9619 results about "Sound sources" patented technology

Sound processing methods and apparatus are provided. A sound capture unit is configured to identify one or more sound sources. The sound capture unit generates data capable of being analyzed to determine a listening zone at which to process sound to the substantial exclusion of sounds outside the listening zone. Sound captured and processed for the listening zone may be used for interactivity with the computer program. The listening zone may be adjusted based on the location of a sound source. One or more listening zones may be pre-calibrated. The apparatus may optionally include an image capture unit configured to capture one or more image frames. The listening zone may be adjusted based on the image. A video game unit may be controlled by generating inertial, optical and/or acoustic signals with a controller and tracking a position and/or orientation of the controller using the inertial, acoustic and/or optical signal.

Acoustic energy is used to control motion in a fluid. According to one embodiment, the invention directs acoustic energy at selected naturally occurring nucleation features to control motion in the fluid. In another embodiment, the invention provides focussed or unfocussed acoustic energy to selectively placed nucleation features to control fluid motion. According to one embodiment, the invention includes an acoustic source, a controller for controlling operation of the acoustic source, and one or more nucleation features located proximate to or in the fluid to be controlled.

A listener can experience the sound of virtual loudspeakers over headphones with a level of realism that is difficult to distinguish from the real loudspeaker experience. Sets of personalized room impulse responses (PRIRs) are acquired for the loudspeaker sound sources over a limited number of listener head positions. The PRIRs are then used to transform an audio signal for the loudspeakers into a virtualized output for the headphones. Basing the transformation on the listener's head position, the system can adjust the transformation so that the virtual loudspeakers appear not to move as the listener moves the head.

A portable assistive listening system for enhancing sound for hearing impaired individuals includes a fully functional hearing aid and a separate handheld digital signal processing (DSP) device. The focus of the present invention is directed to the handheld DSP device and a unique method of processing incoming audio signals. The DSP device includes a programmable digital signal processor, a UWB transceiver for communicating with the hearing aid and/or other wireless audio sources, an LCD display, and a user input device (keypad). The handheld device is user programmable to apply different sound enhancement algorithms for enhancing sound signals received from the hearing aid and/or other audio source. The handheld device is capable of receiving audio signals from multiple sources, and gives the user control over selection of incoming sound sources and selective enhancement of sound. In the context of being user programmable, the digital signal processing device includes a software platform that provides for the ability of the user to select, or plug-in, desired enhancement algorithms for application to selected incoming audio signals. Specifically, the invention focuses on a method of buffering an incoming audio signal, and selectively replaying the buffered audio. The method further includes converting the replayed audio signal to text for display on the handheld DSP device.

A two-channel phase-amplitude stereo encoding and decoding scheme enabling flexible and spatially accurate interactive 3-D audio reproduction via standard audio-only two-channel transmission. The encoding scheme allows associating a 2-D or 3-D positional localization to each of a plurality of sound sources by use of frequency independent inter-channel phase and amplitude differences. The decoder is based on frequency-domain spatial analysis of 2-D or 3-D directional cues in a two-channel stereo signal and re-synthesis of these cues using any preferred spatialization technique, thereby allowing faithful reproduction of positional audio cues and reverberation or ambient cues over arbitrary multi-channel loudspeaker reproduction formats or over headphones, while preserving source separation despite the intermediate encoding over only two audio channels.

Systems and methods for determining ICP based on parameters that can be measured using non-invasive or minimally invasive techniques are provided, wherein a non-linear relationship is used to determine ICP based on one or more variable inputs. The first variable input relates to one or more properties of a cranial blood vessel and/or blood flow, such as acoustic backscatter from an acoustic transducer having a focus trained on a cranial blood vessel, flow velocity in a cranial blood vessel, and the like. Additional variables, such as arterial blood pressure (ABP), may be used in combination with a first variable input relating to one or more properties of a cranial blood vessel, such as flow velocity of the middle cerebral artery (MCA) to derive ICP using a non-linear relationship. Methods and systems for locating target areas based on their acoustic properties and for acoustic scanning of an area, identification of a target area of interest based on acoustic properties, and automated focusing of an acoustic source and/or detector on a desired target area are also provided. Acoustic transducer assemblies are described.

PCT No. PCT/DK95/00089 Sec. 371 Date Dec. 27, 1996 Sec. 102(e) Date Dec. 27, 1996 PCT Filed Feb. 27, 1995 PCT Pub. No. WO95/23493 PCT Pub. Date Aug. 31, 1995A method and apparatus for simulating the transmission of sound from sound sources to the ear canals of a listener encompasses novel head-related transfer functions (HTFs), novel methods of measuring and processing HTFs, and novel methods of changing or maintaining the directions of the sound sources as perceived by the listener. The measurement methods enable the measurement and construction of HTFs for which the time domain descriptions are surprisingly short, and for which the differences between listeners are surprisingly small. The novel HTFs can be exploited in any application concerning the simulation of sound transmission, measurement, simulation, or reproduction. The invention is particularly advantageous in the field of binaural synthesis, specifically, the creation, by means of two sound sources, of the perception in the listener of listening to sound generated by a multichannel sound system. It is also particularly useful in the designing of electronic filters used, for example, in virtual reality systems, and in the designing of an "artificial head" having HTFs that approximate the HTFs of the invention as closely as possible in order to make the best possible representation of humans by the artificial head, thereby making artificial head recordings of optimal quality.

A method and apparatus for processing an audio sound source to create four-dimensional spatialized sound. A virtual sound source may be moved along a path in three-dimensional space over a specified time period to achieve four-dimensional sound localization. A binaural filter for a desired spatial point is applied to the audio waveform to yield a spatialized waveform that, when the spatialized waveform is played from a pair of speakers, the sound appears to emanate from the chosen spatial point instead of the speakers. A binaural filter for a spatial point is simulated by interpolating nearest neighbor binaural filters chosen from a plurality of pre-defined binaural filters. The audio waveform may be processed digitally in overlapping blocks of data using a Short-Time Fourier transform. The localized sound may be further processed for Doppler shift and room simulation.

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.

Apparatus includes generator generating reference signal based on noise emitted from sound source, detector detecting level of reference signal and change in level, unit comparing change with threshold-value range and produce compared result, filter filtering reference signal, adaptive filter having variable filter coefficient, unit updating filter coefficient according to change of level of reference signal for obtaining an updated filter coefficient, unit stopping updating of filter coefficient in response to compared result when change falls outside threshold-value range, unit storing updated filter coefficient each time filter coefficient is updated, generator generating control signal using stored filter coefficient, unit generating control sound based on control signal, microphone detecting synthesis sound pressure of control sound and noise to produce an error signal, and unit setting stored filter coefficient to more accurate coefficient than stored filter coefficient based on error signal, and signal acquired by filtering control signal through filter.

An electronic device and a method for recognizing a speech are provided. The method for recognizing a speech by an electronic device includes: receiving sounds generated from a sound source through a plurality of microphones; calculating power values from a plurality of audio signals generated by performing signal processing on each sound input through the plurality of microphones and calculating direction information on the sound source based on the calculated power values and storing the calculated direction information; and performing the speech recognition on a speech section included in the audio signal based on the direction information on the sound source. As a result, the electronic device may correctly detect only a speech section from an audio signal while improving a speech section detection related processing speed.

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.

A protective case for an electronic device includes a main housing and a lid. The main housing and lid are removably joined to define a water tight volume receiving an electronic device. Air trapped within the protective case transfers acoustic energy from a sound source within the case to at least one membrane wherein the membrane vibrates in response to an air pressure differential transmitting sound to an exterior of the case.

A method and apparatus for capturing image and sound during interactivity with a computer program is provided. The apparatus includes an image capture unit that is configured to capture one or more image frames. Also provided is a sound capture unit. The sound capture unit is configured to identify one or more sound sources. The sound capture unit generates data capable of being analyzed to determine a zone of focus, at which to process sound to the substantial exclusion of sounds outside of the zone of focus. In this manner, sound that is captured and processed for the zone of focus is used for interactivity with the computer program.

A multi-channel audio system is constituted using at least one line array speaker unit, in which plural speakers are arrayed in line, wherein the same audio signal is supplied with a prescribed delay time to each of the speakers, thus forming plural sound beams. The plural sound beams are reflected on a wall surface and a ceiling of a room so as to form plural virtual sound sources surrounding a listening position, and emission directions and intensities of the sound beams are controlled so as to localize a phantom at a prescribed position based on the plural virtual sound sources. By appropriately arranging plural line array speaker units horizontally, vertically, and slantingly in such a way that each line array speaker unit forms sound beams distributed and spread in a sectorial form, it is possible to realize a surround audio system having a high degree of freedom with regard to setup positions for forming virtual sound sources.

A self calibrating dipole microphone formed from two omni-directional acoustic sensors. The microphone includes a sound source acoustically coupled to the acoustic sensors and a processor. The sound source is excited with a test signal, exposing the acoustic sensors to acoustic calibration signals. The responses of the acoustic sensors to the calibration signals are compared by the processor, and one or more correction factors determined. Digital filter coefficients are calculated based on the one or more correction factors, and applied to the output signals of the acoustic sensors to compensate for differences in the sensitivities of the acoustic sensors. The filtered signals provide acoustic sensor outputs having matching responses, which are subtractively combined to form the dipole microphone output.

The invention relates to an automatic focusing method and apparatus, and belongs to the technical field of photography. The method comprises: during a focusing process, acquiring sound information of an environment; according to the sound information, analyzing a sound source position of the sound information; and performing automatic focusing on a target object at the sound source position. According to the invention, a sounding object is focused through the sound source position; the problem is solved that in a conventional touch control type automatic focusing method, focusing has to be controlled through a touch screen by a user, and in case that the user is at a state when he cannot conveniently operate an electronic device, such as a state when the user holds a tablet device by two hands and a state when the user controls the electronic device by use of a remote controller, automatic focusing cannot be applied; and the effect of normal focusing when the user cannot conveniently operate the electronic device is realized.

The present invention relates to an acoustical insulation material containing a first layer formed from a nonwoven web having a density of at least 50 kg/m³ wherein the nonwoven web is formed from thermoplastic [meltblown] fibers having an average fiber diameter of less than about 7 microns; and a second layer of a high loft material. The high loft material of the present invention provides bulk to the first layer and may or may not have sound attenuating properties. Examples of the high loft material include, for example, fiberglass and high loft nonwoven webs. Also disclosed in a method of attenuating sound waves passing from a sound source area to a second area. The method includes positioning an acoustical insulation material containing a first layer formed from a nonwoven web having a density of at least 50 kg/m³ wherein the nonwoven web is formed from thermoplastic [meltblown] fibers having an average fiber diameter of less than about 7 microns; and a second layer of a high loft material, between the sound source area and the second area.

Methods and apparatus for signal processing are disclosed. A discrete time domain input signal x_m(t) may be produced from an array of microphones M₀ . . . M_M. 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 . . . , b_N to separate out different sound sources from input signal x_m(t). One or more fractional delays may optionally be applied to selected input signals x_m(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)b_N, where Δ is between zero and ±1.

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 desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with processor (30). Processor (30) receives aural signals from two sensors (22, 24) each at a different location. The two inputs to processor (30) are converted from analog to digital format and then submitted to a discrete Fourier transform process to generate discrete spectral signal representations. The spectral signals are delayed to provide a number of intermediate signals, each corresponding to a different spatial location relative to the two sensors. Locations of the noise source and the desired source, and the spectral content of the desired signal are determined from the intermediate signal corresponding to the noise source locations. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal with output device (90). Techniques to localize multiple acoustic sources are also disclosed. Further, a technique to enhance noise reduction from multiple sources based on two-sensor reception is described.

Systems and methods for providing object-oriented audio are described. Audio objects can be created by associating sound sources with attributes of those sound sources, such as location, velocity, directivity, and the like. Audio objects can be used in place of or in addition to channels to distribute sound, for example, by streaming the audio objects over a network to a client device. The objects can define their locations in space with associated two or three dimensional coordinates. The objects can be adaptively streamed to the client device based on available network or client device resources. A renderer on the client device can use the attributes of the objects to determine how to render the objects. The renderer can further adapt the playback of the objects based on information about a rendering environment of the client device. Various examples of audio object creation techniques are also described.