614results about "Microphone spatial/constructional arrangements" patented technology

Acoustic device including a member extending transversely of its thickness and capable of sustaining bending waves at least over an intendedly consequentially acoustically active area of the transverse extent of said member, the member having, by reason of orderly design methodology disclosed and claimed, a distribution of resonant modes of its natural bending wave vibration at least over said area that is dependent on values of particular parameters of said members, including geometrical configuration and directional bending stiffness(es), which values have been selected to predetermine said distribution of natural resonant modes being consonant with required achievable acoustic action of said member for operation of said device over a desired operative acoustic frequency range.

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.

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.

A sound collecting system includes a plurality of microphones, a distance estimation module and an adjustment module. The distance estimation module estimates a distance to a user to accordingly provide a user distance. The adjustment module adjusts a part or all of the positions of the microphones according to the user distance.

A new approach to tracking head motion for headphone-based sound is described. Called MTB2.0 for “Motion-Tracked Binaural with 2 Channels”, the method may be used for any 2-channel binaural signals without any increase in the bandwidth requirement. MTB2.0 provides a simple and effective means to improve the quality of headphone-based sound reproduction by sensing the orientation of the listener's head and using the sensed orientation to appropriately modify the signals sent to the two ears. MTB2.0 method increases the realism and removes some of the shortcomings of binaural sound capture and recording, as well as improves the quality of binaural rendering of stereo.

Embodiments of the disclosure can include systems, methods, and apparatus for assigning three-dimensional spatial data to sounds and audio files. In one embodiment, a method can include receiving at least one audio signal, receiving sonic spatial data, associating the sonic spatial data with the at least one audio signal, associating the at least one audio signal and sonic spatial data with a time code, and storing the sonic spatial data, the at least one audio signal, and time code in an encoded sound file.

The preferred embodiment of the present invention provides a system and method for programming and / or charging one or more audio / video devices such that the audio / video devices will be programmed and charged to receive transmitted audio and video signals associated with an event, allowing a user to use the audio / video device to observe the sights and sounds of the event. A preferred embodiment of the present invention includes a cart with a docking port for each of a plurality of personal audio / video devices, a charger configured to charge the power source of each personal audio / video display device, and programming logic configured to program each of the personal audio / video devices.

A method and a system for simultaneously generating configurable three-dimensional (3D) sounds are provided. A 3D sound processing application (3DSPA) in operative communication with a microphone array system (MAS) is provided on a computing device. The MAS forms acoustic beam patterns and records sound tracks from the acoustic beam patterns. The 3DSPA generates a configurable sound field on a graphical user interface using recorded or pre-recorded sound tracks. The 3DSPA acquires user selections of configurable parameters associated with sound sources from the configurable sound field. The 3DSPA dynamically processes the sound tracks using the user selections to generate a configurable 3D binaural sound, surround sound, and / or stereo sound. The 3DSPA measures head related transfer functions (HRTFs) in communication with a simulator apparatus that simulates a human's upper body. The 3DSPA generates the binaural sound by processing the sound tracks with the HRTFs based on the user selections.

An apparatus (300) comprises—one or more inputs (IN1, IN2) to receive two or more different audio signals (S1, S2),—an input (IN—VDI1) to receive a direction signal (SVDI1) and—a signal processing unit (100) arranged to generate a processed audio signal (SAUDIOI) from said two or more different audio signals (S1, S2), said processed audio signal (SAUDIO1) comprising an enhanced audio signal (SENC1) corresponding to a sound (SND2) originating from a location (x2y2) indicated by said direction signal (SVDI1).

A ceiling tile adapted to be supported in an overhead opening and function as a loudspeaker. The tile is in the form of a distributed mode acoustic radiator with a transducer mounted wholly and exclusively on the radiator at a location for coupling to resonant bending wave modes so as to vibrate the radiator and cause it to resonate.

A microphone array-based audio system that supports representations of auditory scenes using second-order (or higher) harmonic expansions based on the audio signals generated by the microphone array. In one embodiment, a plurality of audio sensors are mounted on the surface of an acoustically rigid polyhedron that approximates a sphere. The number and location of the audio sensors on the polyhedron are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeams having at least one eigenbeam of order two (or higher). Beamforming (e.g., steering, weighting, and summing) can then be applied to the resulting eigenbeam outputs to generate one or more channels of audio signals that can be utilized to accurately render an auditory scene.

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.

Conventional audio compression technologies perform a standardized signal transformation, independent of the type of the content. Multi-channel signals are decomposed into their signal components, subsequently quantized and encoded. This is disadvantageous due to lack of knowledge on the characteristics of scene composition, especially for e.g. multi-channel audio or Higher-Order Ambisonics (HOA) content. An improved method for encoding pre-processed audio data comprises encoding the pre-processed audio data, and encoding auxiliary data that indicate the particular audio pre-processing. An improved method for decoding encoded audio data comprises determining that the encoded audio data had been pre-processed before encoding, decoding the audio data, extracting from received data information about the pre-processing, and post-processing the decoded audio data according to the extracted pre-processing information.

A microphone apparatus of the present invention is applied to a camera including a panel type view finder for displaying a picture taken by the camera, a touch panel for pointing a desired position of the picture displayed on the view finder, determining circuit for determining a direction of the real picture in accordance with the position pointed by the touch panel, a microphone device having a plurality of microphone units for collecting a sound that comes from the picture taken by the camera; and directivity generating circuit for generating a sound directivity signal with which the microphone device collects the sound coming from the direction determined by the determining circuit.

An audio processing apparatus includes first and second audio pickup units. The second audio pickup unit includes an audio resistor provided to cover a sound receiving portion to suppress external wind introduction while passing an external audio. A first filter attenuates a signal having a frequency lower than a first cutoff frequency of the output signal of a first A / D converter. A second filter attenuates a signal having a frequency higher than a second cutoff frequency of the output signal of a second A / D converter. A third filter is provided between the first audio pickup unit and the first A / D converter to attenuate a signal having a frequency lower than a third cutoff frequency for suppressing the wind noise.

The sound image control device filters transfer functions H3 and H1 indicating transfer characteristics of a sound from an acoustic transducer (8) to entrances to respective ear canals (1) and (2) as well as filtering transfer functions H4 and H2 from an acoustic transducer (9) to the entrances to the respective ear canals (1) and (2) and generates second transfer functions H6 and H5 indicating transfer characteristics of a sound to the entrances to the respective ear canals (1) and (2) from a target sound source (11) at a location different from the sound sources, the sound image control device being equipped with correction filters (13) and (14) that (i) store characteristic functions E1 and E2 for performing filtering operations on the first transfer functions H1, H2, H3, and H4 and (ii) generate the second transfer functions H5 and H6 from the first transfer functions H1, H2, H3, and H4 using such characteristic functions E1 and E2.

A method of identifying incidents using mobile devices can include receiving a communication from each of a plurality of mobile devices. Each communication can specify information about a detected sound. Spatial and temporal information can be identified from each communication as well as an indication of a sound signature matching the detected sound. The communications can be compared with a policy specifying spatial and temporal requirements relating to the sound signature indicated by the communications. A notification can be selectively sent according to the comparison.

A sound field microphone is provided. The sound field microphone includes a plurality of pressure-gradient microphone capsules symmetrically arranged in three dimensional space on the sides of a virtual polyhedron. The virtual polyhedron defines a first volume. A solid body is located in a-space created between the plurality of microphone capsules. The solid body occupies a second volume which is in the range of between about 1% to about 65% of the first volume.

A microphone array-based audio system that supports representations of auditory scenes using second-order (or higher) harmonic expansions based on the audio signals generated by the microphone array. In one embodiment, a plurality of audio sensors are mounted on the surface of an acoustically rigid sphere. The number and location of the audio sensors on the sphere are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeams having at least one eigenbeam of order two (or higher). Beamforming (e.g., steering, weighting, and summing) can then be applied to the resulting eigenbeam outputs to generate one or more channels of audio signals that can be utilized to accurately render an auditory scene. Alternative embodiments include using shapes other than spheres, using acoustically soft spheres and / or positioning audio sensors in two or more concentric patterns.

A system that models a microphone may include capsules that receive individual signals. The signals may be combined and modified based on a weighting factor. Directivity patterns of a converted signal may be modified or controlled based on the weighting of the signals.

A new approach to capturing and reproducing either live or recorded three-dimensional sound is described. Called MTB2.5 for “Motion-Tracked Binaural 2.5,” the method employs six real or simulated microphones, a head tracker, and special signal-processing procedures to combine the signals picked up by the microphones. In the microphone array, two central microphones are placed at the positions of the ears of a listener facing forward, and four peripheral microphones are located in pairs on either side of the central microphones. Only the low-frequency components of the four peripheral microphone signals are needed, and the total bandwidth required is approximately 2.5 times the bandwidth of one full-bandwidth audio channel. MTB2.5 achieves a high degree of realism by effectively placing the listener's ears in the space where the sounds are occurring, moving the virtual ears in synchrony with the listener's head motions. MTB2.5 also provides a universal format for recording spatial sound.

The invention concerns a headphone and a compatible recording device for spatial sound reproduction provided with two earphones, each earphone having a support defining at least partly a cap-like surface comprehensively covering the listener's ear. Each earphone includes at least five speakers arranged on said support. The speakers are adapted to reproduce an acoustic field, such that it is perceived as being continuous by the human ear, for acoustic frequencies lower than a given maximum frequency.

A sound system for capturing and reproducing sounds produced by a plurality of sound sources. The system comprises a device for receiving sounds produced by the plurality of sound sources and converting the separately received sounds to a plurality of separate audio signals without mixing the audio signals. The system may further comprise a device for separately storing the plurality of separate audio signals on a recording medium without mixing the audio signals and a device for reading the stored audio signals from the recording medium. The system further includes a reproduction system for recreating the plurality of separate audio signals. Also, the system comprises an amplification network which comprises a plurality of amplifier systems, with one or more separate amplifiers in each amplifier system for separately amplifying each of the separate audio signals. The system also comprises a loudspeaker network which comprises a plurality of loudspeaker systems with one or more separate loudspeakers in each loudspeaker system for separately reproducing the plurality of audio signals. A dynamic controller may be used to control the micro relationships of the components within a signal path and the macro relationships among the separate signal paths. The amplifiers and / or loudspeakers for each signal path may be customized based on the characteristics and complexities of the original sound to be reproduced on each signal path.

Systems and methods for adaptive processing of a close microphone array in a noise suppression system are provided. A primary acoustic signal and a secondary acoustic signal are received. In exemplary embodiments, a frequency analysis is performed on the acoustic signals to obtain frequency sub-band signals. An adaptive equalization coefficient may then be applied to a sub-band signal of the secondary acoustic signal. A forward-facing cardioid pattern and a backward-facing cardioid pattern are then generated based on the sub-band signals. Utilizing cardioid signals of the forward-facing cardioid pattern and backward-facing cardioid pattern, noise suppression may be performed. A resulting noise suppressed signal is output.