1578results about "Transducer acoustic reaction prevention" patented technology

A system and method are provided for smoothing the in-room acoustic magnitude response of an audio reproduction system. An in-room acoustic magnitude response analysis is performed to determine a room resonance induced peak associated with an audio signal. A replica of the audio signal is filtered at the room resonance induced peak. The filtered replica signal is added with the audio signal. Through this, smoothing of the room resonance induced peak may be achieved, such that a subjective impression of transient response and dynamics of the audio signal are preserved.

A sound reproducing apparatus and a sound reproducing method. The sound reproducing apparatus includes an actual listening environment feature function database where an actual listening space feature function is stored for correcting the virtual source in response to a feature of an actual listening space provided at the time of listening; and an actual listening space feature correcting unit of reading out the actual listening space feature function stored in the actual listening environment feature function database, and correcting the virtual source based on the reading result. Accordingly, causes of each distortion may be removed to provide sounds having the best quality.

Spatial noise suppression for audio signals involves generating a ratio of powers of difference and sum signals of audio signals from two microphones and then performing noise suppression processing, e.g., on the sum signal where the suppression is limited based on the power ratio. In certain embodiments, at least one of the signal powers is filtered (e.g., the sum signal power is equalized) prior to generating the power ratio. In a subband implementation, sum and difference signal powers and corresponding the power ratio are generated for different audio signal subbands, and the noise suppression processing is performed independently for each different subband based on the corresponding subband power ratio, where the amount of suppression is derived independently for each subband from the corresponding subband power ratio. In an adaptive filtering implementation, at least one of the audio signals can be adaptively filtered to allow for array self-calibration and modal-angle variability.

Speakers deployed in a space and divided into groups associated with different zones produce a mix of sounds that create internal noise pollution and, combined with external noise, an unpleasant environment for occupants. The present invention contemplates sound systems and methods for creating personalized sound zones to address these and related problems.

An active control of an unwanted noise signal at a listening site radiated by a noise source uses a reference signal that has an amplitude and / or frequency such that it is masked for a human listener at the listening site by the unwanted noise signal and / or a wanted signal present at the listening site in order to adapt for the time-varying secondary path in a real time manner such that a user doesn't fell disturbed by an additional artificial noise source.

Improved vocal signals are obtained in headsets and similar devices by including a microphone inside a chamber formed at least in part by the wearer's ear. This second microphone provides a reduced noise input signal. The reduced noise signal is corrected by input from another microphone, located outside the chamber. This correction can include echo cancellation, spectral shaping, frequency extension, and the like.

A tinnitus method and device for providing relief to a person suffering from the disturbing effects of tinnitus is described. The method can be implemented entirely in software to spectrally modify an audio signal in accordance with a predetermined masking algorithm which modifies the intensity of the audio signal at selected frequencies. A predetermined masking algorithm is described which provides intermittent masking of the tinnitus wherein, at a comfortable listening level, during peaks of the audio signal the tinnitus is completely obscured, whereas during troughs the perception of the tinnitus occasionally emerges. In practice it has been found that such intermittent masking provides an immediate sense of relief, control and relaxation for the person, whilst enabling sufficient perception of the tinnitus for habituation and long term treatment to occur. Advantageously the predetermined masking algorithm is specifically tailored to the audiometric configuration of the person. For example, the masking algorithm may be partly tailored to the hearing loss characteristic of the person. A tinnitus rehabilitation device used in conjunction with a personal sound reproducing system is also described.

A voice enhancement logic improves the perceptual quality of a processed signal. The voice enhancement system includes a noise detector and a noise attenuator. The noise detector detects and models the noise associated with rain. The noise attenuator dampens or reduces the rain noise from a signal to improve the intelligibility of an unvoiced, a fully voiced, or a mixed voice segment.

A sound masking system for shaping the ambient noise level in a physical environment. The sound masking system comprises a networked and distributed system having a number of master units coupled together and to a control unit. One or more of the master units may include satellite sound masking units which function to reproduce the sound masking signal generated by the master sound masking unit. Each of the master units is addressable over the network by the control unit enabling the control unit to program the contour, spectral band, and gain characteristics of the sound masking output signal. The system may also include a remote control unit which provides the capability to tune and adjust each master sound masking unit in situ without requiring physical access through the ceiling installation.

An arrangement and method for converting an input signal into a mechanical or acoustical output signal by using a transducer and additional means for generating a desired transfer behavior and for protecting said transducer against overload. Transducers of this kind are for example loudspeaker, headphones and other mechanical or acoustical actuators. The additional means comprise a controller, a power amplifier and a detector. The detector identifies parameters of the transducer model if the stimulus provides sufficient excitation of the transducer. The detector permanently identifies time variant properties of the transducer for any stimulus supplied to the transducer. The controller provided with this information generates a desired linear or nonlinear transfer behavior; in particular electric control linearizes, stabilizes and protects the transducer against electric, thermal and mechanical overload at high amplitudes of the input signal.

A sound masking system and method for providing sound masking in networked workstations or offices. The sound masking system comprises a communication network, a plurality of sound masking devices adapted to emit a sound masking signal, and a remote controller. One or more of the sound masking devices includes a controller, and the controller includes an interface for receiving information from the communication network and a component for controlling or adjusting output characteristics associated with the sound masking signal, such as volume. The controller includes a component responsive to a user input for setting or varying output characteristics associated with the sound masking signal.

A sound pickup apparatus selecting one of a plurality of microphones and performing echo cancellation processing for a plurality of microphones with an echo canceller to output a sound. The sound pickup apparatus sets a “learning mode” when the power supply is turned on, outputs a calibration sound from an echo cancellation calibration sound generator via a speaker, detects an echo at that time with a microphone and obtains an echo cancellation use parameter canceling the echo.

A voice enhancement system is provided for improving the perceptual quality of a processed voice signal. The system improves the perceptual quality of a received voice signal by removing unwanted noise from a voice signal recorded by a microphone or from some other source. Specifically, the system removes sounds that occur within the environment of the signal source but which are unrelated to speech. The system is especially well adapted for removing transient road noises from speech signals recorded in moving vehicles. Transient road noises include common temporal and spectral characteristics that can be modeled. A transient road noise detector employs such models to detect the presence of transient road noises in a voice signal. If transient road noises are found to be present, a transient road noise attenuator is provided to remove them from the signal.

A tinnitus method and device for providing relief to a person suffering from the disturbing effects of tinnitus is described. The method can be implemented entirely in software to spectrally modify an audio signal in accordance with a predetermined masking algorithm which modifies the intensity of the audio signal at selected frequencies. A predetermined masking algorithm is described which provides intermittent masking of the tinnitus wherein, at a comfortable listening level, during peaks of the audio signal the tinnitus is completely obscured, whereas during troughs the perception of the tinnitus occasionally emerges. In practice it has been found that such intermittent masking provides an immediate sense of relief, control and relaxation for the person, whilst enabling sufficient perception of the tinnitus for habituation and long term treatment to occur. Advantageously the predetermined masking algorithm is specifically tailored to the audiometric configuration of the person. For example, the masking algorithm may be partly tailored to the hearing loss characteristic of the person. A tinnitus rehabilitation device used in conjunction with a personal sound reproducing system is also described.

A method comprises processing M subband communication signals and N target-cancelled signals in each subband with a set of beamformer coefficients to obtain an inverse target-cancelled covariance matrix of order N in each band; using a target absence signal to obtain an initial estimate of the noise power in a beamformer output signal averaged over recent frames with target absence in each subband; multiplying the initial noise estimate with a noise correction factor to obtain a refined estimate of the power of the beamformer output noise signal component in each subband; processing the refined estimate with the magnitude of the beamformer output to obtain a postfilter gain value in each subband; processing the beamformer output signal with the postfilter gain value to obtain a postfilter output signal in each subband; and processing the postfilter output subband signals to obtain an enhanced beamformed output signal.

A method and device for tinnitus therapy. The method includes generating pure sounds, each having a predetermined frequency, within an audible range, and waiting for a user to press an input button when the user hears the pure sound. Then, the hearing characteristics of the user are interpreted in conjunction with equal loudness contours. From this interpretation, either a tinnitus masking method or a tinnitus retraining therapy are selected according to the hearing characteristics of the user.

A noise reduction device of the present invention comprises a control filter unit for generating a control sound signal to cancel out a noise, a control speaker for outputting a control sound according to the control sound signal from the control filter unit, an error microphone for detecting a residual sound by superimposing the noise upon the control sound output from the control speaker, and an obstacle detector for detecting an obstacle around the error microphone, wherein the control filter unit generates the control sound signal according to data from the error microphone and the obstacle detector.

Systems and methods for noise suppression using noise subtraction processing are provided. The noise subtraction processing comprises receiving at least a primary and a secondary acoustic signal. A desired signal component may be calculated and subtracted from the secondary acoustic signal to obtaining a noise component signal. A determination may be made of a reference energy ratio and a prediction energy ratio. A determination may be made as to whether to adjust the noise component signal based partially on the reference energy ratio and partially on the prediction energy ratio. The noise component signal may be adjusted or frozen based on the determination. The noise component signal may then be removed from the primary acoustic signal to generate a noise subtracted signal which may be outputted.

A speech processing apparatus which, in the process of performing echo canceling by using a pseudo acoustic echo signal, continuously uses an impulse response used for the previous frame as an impulse response to generate the pseudo acoustic echo signal when a voice is contained in the microphone input signal, and which uses a newly updated impulse response when a voice is not contained in the microphone input signal.

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.

A first device obtains first audio from one or more microphones. The first device generates a first audio signal from the first audio. The first device analyzes the first audio signal to determine noise associated with the first audio. The first device receives a second audio signal from a second device, and processes the second audio signal based at least in part on the determined noise by identifying one or more frequencies of the second audio signal that are between 1-2 Kilohertz. The first device then outputs a modified second audio signal to a speaker.

A process for combining audio channels combines the audio channels to produce a combined audio channel and dynamically applies one or more of time, phase, and amplitude or power adjustments to the channels, to the combined channel, or to both the channels and the combined channel. One or more of the adjustments are controlled at least in part by a measure of auditory events in one or more of the channels and / or the combined channel. Applications include the presentation of multichannel audio in cinemas and vehicles. Not only methods, but also corresponding computer program implementations and apparatus implementations are included.

A tinnitus method and device for providing relief to a person suffering from the disturbing effects of tinnitus is described. The method can be implemented entirely in software to spectrally modify an audio signal in accordance with a predetermined masking algorithm which modifies the intensity of the audio signal at selected frequencies. A predetermined masking algorithm is described which provides intermittent masking of the tinnitus wherein, at a comfortable listening level, during peaks of the audio signal the tinnitus is completely obscured, whereas during troughs the perception of the tinnitus occasionally emerges. In practice it has been found that such intermittent masking provides an immediate sense of relief, control and relaxation for the person, whilst enabling sufficient perception of the tinnitus for habituation and long term treatment to occur. Advantageously the predetermined masking algorithm is specifically tailored to the audiometric configuration of the person. For example, the masking algorithm may be partly tailored to the hearing loss characteristic of the person. A tinnitus rehabilitation device used in conjunction with a personal sound reproducing system is also described.

A process for combining audio channels combines the audio channels to produce a combined audio channel and dynamically applies one or more of time, phase, and amplitude or power adjustments to the channels, to the combined channel, or to both the channels and the combined channel. One or more of the adjustments are controlled at least in part by a measure of auditory events in one or more of the channels and / or the combined channel. Applications include the presentation of multichannel audio in cinemas and vehicles. Not only methods, but also corresponding computer program implementations and apparatus implementations are included.

The invention discloses a method for eliminating noise of a long-distance microphone array and a noise-eliminating system. In the invention, the signals collected from two microphones are treated with respect to beam forming, and then the intensified target phonetic signals and the weakened target phonetic signals are obtained; whether target phonetic signals exist in the signals collected from the two microphones is further tested; the update of the adaptive filter coefficient is controlled based on the test result; and lastly, based on a controlled adaptive filter coefficient, the obtained intensified target phonetic signals and the weakened target phonetic signals undergo an adaptive filter process. According to the invention, the performance of noise elimination is greatly improved without affecting the quality of the target phonetics even in case that the microphones are not increased in number.

A feedback active noise cancellation headphone is disclosed. A plurality of microphone sensors are disposed in front of the left and the right speakers of the headphone for improving the sound reception quality of the microphone sensors, so that the active noise controlling circuit generates an inverse phase soundwave accurately for countering the low frequency noise, and further to improve the noise reduction performance of the active noise cancellation headphone.

Microphone includes: a microphone element; a simulative feedback signal generation section that generates a simulative feedback signal simulating a feedback signal generated by a sound, produced via a speaker, returning the microphone element; and an arithmetic operator that subtracts the simulative feedback signal, generated by the simulative feedback signal generation section, from a sound signal collected by the microphone element, to thereby output the subtraction result as a residual signal. The residual signal output by the arithmetic operator is supplied to an amplifier device of the speaker as an output signal of the microphone. The simulative feedback signal generation section includes a delay circuit that delays the residual signal, output by the arithmetic operator, by a given time, and an adaptive filter that generates the simulative feedback signal by filtering a previous residual signal delayed by the delay circuit. The adaptive filter updates a filter coefficient on the basis of the previous residual signal delayed by the delay circuit and a current residual signal output by the arithmetic operator.

An embodiment of the invention discloses audio equipment, terminal equipment and electronic equipment. The method includes a Type-C male connector, a first loudspeaker and a first noise reduction microphone; the Type-C male connector is coupled to the first noise reduction microphone through a first noise signal output pin and is coupled to the first loudspeaker through a first audio signal input pin; and under a circumstance that the Type-C male connector is inserted in and electrically connected with a Type-C female connector of a target device, the audio equipment outputs a first noise signal collected by the first noise reduction microphone to the target device through the first noise signal output pin, receives through the first audio signal input pin a first audio signal obtained after noise reduction processing is performed on the first noise signal, and transmits the first audio signal to the first loudspeaker. The structure of the noise-reduction audio equipment is simplified, cost is reduced, and the implementation scheme is simple.

A system and method for analyzing a signal to monitor the dynamics of its magnitude and frequency characteristics over time. An electronic circuit for identifying feedback in an audio signal, formed in accordance with embodiments of the invention may comprise a feedback control block operable to determine a candidate frequency having potential feedback such that the feedback control block is further operable to perform an iterative analysis of the magnitude of the audio signal at the candidate frequency to determine the growth characteristics of the signal. The electronic circuit may further include a test filter block operable to deploy a test filter at a candidate frequency and a permanent filter block operable to deploy a permanent filter at the candidate frequency if the feedback control block determines that the growth characteristics of the signal at the candidate frequency comprises feedback characteristics after the test filter has been deployed.