphone signal and to provide the result in a difference signal. The filter controller repeatedly performs a cross-power analysis based on the audio output signal, the front microphone signal and the rear microphone signal and to determine the first set of filter coefficients in dependence on the result of the cross-power analysis.

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Ballerup, Denmark

Mads Dyrholm