2884 results about "Microphone array" patented technology

A natural language interface control system for operating a plurality of devices consists of a first microphone array, a feature extraction module coupled to the first microphone array, and a speech recognition module coupled to the feature extraction module, wherein the speech recognition module utilizes hidden Markov models. The system also comprises a natural language interface module coupled to the speech recognition module and a device interface coupled to the natural language interface module, wherein the natural language interface module is for operating a plurality of devices coupled to the device interface based upon non-prompted, open-ended natural language requests from a user.

The ability to combine multiple audio signals captured from the microphones in a microphone array is frequently used in beamforming systems. Typically, beamforming involves processing the output audio signals of the microphone array in such a way as to make the microphone array act as a highly directional microphone. In other words, beamforming provides a “listening beam” which points to a particular sound source while often filtering out other sounds. A “generic beamformer,” as described herein automatically designs a set of beams (i.e., beamforming) that cover a desired angular space range within a prescribed search area. Beam design is a function of microphone geometry and operational characteristics, and also of noise models of the environment around the microphone array. One advantage of the generic beamformer is that it is applicable to any microphone array geometry and microphone type.

A system that includes a desk top assembly of a display and sensors mounted on a robotic arm. The arm moves the assembly so that it remains within position and orientation tolerances relative to the user's head as the user looks around. Near-field speaker arrays supply audio and a microphone array senses a user's voice. Filters are applied to head motion to reduce latency for arm's tracking of the head. The system is full duplex with other systems allowing immersive collaboration. Lighting and sound generation take place close to the user's head. A haptic interface device allows the user to grab the display / sensor array and move it about. Motion acts as a planar selection device for 3D data. Planar force feedback allows a user to “feel” the data. Users see not only each other through display windows, but can also see the positions and orientations of each others' planar selections of shared 3D models or data.

A system and method for teleconferencing and recording of meetings. The system uses a variety of capture devices (a novel 360° camera, a whiteboard camera, a presenter view camera, a remote view camera, and a microphone array) to provide a rich experience for people who want to participate in a meeting from a distance. The system is also combined with speaker clustering, spatial indexing, and time compression to provide a rich experience for people who miss a meeting and want to watch it afterward.

Targeted sound detection methods and apparatus are disclosed. A microphone array has two or more microphones M0 . . . MM. Each microphone is coupled to a plurality of filters. The filters are configured to filter input signals corresponding to sounds detected by the microphones thereby generating a filtered output. One or more sets of filter parameters for the plurality of filters are pre-calibrated to determine one or more corresponding pre-calibrated listening zones. Each set of filter parameters is selected to detect portions of the input signals corresponding to sounds originating within a given listening zone and filter out sounds originating outside the given listening zone. A particular pre-calibrated listening zone is selected at a runtime by applying to the plurality of filters a set of filter coefficients corresponding to the particular pre-calibrated listening zone. As a result, the microphone array may detect sounds originating within the particular listening sector and filter out sounds originating outside the particular listening zone. Sounds are detected with the microphone array. A particular listening zone containing a source of the sound is identified. The sound or the source of the sound is characterized and the sound is emphasized or filtered out depending on how the sound is characterized.

Targeted sound detection methods and apparatus are disclosed. A microphone array has two or more microphones M0 . . . MM. Each microphone is coupled to a plurality of filters. The filters are configured to filter input signals corresponding to sounds detected by the microphones thereby generating a filtered output. One or more sets of filter parameters for the plurality of filters are pre-calibrated to determine one or more corresponding pre-calibrated listening zones. Each set of filter parameters is selected to detect portions of the input signals corresponding to sounds originating within a given listening zone and filter out sounds originating outside the given listening zone. A particular pre-calibrated listening zone is selected at a runtime by applying to the plurality of filters a set of filter coefficients corresponding to the particular pre-calibrated listening zone. As a result, the microphone array may detect sounds originating within the particular listening sector and filter out sounds originating outside the particular listening zone.

In one embodiment, a directional microphone array having (at least) two microphones generates forward and backward cardioid signals from two (e.g., omnidirectional) microphone signals. An adaptation factor is applied to the backward cardioid signal, and the resulting adjusted backward cardioid signal is subtracted from the forward cardioid signal to generate a (first-order) output audio signal corresponding to a beampattern having no nulls for negative values of the adaptation factor. After low-pass filtering, spatial noise suppression can be applied to the output audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order output audio signals.

Embodiments include methods and apparatuses for sensing acoustic waves for a conferencing application. A conferencing apparatus includes a plurality of directional microphones oriented to cover a corresponding plurality of direction vectors and disposed in a housing. An orientation sensor is configured to generate an orientation signal indicative of an orientation of the housing. A processor is operably coupled to the plurality of directional microphones and the orientation sensor. The processor is configured to automatically adjust a signal processing characteristic of one or more directional microphones of the plurality of directional microphones responsive to the orientation signal.

Systems and methods for utilizing inter-microphone level differences (ILD) to attenuate noise and enhance speech are provided. In exemplary embodiments, primary and secondary acoustic signals are received by omni-directional microphones, and converted into primary and secondary electric signals. A differential microphone array module processes the electric signals to determine a cardioid primary signal and a cardioid secondary signal. The cardioid signals are filtered through a frequency analysis module which takes the signals and mimics a cochlea implementation (i.e., cochlear domain). Energy levels of the signals are then computed, and the results are processed by an ILD module using a non-linear combination to obtain the ILD. In exemplary embodiments, the non-linear combination comprises dividing the energy level associated with the primary microphone by the energy level associated with the secondary microphone. The ILD is utilized by a noise reduction system to enhance the speech of the primary acoustic signal.

An apparatus for measuring a plurality of loudspeakers arranged at different positions includes a generator of a test signal for a loudspeaker; a microphone device configured for receiving a plurality of different sound signals in response to one or more loudspeaker signals emitted by one of the loudspeakers in response to the test signal; a controller for controlling emissions of the loudspeaker signals by the loudspeakers and for handling the different sound signals so that a set of sound signals recorded by the microphone device is associated with each loudspeaker in response to the test signal; and an evaluator for evaluating the set of sound signals for each loudspeaker to determine at least one loudspeaker characteristic for each loudspeaker and for indicating a loudspeaker state using the at least one loudspeaker characteristic. This scheme allows automatic, efficient and accurate measurement of loudspeakers arranged in a three-dimensional configuration.

The present invention provides a solution to the needs described above through a method and apparatus for reducing echo and noise. The apparatus includes a microphone array for receiving and audio signal, the audio signal including a voice signal component and a noise signal component. The apparatus further includes a voice processing path having an input coupled to the microphone array and a noise processing path having an input coupled to the microphone array. The voice processing path is adapted to detect voice signals and the noise processing path is adapted to detect noise signals. A first echo controller is coupled to the voice processing path and a second echo controller is coupled to the noise processing path. A noise reducer is coupled to the output of the first echo controller and second echo controller.

Embodiments of the present invention relate to methods, systems, and computer program products for signal processing. A first plurality of microphone signals is obtained by a first microphone array. A second plurality of microphone signals is obtained by a second microphone array different from the first microphone array. The first plurality of microphone signals is beamformed by a first beamformer comprising beamforming weights to obtain a first beamformed signal. The second plurality of microphone signals is beamformed by a second beamformer comprising the same beamforming weights as the first beamformer to obtain a second beamformed signal. The beamforming weights are adjusted such that the power density of echo components and / or noise components present in the first and second plurality of microphone signals is substantially reduced.

Systems and methods are disclosed that facilitate real-time information exchange in a multimedia conferencing environment. Data Client(s) facilitate data collaboration between users and are maintained separately from audio / video (AV) Clients that provide real-time communication functionality. Data Clients can be remotely located with respect to one another and with respect to a server. A remote user Stand-in Device can be provided that comprises a display to present a remote user to local users, a digital automatic pan / tilt / zoom camera to capture imagery in, for example, a conference room and provide real-time information to an AV Client in a remote office, and a microphone array that can similarly provide real-time audio information from the conference room to an AV Client in the remote office. The invention further facilitates file transfer and presentation broadcast between Data Clients in a single location or in a plurality of disparate locations.

A communication system (e.g., a speakerphone) includes an array of microphones, a speaker, memory and a processor. The processor may perform a virtual broadside scan on the microphone array and analyze the resulting amplitude envelope to identify acoustic source angles. Each of the source angles may be further investigated with a directed beam (e.g., a hybrid superdirective / delay-and-sum beam) to obtain a corresponding beam signal. Each source may be classified as either intelligence or noise based on an analysis of the corresponding beam signal. The processor may design a virtual beam pointed at an intelligence source and having nulls directed at one or more of the noise sources. Thus, the virtual beam may be highly sensitive to the intelligence source and insensitive to the noise sources.

According to the present invention, a tilt sensor is used to determine the tilt angle of a speakerphone and the surface on which it rests. This information is used to optimize both the receive and transmit signals for the chosen tilt angle. The information can also be used to adjust performance of any beamformer(s) where the speakerphone incorporates a microphone array or loudspeaker array. In one embodiment, vibrational data is provided by the tilt sensor for enhancing the receive signal and acoustic echo cancellation.

Systems and methods are disclosed that facilitate real-time information exchange in a multimedia conferencing environment. Data Client(s) facilitate data collaboration between users and are maintained separately from audio / video (AV) Clients that provide real-time communication functionality. Data Clients can be remotely located with respect to one another and with respect to a server. A remote user Stand-in Device can be provided that comprises a display to present a remote user to local users, a digital automatic pan / tilt / zoom camera to capture imagery in, for example, a conference room and provide real-time information to an AV Client in a remote office, and a microphone array that can similarly provide real-time audio information from the conference room to an AV Client in the remote office. The invention further facilitates file transfer and presentation broadcast between Data Clients in a single location or in a plurality of disparate locations.

A method and system for enhancing a target sound signal from multiple sound signals is provided. An array of an arbitrary number of sound sensors positioned in an arbitrary configuration receives the sound signals from multiple disparate sources. The sound signals comprise the target sound signal from a target sound source, and ambient noise signals. A sound source localization unit, an adaptive beamforming unit, and a noise reduction unit are in operative communication with the array of sound sensors. The sound source localization unit estimates a spatial location of the target sound signal from the received sound signals. The adaptive beamforming unit performs adaptive beamforming by steering a directivity pattern of the array of sound sensors in a direction of the spatial location of the target sound signal, thereby enhancing the target sound signal and partially suppressing the ambient noise signals, which are further suppressed by the noise reduction unit.

An array of microphones utilizes two sets of two microphones for noise suppression. A primary microphone and secondary microphone of the three microphones may be positioned closely spaced to each other to provide acoustic signals used to achieve noise cancellation. A tertiary microphone may be spaced with respect to either the primary microphone or the secondary microphone in a spread-microphone configuration for deriving level cues from audio signals provided by tertiary and the primary or secondary microphone. Signals from two microphones may be used rather than three microphones. The level cues are expressed via an inter-microphone level difference (ILD) which is used to determine one or more cluster tracking control signals. The ILD based cluster tracking signals are used to control the adaptation of null-processing noise cancellation modules. A noise cancelled primary acoustic signal and ILD based cluster tracking control signals are used during post filtering to adaptively generate a mask to be applied against a speech estimate signal.

A speaker array and microphone arrays positioned on both sides of the speaker array are provided. A plurality of focal points each serving as a position of a talker are set in front of the microphone arrays respectively symmetrically with respect to a centerline of the speaker array, and a bundle of sound collecting beams is output toward the focal points. Difference values between sound collecting beams directed toward the focal points that are symmetrical with respect to the centerline are calculated to cancel sound components that detour from the speaker array to microphones. Then, it is estimated based on totals of squares of peak values of the difference values for a particular time period that the position of the talker is close to which one of the focal points, and the position of the talker is decided by comparing the totals of the squares of the peak values of the sound collecting beams directed to the focal points that are symmetrical mutually.

A telepresence video conference endpoint device includes spaced-apart microphone arrays each configured to transduce sound into corresponding sound signals. A processor receives the sound signals from the arrays and determines a direction-of-arrival (DOA) of sound at each array based on the set of sound signals from that array, determines if each array is blocked or unblocked based on the DOA determined for that array, selects an array among the arrays based on whether each array is determined to be blocked or unblocked, and perform subsequent sound processing based on one or more of the sound signals from the selected array.

Methods and apparatus for signal processing are disclosed. A discrete time domain input signal xm(t) may be produced from an array of microphones M0 . . . MM. A listening direction may be determined for the microphone array. The listening direction is used in a semi-blind source separation to select the finite impulse response filter coefficients b0, b1 . . . , bN to separate out different sound sources from input signal xm(t). One or more fractional delays may optionally be applied to selected input signals xm(t) other than an input signal x0(t) from a reference microphone M0. Each fractional delay may be selected to optimize a signal to noise ratio of a discrete time domain output signal y(t) from the microphone array. The fractional delays may be selected to such that a signal from the reference microphone M0 is first in time relative to signals from the other microphone(s) of the array. A fractional time delay Δ may optionally be introduced into an output signal y(t) so that: y(t+Δ)=x(t+Δ)*b0+x(t−1+Δ)*b1+x(t−2+Δ)*b2+ . . . +x(t−N+Δ)bN, where Δ is between zero and ±1.

The embodiment of the invention provides a microphone array adjustment method, a microphone array and an electronic device and relates to the field of digital signal processing so that wave beams can be adjusted dynamically according to a space position of an acoustic source and thus the quality of audio acquisition is improved. The microphone array adjustment method is applicable to the microphone array which includes at least one microphone used for collecting voices made by the acoustic source. The microphone array adjustment method includes the following steps: detecting a space parameter of the acoustic source in a space; determining a corresponding wave-beam forming parameter according to the detected space parameter; and according to the wave-beam forming parameter, adjusting the microphone array so as to enable the microphone array wave-beam forming to conform to the space position of the acoustic source.

A system is provided for controlling personalized settings in a vehicle. The system includes a microphone for receiving spoken commands from a person in the vehicle, a location recognizer for identifying location of the speaker, and an identity recognizer for identifying the identity of the speaker. The system also includes a speech recognizer for recognizing the received spoken commands. The system further includes a controller for processing the identified location, identity and commands of the speaker. The controller controls one or more feature settings based on the identified location, identified identity and recognized spoken commands of the speaker. The system also optimizes on the beamforming microphone array used in the vehicle.

A failure detection system includes an omnidirectional microphone array device having a plurality of microphone elements and a directivity control device that calculates a delay time of a voice propagated from a sound source to each microphone element and forms a directivity of the voice using the delay time and the voice collected by the omnidirectional microphone array device, and detects a failure of the microphone element. A smoothing unit calculates an average power of one microphone element. An average calculator calculates a total average power of a plurality of usable microphone elements included in the omnidirectional microphone array device. A comparison unit compares whether or not a difference between the average power and the total average power exceeds a range of ±6 dB, and determines whether the microphone element is in failure based on the comparison result.

A system that includes a desk top assembly of a display and sensors mounted on a robotic arm. The arm moves the assembly so that it remains within position and orientation tolerances relative to the user's head as the user looks around. Near-field speaker arrays supply audio and a microphone array senses a user's voice. Filters are applied to head motion to reduce latency for arm's tracking of the head. The system is full duplex with other systems allowing immersive collaboration. Lighting and sound generation take place close to the user's head. A haptic interface device allows the user to grab the display / sensor array and move it about. Motion acts as a planar selection device for 3D data. Planar force feedback allows a user to “feel” the data. Users see not only each other through display windows, but can also see the positions and orientations of each others' planar selections of shared 3D models or data.

A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone signal. A processor determines the speaker's direction relative to the microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals.

A beamforming method employs a plurality of microphones arranged in an array with respect to a reference point. The method includes acquiring microphone signals from the microphones and combining the microphone signals (x1 . . . xM) to obtain Virtual Microphones, combining the microphone signals to obtain a pair of directional Virtual Microphones having respective signals determining respective patterns of radiation with a same origin corresponding to the reference point and rotated at different pattern direction angles, defining a separation angle between them, obtaining a sum radiation signal of a sum Virtual Microphone with a sum radiation pattern, associating a respective weight to the signals of the pair of directional Virtual Microphones, obtaining respective weighted signals of radiation and summing the weighted signals, computing respective weights as a function of a determined pattern direction angle of the pattern of radiation of the pair of directional Virtual Microphones and of the separation angle.

An audio device is provided, employing a defect detection method to detect defectiveness within a microphone array. The microphone array comprising a first microphone and a second microphone, respectively, generates a first audio signal and a second audio signal from ambient audio signals. An error detector is provided to detect functions of the first and second microphones based on the first and second audio signals to generate a status signal. A digital signal processor (DSP) processes the first and second audio signals based on the status signal. If the status signal indicates that only the first microphone or the second microphone is defective, the DSP switches to a single microphone mode in which only the remaining normal microphone is enabled. If the status signal indicates that both the first and second microphones are defective, the DSP generates an error indication signal and stops processing the first and second audio signals.