343 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.

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.

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 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.

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.

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.

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.