2214results about "Signal processing" 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.

A context-aware voice guidance method is provided that interacts with other voice services of a user device. The voice guidance does not provide audible guidance while the user is making a verbal request to any of the voice-activated services. Instead, the voice guidance transcribes its output on the screen while the verbal requests from the user are received. In some embodiments, the voice guidance only provides a short warning sound to get the user's attention while the user is speaking on a phone call or another voice-activated service is providing audible response to the user's inquires. The voice guidance in some embodiments distinguishes between music that can be ducked and spoken words, for example from an audiobook, that the user wants to pause instead of being skipped. The voice guidance ducks music but pauses spoken words of an audio book in order to provide voice guidance to the user.

A unified communications system enables a user to simultaneously manage real time voice communication with background audio streams such as a music. In one embodiment, the system comprises a plurality of devices connected through a persistent voice over IP channel, where each device is playing background audio independently, and each device is operatively coupled to a microphone. When a connected user speaks, Voice Activity Detection results in the automatic adjustment of background audio on other connected devices, the adjustments based upon user-input settings on each device.

A method of providing navigation on an electronic device when the display screen is locked. The method receives a verbal request to start navigation while the display is locked. The method identifies a route from a current location to a destination based on the received verbal request. While the display screen is locked, the method provides navigational directions on the electronic device from the current location of the electronic device to the destination. Some embodiments provide a method for processing a verbal search request. The method receives a navigation-related verbal search request and prepares a sequential list of the search results based on the received request. The method then provides audible information to present a search result from the sequential list. The method presents the search results in a batch form until the user selects a search result, the user terminates the search, or the search items are exhausted.

An ear terminal includes a sealing section arranged for use in the ear meatus of a human, with an inner microphone having a sound inlet for being directed into the meatus and an electronic unit including filtering elements coupled to the inner microphone for filtering the signal from the inner microphone, the filtering elements being programmable to transform the signals based on the sounds received in the ear by the inner microphone into sounds having essentially the characteristics of spoken sounds of the wearer of the ear terminal.

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 some embodiments, spatially realistic audio may be provided for a conference call. Voices from participants on the left side of a display, in a conference call, may be directed through audio on the left side of the display at the other conferencing system in the conference call (similarly for voices from the center and right side of the display). In some embodiments, two speakers may be used in the system to create synthesized stereo sound at a location specified by directional information received as side information along with the existing audio channel. The location may be determined by using beamforming with integrated microphones on a camera or speakerphone. In some embodiments, the audio signal and directional information may be sent in the form of a left audio channel and a right audio channel.

The present invention provides a process for separating a good quality information signal from a noisy acoustic environment. The separation process uses a set of a least two spaced-apart transducers to capture noise and information components. The transducer signals, which have both a noise and information component, are received into a separation process. The separation process generates one channel that is substantially only noise, and another channel that is a combination of noise and information. An identification process is used to identify which channel has the information component. The noise signal is then used to set process characteristics that are applied to the combination signal to efficiently reduce or eliminate the noise component. In this way, the noise is effectively removed from the combination signal to generate a good qualify information signal. The information signal may be, for example, a speech signal, a seismic signal, a sonar signal, or other acoustic signal.

Two or more microphones receive acoustic signals and generate audio signals that are processed to determine what portion of the audio signals result from (i) incoherence between the audio signals and/or (ii) audio-signal sources having propagation speeds different from the acoustic signals. The audio signals are filtered to reduce that portion of one or more of the audio signals. The present invention can be used to reduce turbulent wind-noise resulting from wind or other airjets blowing across the microphones. Time-dependent phase and amplitude differences between the microphones can be compensated for based on measurements made in parallel with routine audio system processing.

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.

A sound source signal from a target sound source is allowed to be separated from a mixed sound which consists of sound source signals emitted from a plurality of sound sources without being affected by uneven sensitivity of microphone elements. A beamformer section 3 of a source separation device 1 performs beamforming processing for attenuating sound source signals arriving from directions symmetrical with respect to a perpendicular line to a straight line connecting two microphones 10 and 11 respectively by multiplying output signals from the microphones 10 and 11 after spectrum analysis by weighted coefficients which are complex conjugate to each other. Power computation sections 40 and 41 compute power spectrum information, and target sound spectrum extraction sections 50 and 51 extract spectrum information of a target sound source based on a difference between the power spectrum information.

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.

Systems and methods for utilizing inter-microphone level differences to attenuate noise and enhance speech are provided. In exemplary embodiments, energy estimates of acoustic signals received by a primary microphone and a secondary microphone are determined in order to determine an inter-microphone level difference (ILD). This ILD in combination with a noise estimate based only on a primary microphone acoustic signal allow a filter estimate to be derived. In some embodiments, the derived filter estimate may be smoothed. The filter estimate is then applied to the acoustic signal from the primary microphone to generate a speech estimate.

A hearing compensation system comprises a plurality of bandpass filters having an input connected to an input transducer and each bandpass filter having an output connected to the input of one of a plurality of multiplicative automatic gain control (MAGC) circuits whose outputs are summed together and connected to the input of an output transducer. The MAGC circuits attenuate acoustic signals having a constant background level without the loss of speech intelligibility. The identification of the background noise portion of the acoustic signal is made by the constancy of the envelope of the input signal in each of the several frequency bands. The background noise that will be suppressed includes multi-talker speech babble, fan noise, feedback whistle, florescent light hum, and white noise. For use in the consumer electronics field background acoustic noise may be sensed and used to adjust gain in the various MAGC circuits so as to improve a user's listening experience, whether the user is hearing impaired or not.

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.

An adaptive signal processing system eliminates noise from input signals while retaining desired signal content, such as speech. The resulting low noise output signal delivers improved clarity and intelligibility. The low noise output signal also improves the performance of subsequent signal processing systems, including speech recognition systems. An adaptive beamformer in the signal processing system consistently updates beamforming signal weights in response to changing microphone signal conditions. The adaptive weights emphasize the contribution of high energy microphone signals to the beamformed output signal. In addition, adaptive noise cancellation logic removes residual noise from the beamformed output signal based on a noise estimate derived from the microphone input signals.

The present application relates in one aspect to a method of controlling diaphragm excursion of an electrodynamic loudspeaker. The method comprises dividing the audio input signal into at least a low-frequency band signal and a high-frequency band signal by a band-splitting network and applying the low-frequency band signal to a diaphragm excursion estimator. The instantaneous diaphragm excursion is determined based on the low-frequency band signal. The determined instantaneous diaphragm excursion is compared with an excursion limit criterion. The low-frequency band signal is limited based on a result of the comparison between the instantaneous diaphragm excursion and the excursion limit criterion to produce a limited low-frequency band signal which is combined with the high-frequency band signal to produce an excursion limited audio signal.

Embodiments disclosed herein enable detection and improvement of the quality of the audio signal using a mobile device by determining the loss in the audio signal and enhancing audio by streaming the remainder portion of audio. Embodiments disclosed herein enable an improvement in the sound quality rendered by rendering devices by emitting an test audio signal from the source device, measuring the test audio signal using microphones, detecting variation in the frequency response, loudness and timing characteristics using impulse responses and correcting for them. Embodiments disclosed herein also compensate for the noise in the acoustic space by determining the reverberation and ambient noise levels and their frequency characteristics and changing the digital filters and volumes of the source signal to compensate for the varying noise levels.

Embodiments disclosed herein may include determining a signal parameter of a first microphone and a second microphone associated with a computing device. Embodiments may include generating a reference parameter based upon at least one of the parameter of the first microphone and the parameter of the second microphone. Embodiments may include adjusting a tolerance of at least one of the first microphone and the second microphone, based upon the reference parameter. Embodiments may include receiving, at the first microphone, a first speech signal, the first speech signal having a first speech signal magnitude and receiving, at the second microphone, a second speech signal, the second speech signal having a second speech signal magnitude. Embodiments may include comparing at least one of the first speech signal magnitude and the second speech signal magnitude with a third speech signal magnitude and detecting an obstructed microphone based upon the comparison.

The sound device includes an audio-information output unit, an analysis unit, an audio-division-spectrum output unit, a noise-division-spectrum output unit and a correction unit. The analysis unit receives audio information from the audio-information output unit, and then outputs sound spectrum information. The noise-division-spectrum output unit outputs sound-volume information for each critical band width of a noise, and the audio-division-spectrum output unit outputs the sound-volume information for each critical band width of the sound-spectrum information. The correction unit corrects the information from the audio-division-spectrum output unit based on the information from the noise-division-spectrum output unit. The audio-signal properties can be well corrected corresponding to the auditory-sense properties of the human, and thus the audio sound, in which an uncomfortable feeling to the auditory sense of the human has been adequately controlled, can be transmitted to a user.

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.

The present disclosure relates to a method for processing sound source in terminal capable of receiving an outside sound source through at least two microphones and displaying a position of the outside sound source on a display unit, and a terminal using the same, wherein the terminal includes: at least two microphones configured to receive an outside sound source; a display unit configured to display a predetermined data; and a controller configured to obtain position information of the sound source using at least one of an amplitude, a phase and a period of the outside sound source received from the two microphones, and controllably display the position information of the sound source on a display unit.

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 apparatus includes first and second microphone arrangements, arranged to output first and second signals respectively and is operable in a first mode and a second mode. In the first mode, an output signal is generated based on the second signal and a third signal, where the second signal and, optionally, the first signal, can be used to compensate for ambient noise, for example, for noise cancellation when a telephone call is relayed through a speaker. In the second mode, an output signal is generated based on the first and second signals. In this manner, the combination of the first and second microphone arrangements provides a directional sensitivity that can pick up sound from a remote source, for example, in an audio or video recording session. The apparatus may include a sensor to allow automatic switching between one or more of modes, directional sensitivity patterns and types of recording session.

A sound reinforcement system which enables handsfree and high-quality sound reinforcement without requiring a person who is speaking to move to a microphone or move a microphone. At least one microphone and a plurality of speakers are arranged in a room. A speaker output adjusting section outputs sound picked up by the microphone to the plurality of speakers at predetermined levels.

The invention relates to an energy saving headset 100. The headset 100 comprises a power management unit 150 that is operable to reduce the power consumption of the headset 100 when a user 110 is not present. The power management unit 150 uses capacitive sensing to detect the presence of the user 110. Capacitive sensing is advantageous since it provides a flexible and reliable sensor that can accurately detect the presence or absence of a user 110 either by detecting user proximity or user contact. Moreover, in various embodiments, the sensitivity of a capacitive sensor may be adjusted to account for user movement or changes in environmental conditions, such as, for example, the presence of water, or sweat, on the headset 100 to further improve sensing reliability. The invention further relates to headsets using user presence signals based on capacitive sensing to control other functions of the headset or to control external devices to which the headset is connected, either wirelessly or by wires.

The invention provides a method for determining a set of filter coefficients for an acoustic echo compensator in a beamformer arrangement. The acoustic echo compensator compensates for echoes within the beamformed signal. A plurality of sets of filter coefficients for the acoustic echo compensator is provided. Each set of filter coefficients corresponds to one of a predetermined number of steering directions of the beamformer arrangement. The predetermined number of steering directions is equal to or greater than the number of microphones in the microphone array. For a current steering direction, a current set of filter coefficients for the acoustic echo compensator is determined based on the provided sets of filter coefficients.

A sound zoom method, medium, and apparatus generating a signal in which a target sound is removed from sound signals input to a microphone-array by adjusting a null width that restricts a directivity sensitivity of the microphone array, and extracting a signal corresponding to the target sound from the sound signals by using the generated signal. Thus, a sound located at a predetermined position away from the microphone array can be selectively obtained so that a target sound is efficiently obtained.

An audio processor for converting a multi-channel audio input signal, such as a B-format sound field signal, into a set of audio output signals, such as a set of two or more audio output signals arranged for headphone reproduction or for playback over an array of loudspeakers. A filter bank splits each of the input channels into frequency bands. The input signal is decomposed into plane waves to determine one or two dominant sound source directions. The(se) are used to determine a set of virtual loudspeaker positions selected such that the dominant direction(s) coincide(s) with virtual loudspeaker positions. The input signal is decoded into virtual loudspeaker signals corresponding to each of the virtual loudspeaker positions, and the virtual loudspeaker signals are processed with transfer functions suitable to create the illusion of sound emanating from the directions of the virtual loudspeakers. A high spatial fidelity is obtained due to the coincidence of virtual loudspeaker positions and the determined dominant sound source direction(s). Improved performance can be obtained in the case where Head-Related Transfer Functions are used by differentiating the phase of a high frequency part of the HRTFs with respect to frequency, followed by a corresponding integration of this part with respect to frequency after combining the components of HRTFs from different directions.