355 results about "Directional microphone" patented technology

A multi-party conversation analyzer and logger uses a variety of techniques including spectrographic voice analysis, absolute loudness measurements, directional microphones, and telephonic directional separation to determine the number of parties who take part in a conversation, and segment the conversation by speaking party. In one aspect, the invention monitors telephone conversations in real time to detect conditions of interest (for instance, calls to non-allowed parties or calls of a prohibited nature from prison inmates). In another aspect, automated prosody measurement algorithms are used in conjunction with speaker segmentation to extract emotional content of the speech of participants within a particular conversation, and speaker interactions and emotions are displayed in graphical form. A conversation database is generated which contains conversation recordings, and derived data such as transcription text, derived emotions, alert conditions, and correctness probabilities associated with derived data. Investigative tools allow flexible queries of the conversation database.

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.

Techniques are provided to suppress noise and interference using an array microphone and a combination of time-domain and frequency-domain signal processing. In one design, a noise suppression system includes an array microphone, at least one voice activity detector (VAD), a reference generator, a beam-former, and a multi-channel noise suppressor. The array microphone includes multiple microphones—at least one omni-directional microphone and at least one uni-directional microphone. Each microphone provides a respective received signal. The VAD provides at least one voice detection signal used to control the operation of the reference generator, beam-former, and noise suppressor. The reference generator provides a reference signal based on a first set of received signals and having desired voice signal suppressed. The beam-former provides a beam-formed signal based on a second set of received signals and having noise and interference suppressed. The noise suppressor further suppresses noise and interference in the beam-formed signal.

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.

In an information processing system having an equipment controller for enabling the digitization of complex work processes conducted during testing and / or operation of machinery and equipment, a local wireless communications network is implemented using at least one fixed point wireless communications access station and one or more voice-responsive wireless mobile computing / communications devices that are capable of accurate vocal command recognition in noisy industrial environments. The computing / communications device determines spectral characteristics of ambient non-speech background noises and provides active noise cancellation through the use of a directional microphone and an adaptive noise tracking and subtraction process. The device is carried by a user and is responsive to one or more vocal utterances of the user for communicating data to the information processing system and / or generating operational control commands to provide to the equipment controller for controlling the machinery or equipment.

An interactive multimedia book provides hands-on multimedia instruction to the user in response to voiced commands. The book is implemented on an easy to use computer system which is suitable to various environments in which the book might be used. The interactive multimedia book is published on a computer readable medium with the necessary software to support the interactive operation of the book. Alternatively, the book may be downloaded form a remote site using a network, such as the Internet, in which case the content of the book and the necessary software are copied to a local medium, such as a computer hard disk. The content includes both text and audio / video clips. The interactive multimedia book is accessed by a computer system which is equipped with a microphone and voice recognition software. Voiced commands and natural language queries are the primary user input to the computer system. The computer system is also equipped with a high resolution display, a voice synthesizer and a speaker or headphone system to provide output to the user. A combination headphone and directional microphone can be especially convenient in some environments as, for example, the wood shop where the headphones allow the user to better hear the instruction over the din of machine noise while at the same time protecting the user's hearing. The displayed text is written in a markup language, such as HyperText Markup Language (HTML), and contains hyperlinks which link the current topic with other related topics. The user may command the book to read the text and, as the text is read by the voice synthesizer, a word which is also a hyperlink will change its attributes upon being spoken. The user will be able to observe or hear this and, without having to click a mouse button, simply utter the word which is the hyperlink to navigate to the linked topic.

A system and method for audio telepresence. The system includes a user station and a telepresence unit. The telepresence unit includes a directional microphone for capturing sounds at the remote location, and means for converting the captured sounds into a stream of data to be communicated to the user station. The user station includes means for receiving the stream of data and a plurality of speakers for recreating the sounds of the remote location. The user station and the speakers are located within an anechoic chamber where sound reflections are substantially absorbed by anechoic linings of the chamber walls. Because of the substantial lack of sound reflection within the anechoic chamber, a user within the anechoic chamber will be able to experience an aural ambience that closely resembles the sounds captured at the remote location. The user station may include microphones for capturing the user's voice, and the telepresence unit may include speakers for projecting the user's voice at the remote location. Feedback suppression, audio direction steering, and head-coding techniques may also be used to enhance the user's sense of remote presence.

A speech from a speaker proximate to one or more microphones within an environment can be received. The microphones can be a directional microphone or an omni-directional microphone. The speech can be processed to produce an utterance to determine the identity of the speaker. The identity of the speaker can be associated with a voiceprint. The identity can be associated with a user's credentials of a computing system. The credentials can uniquely identify the user within the computing system. The utterance can be analyzed to establish a zone in which the speaker is present. The zone can be a bounded region within the environment. The zone can be mapped within the environment to determine a location of the speaker. The location can be a relative or an absolute location.

A transducer assembly utilizing at least two MEMS transducers is provided, the transducer assembly preferably defining either an omnidirectional or directional microphone. In addition to at least first and second MEMS transducers, the assembly includes a signal processing circuit electrically connected to the MEMS transducers, a plurality of terminal pads electrically connected to the signal processing circuit, and a transducer enclosure housing the first and second MEMS transducers. The MEMS transducers may be electrically connected to the signal processing circuit using either wire bonds or a flip-chip design. The signal processing circuit may be comprised of either a discrete circuit or an integrated circuit. The first and second MEMS transducers may be electrically connected in series or in parallel to the signal processing circuit. The first and second MEMS transducers may be acoustically coupled in series or in parallel.

Systems and methods that enable high quality audio teleconferencing are disclosed. In one embodiment of the present invention, a signal processor receives signals from a spatially dispersed set of directional microphones, processing the microphone signals and the far-end received audio into a signal for transmission to a far-end party. The processing may comprise the use of one or more algorithms that reduce conference room noise and may selectively increase participant audio levels by processing the microphone signals using beamforming techniques. An embodiment of the present invention may also comprise one or more omni-directional microphones that may be used in cooperation with the directional microphones to adjust for background noise, acoustic echo, and the existence of side conversations.

A system for selectively picking up a speech signal focuses on a speaker within a group of speakers who wishes to communicate something to the system using an image analysis algorithm to identify, based on a recognition feature, a position of at least one person who wishes to give the system voice commands. The detected position is then used to adapt a directional microphone to the at least one person.

A clear, high quality voice signal with a high signal-to-noise ratio is achieved by use of an adaptive noise reduction scheme with two microphones in close proximity. The method includes the use of two omini directional microphones in a highly directional mode, and then applying an adaptive noise cancellation algorithm to reduce the noise.

A microphone having an optical component for converting the sound-induced motion of the diaphragm into an electronic signal using a diffraction grating. The microphone with inter-digitated fingers is fabricated on a silicon substrate using a combination of surface and bulk micromachining techniques. A 1 mm×2 mm microphone diaphragm, made of polysilicon, has stiffeners and hinge supports to ensure that it responds like a rigid body on flexible hinges. The diaphragm is designed to respond to pressure gradients, giving it a first order directional response to incident sound. This mechanical structure is integrated with a compact optoelectronic readout system that displays results based on optical interferometry.

The invention features a miniature, second-order, microcrystalline silicon microphone diaphragm formed using silicon microfabrication techniques. The diaphragm is composed of two or more rigid diaphragm elements hinged to one another providing second- or higher-order response depending on the number of diaphragm elements used. The response of the differential diaphragm has a response that is highly dependent on the direction of the incident sound. The diaphragms are useful for constructing highly innovative microphones that have far greater directionality, better sensitivity, wider frequency response, and lower noise than is achievable with current technology.

A directional microphone system comprises two membranes that, on the one hand, are respectively acoustically connected via an air volume with one of two spatially separate sound entrance ports, and on the other hand are acoustically coupled with one another via a third air volume, as well as an output generator configured to generate at least one output signal of the directional microphone from the vibration of one of the two membranes.

Techniques of making a recording of or transmitting a sound field from either multiple monaural or directional sound signals that reproduce through multiple discrete loud speakers a sound field with spatial harmonics that substantially exactly match those of the original sound field. Monaural sound sources are positioned during mastering to use contributions of all speaker channels in order to preserve the spatial harmonics. If a particular arrangement of speakers is different than what is assumed during mastering, the speaker signals are rematrixed at the home, theater or other sound reproduction location so that the spatial harmonics of the sound field reproduced by the different speaker arrangement match those of the original sound field. An alternative includes recording or transmitting directional microphone signals, or their spatial harmonic components, and then matrixing these signals at the sound reproduction location in a manner that takes into account the specific speaker arrangement. The techniques are described for both a two dimensional sound field and the more general three dimensional case, the latter based upon using spherical harmonics.

An audio signal processing method comprises the steps of emitting a sound at a virtual sound image location in space on the outer side of a closed surface, generating measurement-based directional transfer functions corresponding to a plurality of positions on the closed surface based on a result of measuring the sound at the plurality of respective positions on the closed surface by using a directional microphone, generating composite transfer functions corresponding to the plurality of respective positions on the closed surface by respectively adding, at a specified ratio, the measurement-based directional transfer functions and auxiliary transfer functions and generating reproduction audio signals corresponding to the plurality of respective positions on the closed surface by performing a calculation process on an input audio signal in accordance with the set of composite functions.

A sound-capturing arrangement uses a set of directional microphones that lie approximately on a sphere having a diameter of 0.9 ms sound travel, which approximates the inter-aural time delay. Advantageously, one directional microphone points upward, one directional microphone points downward, and the odd number of microphones are arranged relatively evenly in the horizontal plane. On one embodiment, the signals from the microphones that point upward and downward are combined with the signals of the horizontal microphones before the signals of the horizontal microphones are transmitted or recorded.

There is provided a loudspeaker array device which has a large freedom of installation place and whose sound beam can easily be set by a user. A method for setting the sound beam of the loudspeaker array device is also provided. The loudspeaker array device (1) makes the sound beam formed by the audio signal limited to the band where the angle of the sound beam can be adjusted, sweep from 0 to 180 degrees in the front of the loudspeaker array (10) and the direct sound and the reflected sound of the sound beam are collected by a non-directional microphone (2). The collected sound data is analyzed to detect a peak equal to or above a threshold value. Symmetry of each peak is checked. When symmetry is present, the angle at which the peak has been detected is set as the angle for outputting the sound beam for each channel of the surround sound. Thus, according to the shape of the room where the loud-speaker array device is installed and the installation position, it is possible to set the sound beam emission angle at an optimal position.

The invention discloses audio signal processing equipment and an audio signal processing method as well as electronic equipment. The audio signal processing equipment comprises a microphone array, an audio localization device, a camera, an image localization device and a sound source classifier, wherein the microphone array comprises a plurality of directional microphones having different sound pickup areas; the audio localization device is used for identifying a first group of sound sources and for determining position of each sound source in an audio coordinate system; the camera is used for capturing scene images of a current scene, wherein the current scene at least covers the sound pickup areas of the plurality of directional microphones; the image localization device is used for identifying a second group of sound sources and for determining position of each sound source in an image coordinate system; and the sound source classifier is used for classifying each sound source in the first and second groups of sound sources in accordance with a registration relation between audio and the image coordinate system, the position of each sound source in the audio coordinate system as well as the position of each sound source in the image coordinate system. Therefore, the precise classification of the sound sources can be achieved on the basis of double localization of the directional microphones and the camera.

A microphone capsule for an in-the-ear hearing aid is disclosed. The capsule can include a top plate having first and second spaced openings defining front and rear sound inlets, and a directional microphone cartridge enclosing a diaphragm. The diaphragm is oriented generally perpendicular to the top plate and divides the directional microphone cartridge housing into a front chamber and a rear chamber. A front sound passage communicates between the front sound inlet and the front chamber, and a rear sound passage communicates between the rear sound inlet and the rear chamber. Front and rear acoustic damping resistors having selected resistance values are associated with the front and rear sound passages. The acoustic resistor pair provides a selected time delay, such as about 4 microseconds, between the front and rear sound passages. The use of two acoustic resistors instead of one levels the frequency response, compared to the frequency response provided by a rear acoustic damping resistor alone.

A portable computer system contained within a housing that comprises an electronic muscle material for performing a plurality of functions. The electronic muscle can be used to replace buttons or keys used in many PDAs. The movement of the electronic muscle material can charge the battery. When handled, the electronic muscle material can further detect the left- or right-handedness of the user and, based on the handedness, can generate function buttons or other alterations to accommodate the user's hand preference and finger placement. The electronic muscle can change shape to accommodate the user's hand for comfort and, further, as a security function to identify and authorize the user. Additionally, the electronic muscle material in the housing can be caused to vibrate at given frequencies so that it can generate an alarm and can function as a speaker or a dynamically directional microphone.

The present invention provides a highly directional audio response that is flat over five octaves or more by the use of multiple colinear arrays followed by signal processing. Each of the colinear arrays has a common center, but a different spacing so that it can be used for a different frequency range. The response of the microphones for each spacing are combined and filtered so that when the filtered responses are added, the combined response is flat over the selected frequency range. To improve the response, the output of the microphones for a given array spacing can also be filtered with windowing functions. To receive the response from other directions a "steering" delay may also be introduced in the microphone signals before they are combined. The invention also extends to two and three dimensional arrays.

Apparatus and methods are provided for enabling a microphone module that connects to a headset jack on a game controller. The microphone may include directional or omni-directional microphone element(s) that pick up voice input from people near a peripheral, such as a game controller, but reject background sounds from the room. User controls may also be available to set the microphone gain level, either manually or automatically, and / or to mute or unmute the microphone. The microphone of the invention can be used on a wired or wireless game controller. When used with a wireless game controller, wireless microphone capabilities and freedom of movement are provided without headdress requirements imposed by headsets.

A hearing aid apparatus is disclosed that employs both an omnidirectional microphone and at least one directional microphone of at least the first order. The electrical signals output from the directional microphone are supplied to an equalization amplifier which at least partially equalizes the amplitude of the low frequency electrical signal components of the electrical signal with the amplitude of the mid and high frequency electrical signal components of the electrical signals of the directional microphone. A switching circuit accepts the signals output from both the omnidirectional microphone and the directional microphone. The switching circuit connects the signal from the omnidirectional microphone to an input of a hearing aid amplifier when the switching circuit is in a first switching state, and connects the output of the equalization circuit to the hearing aid amplifier input when the switching circuit is in a second switching state. The switching circuit may be automatically switched in response to sensed ambient noise levels.