Koninklijke Philips Corporation N.v.

U.S. Philips Corporation

Nxp B.v.

U.S. Phillips Corporation

Cornelis Pieter Janse

Rene Martinus Maria Derkx

Harm Jan Willem Belt

Patrick Kechichian

Alexander Franciscus Kolen

Franciscus Hendrikus Van Heesch

Harold Agnes Wilhelmus Schmeitz

Alberto Fazzi

David Antoine Christian Marie Roovers

Leonardus M Driessen

Leon C A Van Stuivenberg

Patrick A Timmermans

Harm J W Belt

Sriram Srinivasan

Ivo Leon Diane Marie Merks

Bahaa Eddine Sarroukh

Wouter Joos Tirry

Jakob Van De Laar

Wei Pien Lee

Teunis Johannes Vink

Alexander Cornelis Geerlings

Mahdi Triki

Rik Jozef Martinus Janssen

Brian Brand Antonius Johannes Bloemendal

Franciscus A Van De Laar

Gerardus Paul Maria Egelmeers

Paul D Lahaije

Johannes M Meijer

Paulo M Castello Da Costa

Ronald Petrus Nicolaas Duisters

Leonardus Cornelis Antonius Van Stuivenberg

Rainer Thaden

Corrado Boscarino

Felip Riera-Palou

Bahaa Eddine Sarrukh

Brian Brand Antonius Johannes Bloememdal

Bart Leonardus Martinus Smeets

David Paul Walker

Rene Martinus Marina Derkx

Rene M M Derkx

An audio capture apparatus comprises a first beamformer () which is arranged to generate a beamformed audio output signal. An adapter () adapts beamform parameters of the first beamformer and a detector () detects an attack of speech in the beamformed audio output signal. A controller () controls the adaptation of the beamform parameters to occur in a predetermined adaptation time interval determined in response to the detection of the attack of speech. The beamformer () may generate noise reference signal(s) and the detector () may be arranged to detect the attack of speech in response to a comparison of a signal level of the beamformed audio output signal relative to a signal level of the at least one noise reference signal.

Audio capture using beamforming

An audio capture apparatus comprises a microphone array () and a beamformer () arranged to generate a beamformed audio output signal and a noise reference signal. A first and second transformer () generates a first and second frequency domain signal from a frequency transform of the beamformed audio output signal and noise reference signal respectively. A difference processor () generates time frequency tile difference measures which for a given frequency is indicative of a difference between a monotonic function of a norm (magnitude) of a time frequency tile value of the first frequency domain signal and a monotonic function of a norm of a time frequency tile value of the second frequency domain signal for the first frequency. An estimator () generates an estimate indicative of whether the audio output signal comprises a point audio source in response to a combined difference value for time frequency tile difference measures for frequencies above a frequency threshold.

An apparatus for capturing audio comprises a first beamformer () coupled to a microphone array () and arranged to generate a first beamformed audio output. A plurality of constrained beamformers () each generates a constrained beamformed audio output. A first adapter () adapts beamform parameters of the first beamformer () and a second adapter () adapts constrained beamform parameters for the plurality of constrained beamformers (). A difference processor () determines a difference measure for the constrained beamformers () where the difference measure is indicative of the difference between beams formed by the first beamformer () and the constrained beamformers (). The second adapter () is arranged to adapt constrained beamform parameters with the constraint that beamform parameters are adapted only for constrained beamformers of the plurality of constrained beamformers () for which a difference measure has been determined that meets a similarity criterion.

Method and apparatus for audio capture using beamforming

A beamforming audio capture apparatus comprises a microphone array () which is coupled to a first beamformer () and a second beamformer (). The beamformers () are filter-and-combine beamformers comprising a plurality of beamform filters each having an adaptive impulse response. A difference processor () determines a difference measure between beams of the first beamformer () and the second beamformer () in response to a comparison of the adaptive impulse responses of the two beamformers (). The difference measure may e.g. be used to combine the output signals of the beamformers (). An improved difference measure less sensitive to e.g. diffuse noise may be provided.

An audio signal processing apparatus comprises a receiver () receiving an audio signal sampled at a first sampling frequency, the audio signal having a maximum frequency below half the first sampling frequency by a first frequency margin. A filter bank () generates subband signals for the digital audio signal using overlapping sub-filters. A first frequency shifter () applies a frequency shift to at least one subband of the set of subbands and a decimator () decimates the subband signals by a decimation factor resulting in a decimated sampling frequency being at least twice a bandwidth of each of the overlapping sub-filters. The frequency shift for a subband is arranged to shift the subband to a frequency interval being a multiple of a frequency interval from zero to half the decimated sample frequency. The subband may be individually processed and the processed subbands may subsequently be combined to generate a full band output signal.

Audio signal processing

An inhaler with a housing (H) comprising an air-inlet (A_I) and a an air-outlet (A_O). Inside the housing (H) a flow path (FP) is defined between air-inlet (A_I) and air-outlet (A_O) where a dispenser (DP) is arranged to dispense an aerosol or a dry powder in the flow path (FP). Two sensors (S, S), e.g. microphones, are positioned spaced apart at external surfaces of the housing (H) to sense sound or vibrations resulting from a flow in the flow path (FP) at two different positions. This allows a precise detection of flow velocity during inhalation based on the sound or vibrations sensed by the two sensors (S, S), thus allowing examination of correct use of the inhaler. Further, the use of two spaced apart sensors (S, S) facilitates identification of priming and firing events in the sensed sound or vibrations which may also be used in evaluating the use of the inhaler. Preferably, a noise reduction algorithm is used that exploits the differences in the captured sound or vibrations from the two sensors (S, S), so at to allow precise flow measurements even in noisy environments.

Inhaler with two microphones for detection of inhalation flow

Communications equipment for use by a call center to enable communications between the call center and one or more user devices is provided, with the communications equipment comprising audio processing circuitry for processing audio signals received from the user devices over a communications network, wherein the processing circuitry is configured to perform acoustic echo cancellation and/or noise suppression and/or dereverberation processing on the received audio signals to cancel acoustic echoes and/or suppress noise and/or suppress reverberation that is present in the audio signals received from the user device.

Audio signal processing in a communication system

An audio echo suppressor includes a first receiver for receiving a first audio signal for rendering by a loud-speaker and a second receiver for receiving a microphone signal. A linear echo-cancellation filter generates a first compensation signal from the first audio signal and a compensator generates a residual signal by compensating the microphone signal for the first compensation signal. A first adapter determines filter parameters for the linear echo-cancellation filter. An estimator generates distortion measures where each distortion measure is indicative of the contribution to the residual signal in a frequency interval outside a first frequency interval. The residual signal results from rendering of signal components of the first audio signal that are within the first frequency interval. An echo reducer performs echo suppression based on the distortion measures.

Echo cancellation

An echo cancellation device including a first adaptive filter for producing a first echo cancellation signal, a second adaptive filter for producing a second echo cancellation signal, and a post-processor for suppressing any remaining echo. The first adaptive filter and the second adaptive filter are designed for canceling a first part of the echo impulse response and a second part of the echo impulse response respectively. The device may be utilized in a mobile telephone.

An improved determination of an acoustic coupling between a loudspeaker and a microphone, which results in a better estimate of the acoustic coupling during double talk, is achieved in a device for determining an acoustic coupling between a far-end talker signal as produced by the loudspeaker and a combined signal as picked up by the microphone. The combined signal comprises an echo of the far-end talker signal and a near-end talker signal. The acoustic coupling is derived from a ratio of changes in an envelope of the combined signal to changes in the envelope of the far-end talker signal in a predetermined time interval. In this way, a systematic error during double talk is strongly reduced.

Determining an acoustic coupling between a far-end talker signal and a combined signal

A multi-channel acoustic echo canceller arrangement comprises a microphone () providing a microphone signal having contributions from at least two audio sources () to be cancelled. An echo canceling circuit () performs echo cancellation of the two audio sources () based on channel estimates for channels from each of the audio sources () to the microphone (). An estimation circuit () generates each of the channel estimates as a combination of a previous channel estimate and a channel estimate update where the combination includes applying a relative weight to the channel estimate update relative to the previous channel estimate. A weight processorvaries the relative weight in response to a time value. The arrangement may provide improved echo-cancellation for scenarios wherein the rendering of sound from the audio sources () is time varying, such as when time varying decorrelation filters are used.

Acoustic multi-channel cancellation

A microphone system is provided, wherein the microphone system comprises a microphone array comprising a plurality of microphone units each adapted to generate a primary signal indicative of an acoustic wave received from the respective microphone unit, a first echo cancellation unit, an integrator unit, and a combination unit, wherein the microphone system is adapted to generate a first dipole response and a monopole response from the primary signals, wherein the integrator unit is adapted to generate a first integrated dipole response by integrating the first dipole response, wherein the first echo cancellation unit is adapted to generate a first echo cancelled integrated dipole response from the first integrated dipole response, and wherein the combination unit is adapted to combine the monopole response and the first echo cancelled integrated dipole response.

Microphone system and method of operating the same

A digital image () which is displayed to a user () is modified to include an aspect () of a detected at least one characteristic of the user () to give the user () the impression that the user () is present within the scene displayed by the digital image ().

Method and apparatus for modifying a digital image

A planar audio tracking system comprises a square array of four microphones (M, M, M, M) arranged as first and second cross-dipole microphones and a virtually constructed monopole microphone. The signals from these microphones undergo directional pre-processing and the results are applied to a filtered sum beamformer (FSB) (). The FSB identifies functions (h(0), h(&#x3c0;/2), and h) of the FSB which are representative of impulse responses from desired audio source(s) to the first and second cross-dipole and the monopole microphone, respectively. The functions of the first cross-dipole and the monopole microphones and the functions of the second cross-dipole and the monopole microphones are cross correlated to produce respective estimates (&#x3c8;(l) and &#x3c8;(l)) representative of the lag of the most dominant audio source. An angle-estimate ({circumflex over (&#x3c6;)}) of the most dominant source is determined using the estimates of lag. Other embodiments of the tracking system may comprise 3 microphones arranged in a circular array and forming first and second cross-dipoles and a virtual monopole.

Method of, and apparatus for, planar audio tracking

A method for detecting speech using a first microphone adapted to produce a first signal (x), and a second microphone adapted to produce a second signal (x), the method comprising the steps of:

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Eindhoven, Netherlands

Cornelis Pieter Janse