Abstract: The present invention provides a loudspeaker with at least two diaphragm assemblies, in which the reactive forces acting on the magnets in each assembly cancel. By linking the two magnets together via a non-rigid material, resonance in the drive assemblies is highly damped and corresponding distortion of the loudspeaker output is reduced.

Abstract: Provided is an audio output device capable of preventing noises and improving the S/N ratio even if no audio signal is inputted in the middle of the input of audio signals, or even if an audio-signal input state and a no-signal state are alternately repeated. In the provided audio output device, a multiplier is provided on the input side of each of the delayers. Each multiplier multiplies the addition output of the corresponding one of adders by a multiplier coefficient supplied by the coefficient counter. If there is no input of digital audio signals into a ?? modulator, the counter control circuit decreases the output of the coefficient counter down to 0 stepwise at predetermined intervals.

Abstract: A signal processing apparatus includes a first audio output unit configured to output audio of a first audio signal input from a first signal input line, a first pickup unit connected to the first signal input line, a second audio output unit configured to output audio of a second audio signal input from a second signal input line, a second pickup unit connected to the second signal input line, a connecting line that connects the above units to ground, and a first reducing unit configured to at least reduce a first sound leakage signal, being the first audio signal leaking into the second signal input line from the first audio output unit, by using the first audio signal, or reducing a second sound leakage signal, being the second audio signal leaking into the second signal input line from the second audio output unit, by using the second audio signal.

Abstract: An electroencephalogram measurement apparatus includes: an electroencephalogram measurement section for measuring an electroencephalogram of a user by using a plurality of electrodes; an electro-acoustic transducer for presenting an acoustic signal to the user, the electro-acoustic transducer being in a vicinity of at least one electrode among the plurality of electrodes while the electroencephalogram measurement section is worn by the user; an amplitude envelope extraction section for extracting an amplitude envelope of the acoustic signal presented by the electro-acoustic transducer; a frequency analysis section for applying a frequency analysis to the amplitude envelope extracted by the amplitude envelope extraction section; and a noise estimation section for estimating an electrical noise which is mixed at the at least one electrode by using a previously provided set of transform rules and the extracted amplitude envelope.

Abstract: Methods and apparatuses for reduced microphone handling noise are disclosed. In one example, a microphone system includes a microphone to output a microphone output signal and a sensor adapted to output a sensor signal indicating whether the sensor is in proximity to or touching a user finger. A processor is adapted to process the microphone output signal using touch mode signal processing responsive to a determination the sensor is in proximity to or touching the user finger.

Abstract: An audio processing system processes an audio signal that may come from one or more microphones. The audio processing system may use information from one or more non-acoustic sensors to improve a variety of system characteristics, including responsiveness and quality. Those audio processing systems that use spatial information, for example to separate multiple audio sources, are undesirably susceptible to changes in the relative position of any audio sources, the audio processing system itself, or any combination thereof. Using the non-acoustic sensor information may decrease this susceptibility advantageously in an audio processing system.

Abstract: A method of interference suppression is provided that includes receiving a first audio signal from a first audio capture device and a second audio signal from a second audio capture device wherein the first audio signal includes a first combination of desired audio content and interference and the second audio signal includes a second combination of the desired audio content and the interference, performing blind source separation using the first audio signal and the second audio signal to generate an output interference signal and an output audio signal including the desired audio content with the interference suppressed, estimating interference remaining in the output audio signal using the output interference signal, and subtracting the estimated interference from the output audio signal to generate a final output audio signal with the interference further suppressed.

Abstract: An audio enhancement system includes a display unit configured to exhibit a waveform corresponding to a microphone signal that is subject to an audio interference. The audio enhancement system also includes an interference reduction unit coupled to the microphone signal and configured to provide a reduction in the audio interference, wherein a reduced audio interference is indicated by the waveform in real time. A microphone signal enhancement method is also provided.

Abstract: Wind noise is detected in and removed from an acoustic signal. Features may be extracted from the acoustic signal. The extracted features may be processed to classify the signal as including wind noise or not. The wind noise may be removed before or during processing of the acoustic signal. The wind noise may be suppressed by estimating a wind noise model, deriving a modification, and applying the modification to the acoustic signal. In audio devices with multiple microphones, the channel exhibiting wind noise (i.e., acoustic signal frame associated with the wind noise) may be discarded for the frame in which wind noise is detected.

Abstract: The present invention provides a novel system and method for monitoring the audio signals, analyze selected audio signal components, compare the results of analysis with a threshold value, and enable or disable noise reduction capability of a communication device.

Abstract: Noise reduction is provided to audio captured by a headset by employing spatially separated microphones provided by a headset and a mobile phone. Primary audio is captured by a headset microphone and includes both voice audio and ambient noise. Secondary audio is captured by a mobile phone microphone and includes ambient audio. Noise reduction is performed using the primary and secondary audio to generate a noise-reduce audio.

Abstract: A method and computing system for suppressing noise in an audio signal, comprising: receiving the audio signal at signal processing means; determining that another signal is input to the signal processing means, the input signal resulting from an activity which generates noise in the audio signal; and selectively suppressing noise in the audio signal in dependence on the determination that the input signal is input to the signal processing means to thereby suppress the generated noise in the audio signal.

Abstract: Systems and methods for controlling adaptivity of signal modification, such as noise suppression, using a phantom coefficient are provided. The process for controlling adaptivity comprises receiving a signal. Determinations may be made of whether an adaptation coefficient satisfies an adaptation constraint and of whether the phantom coefficient satisfies the adaptation constraint. The phantom coefficient may be updated, for example, toward a current observation. The adaptation coefficient may be updated, for example, toward the phantom coefficient, based on whether the phantom coefficient satisfies an adaptation constraint of the signal. A modified signal may be generated by applying the adaptation coefficient to the signal based on whether the adaptation coefficient satisfies the adaptation constraint. Accordingly, the modified signal may be outputted.

Abstract: A noise suppression apparatus selectively uses an adaptive beamformer and fixed beamformer for each frequency. A direction of a null of the fixed beamformer is determined from a direction of a null automatically formed by the adaptive beamformer. Filter coefficients of the adaptive beamformer based on an output power minimization rule are calculated by a minimum norm method using a norm of the filter coefficients as a constraint. The above selection is made based on, for example, a depth of a null automatically formed by the adaptive beamformer in the selection.

Abstract: A system for compensating and driving a loudspeaker includes an open loop loudspeaker controller that receives and processes an audio input signal and provides an audio output signal. A dynamic model of the loudspeaker receives the audio output signal, and models the behavior of the loudspeaker and provides predictive loudspeaker behavior data indicative thereof. The open loop loudspeaker controller receives the predictive loudspeaker behavior data and the audio input signal, and provides the audio output signal as a function of the audio input signal and the predictive loudspeaker behavior data.

Abstract: A signal processing apparatus includes: one or more detection means for detecting movement of a diaphragm of a speaker in correspondence with feedback methods that are different feedback methods; analog-to-digital conversion means for converting one or more detection signals acquired by the detection means into a digital form; feedback signal generating means for generating feedback signals corresponding to the feedback methods using the digital detection signals; synthesis means for combining an audio signal to be output as a driving signal of the speaker with the feedback signals; correction equalizer means for setting an equalizing characteristic to allow a sound reproduced by the speaker to have a target frequency characteristic by changing the digital audio signal; feedback operation setting means for setting feedback methods in which a feedback operation up to combining the audio signal with the feedback signal is performed and the feedback operation is not performed equalizing characteristic changing a

Abstract: The present invention provides a speech enhancing method for communication earphone including two parts: sending end noise reduction processing and receiving end noise reduction processing, wherein the sending end noise reduction processing part includes: determining a wearing condition of the earphone by comparing energy difference of sound signals picked up by microphones of the communication earphone; if the earphone is normally worn, subjecting the sound signal first to multi-microphone noise reduction and then to single channel noise reduction to further suppress residuary stationary noise; otherwise suppressing stationary noise in the sound signal by single channel noise reduction directly.

Abstract: One aspect of the invention provides a method for enhancing speech output by an electro-acoustical transducer in a noisy listening environment. In some embodiments, this method includes: filtering an input audio signal x(t) using a filter H(z) to produce a filtered audio signal x(t) formula (I), wherein x(t) formula (I)—H(z)x(t); providing to an electro-acoustical transducer a signal corresponding to the filtered audio signal x(t) formula (I) to produce a sound wave corresponding to the filtered audio signal; and prior to filtering the audio signal using the filter, configuring the filter such that, with respect to one or more frequencies, the filtered audio signal has a higher signal level than the input audio signal, and such that the overall signal level of the filtered audio signal (slƒ) is substantially related to the overall signal level of the input signal (slr) such that si/=sl/×c.

Abstract: An apparatus for a portable electronic device for receiving a jack of a headset, the jack including a set of lines, the set of lines including at least one audio line, a ground signal and a microphone signal line, the apparatus comprising a set of switches for receiving the ground signal line and the microphone signal line and a sensing circuit for reducing induced noise from the headset, wherein the sensing circuit is located between the set of switches and the microphone signal line and ground signal line.

Abstract: A system is provided to protect a loudspeaker (144) by controlling a level of an applied audio signal. A control signal is generated by applying an input audio signal (115) to the collective operations of a gain control system (100). The gain control system (100) uses the input audio signal (115) in conjunction with at least one parameter to derive an estimated stress associated with the loudspeaker (144). The estimated stress is compared with a protection threshold stress (127). If the protection threshold stress is exceeded, a gain applied by a gain component (134) is selectively adjusted to modify the input audio signal (115). The resulting gain-controlled audio signal (116) is employed to drive the loudspeaker (144).

Abstract: Zoom motor noise in a camera audio recording is reduced by detecting activity of the zoom motor, transforming a audio signal into the frequency domain during zoom motor activity, and scaling the frequency domain signal during zoom motor activity in each of a series of frequency bins by a scaling factor derived from a pre-stored zoom motor noise spectrum to produce a processed audio signal in the frequency domain. The processed audio signal is then transformed back to the time domain.

Abstract: An audio apparatus is provided. The audio apparatus includes at most one electroacoustic transducer; and an audio controller, coupled to the electroacoustic transducer, for actively controlling the electroacoustic transducer to function as a loudspeaker or a microphone, wherein the loudspeaker converts output electrical signals to output sounds, and the microphone converts input sounds to input electrical signals.

Abstract: Interference in an audio signal is reduced by estimating a target signal using beam-forming in a direction of the signal source. A set of estimates of interference is determined by using a microphone array filtering matrix to block the target signal in the audio signal. A set of filters is optimized by minimizing an objective function measuring a mismatch between the set of estimates of interference and the estimate of the target signal. The minimizing uses a sparse regularization of coefficients of the set of filters. The set of estimates of interference are filtered using the set of filters after the optimizing. Then, the estimate of interference after the optimizing is subtracted from the target signal.

Abstract: An active vibration/noise control device which is provided with a plurality of cancel signal generation parts for generating output signals for respectively cancelling noises generated at a plurality of vibration/noise generation sources. The effect of the suspension of either of first and second cancel signal generation parts on the other is reduced. According to the operating state of the first cancel signal generation part, the simulated transmission properties of the second cancel signal generation part are adjusted. Consequently, without regard to the operating state of the first cancel signal generation part, the noise control performance of the second cancel signal generation part can be maintained.

Abstract: A receiving device for mobile reception of a receive signal including an audio signal associated with a currently received piece of music, and a method for playback in a mobile receiver are disclosed. A receiving circuit receives the receive signal and outputs the audio signal and information associated with the audio signal. The information includes one or more information fields associated with the currently received piece of music. An arithmetic unit evaluates the information associated with the audio signal. A database is configured for storing database entries. Each database entry includes data fields and associated audio data. The arithmetic unit assigns the information to an assigned database entry by comparing the information fields with the data fields. The arithmetic unit outputs the audio data of the assigned database entry based on a detected disturbance in the reception of the audio signal.

Abstract: A device and method are provided that reduce interference between a wireless communication device and a speaker. Generally speaking, a microphone input is monitored for detecting noise created by the interference. If noise is detected, a power transmission level of the wireless device is reduced from a standard power transmission level.

Abstract: Disclosed herein are system, method and apparatus with environmental noise cancellation. The instant disclosure is particularly adapted to a receiver module having at least two inputs. The two inputs respectively receive a main audio portion and the audio with majority of environmental noise. The system firstly calibrates the audio signals to reduce the error caused by the difference between the two inputs. An adaptive beamforming technology and a speech extractor are respectively used to extract the environmental noise portion with less main audio and the main audio portion with less noise. After a process of time-to-frequency domain transformation, a non-linear noise suppression technology is introduced into estimating the environmental noise and acquiring a gain. After noise suppression processed with the gain, a sequence of audio signals is output after a frequency-to-time domain transformation.

Abstract: A speech signal processing device is equipped with a power acquisition unit, a probability distribution acquisition unit, and a correspondence degree determination unit. The power acquisition unit accepts an inputted speech signal and, based on the accepted speech signal, acquires power representing the intensity of a speech sound represented by the speech signal. The probability distribution acquisition unit acquires a probability distribution using the intensity of the power acquired by the power acquisition unit as a random variable. The correspondence degree determination unit determines whether a correspondence degree representing a degree that power acquired by the power acquisition unit in a case that a predetermined reference speech signal is inputted into the power acquisition unit corresponds with predetermined reference power is higher than a predetermined reference correspondence degree, based on the probability distribution acquired by the probability distribution acquisition unit.

Abstract: Systems and methods are described by which microphones comprising a mechanical filter can be accurately calibrated to each other in both amplitude and phase.

Abstract: A voice input device includes a first microphone (710-1) that includes a first diaphragm, a second microphone (710-2) that includes a second diaphragm, and a differential signal generation section (720) that generates a differential signal that indicates a difference between a first voltage signal and a second voltage signal, the first diaphragm and the second diaphragm being disposed so that a noise intensity ratio is smaller than an input voice intensity ratio (input voice component intensity ratio), and the differential signal generation section (720) including a gain section (760) that amplifies the first voltage signal by a predetermined gain, and a differential signal output section (740) that generates and outputs a differential signal that indicates a difference between the first voltage signal amplified by the gain section and the second voltage signal.

Abstract: A system for noise removal is coupled to a signal unit that provides a digital signal. The noise removal system includes a transformation module to transform the digital signal into an f-digital signal, a threshold filter to generate a noiseless signal from the f-digital signal based on a threshold profile, and a signal synthesizer to provide a gain to the noiseless signal and to transform the noiseless signal into an output signal.

Abstract: There is provided a sound collecting device, including: an orientation direction forming section that forms an orientation direction of a microphone array; and a control section that, when a characteristic in a frequency band of a synthesized signal obtained by synthesizing the acoustic signals corresponds to a characteristic of an acoustic signal corresponding to a sound other than a target sound, controls the orientation direction forming section such that an orientation direction that is a direction that is different than an orientation direction of the microphone array at a present point in time is formed, and, when the characteristic in the frequency band of the synthesized signal does not correspond to a characteristic of an acoustic signal corresponding to a sound other than the target sound, controls the orientation direction forming section such that the orientation direction of the microphone array is maintained.

Abstract: A correction spectrum calculation unit 6 obtains a correction spectrum by smoothing an estimated noise spectrum in accordance with the degree of its variations, and a suppression quantity limiting coefficient calculation unit 7 decides a suppression quantity limiting coefficient from the correction spectrum. A suppression quantity calculation unit 9 obtains a suppression coefficient based on the suppression quantity limiting coefficient, and the spectrum suppression unit 10 carries out amplitude suppression of spectral components of an input signal.

Abstract: A headset assembly with a recording function for communication includes a left earphone, a right earphone, a call module, and a recording module. The left and the right earphones respectively have a speaker and a microphone. The call module is electrically connected to the left or the right earphone. The recording module is electrically connected to the left and the right earphones. In a first operation mode, the call module communicates with an external communication device through the microphone and the speaker of the left or the right earphone. In a second operation mode, the recording module receives an ambient sound signal through the microphones simultaneously, and records and stores the ambient sound signal. As the microphones are placed in a left auricle and a right auricle, recorded sound may have an effect of a head-related transfer function (HRTF), thus achieving an effect of a stereo sound field during playback.

Abstract: Electronic devices are provided that communicate over cables and other communications paths that include optical and electrical paths. A cable may include wires for forming an electrical path and one or more optical fibers for forming an optical path. Connectors at one or both ends of the cable may include electrical contacts and an optical coupling structure associated with the optical path. Optical paths may be included in connectors such as tip-ring-sleeve connectors and connectors of other types. Interface circuitry may be included in a connector to convert between optical and electrical signaling schemes. Wavelength-division-multiplexing may be used to support bidirectional communications. Breakout boxes and other equipment may be connected using the cables. Digital signals such as digital noise cancellation signals may be conveyed over the optical paths. Power and other electrical signals may be conveyed over the electrical paths.

Abstract: An information handling system has a haptic generation module to generate haptic effects including haptic noise and a haptic noise reduction module. The haptic noise reduction module receives characteristics of sound representative of haptic noise generated by a haptic generation module of a device and entering an audio input module of the device, the characteristics including frequencies and timing. It also detects the generation of haptic effects, the generations occurring after the receiving characteristics. It also reduces the effects of haptic noise on digital data representing audio input to the device based upon the received characteristics of the sound. It may reduce the effects by subtracting amplitudes of audio waves representing the haptic noise from amplitudes of audio waves representing the audio input.

Abstract: An apparatus for extracting a signal from convolutive mixtures includes a receiving unit which includes two or more receivers and receives a signal; a transfer function calculation unit which calculates transfer functions for demixing; and a demixing unit which demixes the received signal using the calculated transfer functions. The transfer function is determined such that a signal is extracted from a source closest to the receivers, and is calculated on the basis of a transfer function for a path to each receiver being approximated to a delta function as closer to the source.

Abstract: A method and apparatus for providing simultaneous enhancement of transmission loss and absorption coefficient using activated cavities is presented. A layer of material is provided, and a backing plate having a plurality of cavities on the top surface of said backing plate, is disposed adjacent a top surface of said layer of material. A screen is disposed along the top surface of said cavities on said backing plate and at least one cavity includes an actuator disposed within the cavity and a control system receiving a signal from the microphone and receiving a signal from the accelerometer and providing a drive signal to the actuator to provide an acoustic output to provide simultaneous insertion loss and absorption which serves to minimize a linear combination of the signal from the microphone and the signal from the accelerometer.

Abstract: The present invention provides a spatially pre-processed target-to-jammer ratio weighted filter and a method thereof, which uses two microphones to receive audio signals. The audio signals are divided into a plurality of sinusoidal waves by a fast Fourier transform (FFT) module, and a beamformer uses the sinusoidal waves to generate beamformed signals. A reference generator generates at least one reference signal. The beamformed signals and reference signals are used to work out power spectral densities (PSD), and a target-to-jammer ratio (TJR) is worked out with the power spectral densities. TJR is used to determine whether a sound source exists. According to the determination result, a noise estimator is switched to eliminate noise from the beamformed signals and generate output signals. An inverse fast Fourier transform (IFFT) module recombines the output signals and then outputs the recombined signals.

Abstract: Provided is a method and apparatus for encoding/decoding an audio signal. Sections which are not used to output noise components near important spectral components and sub-bands which are not used to output noise components, are determined to be encoded or decoded, so that the efficiency of encoding and decoding an audio signal increases, and sound quality can be improved using less bits.

Abstract: A noise reduction circuit for reducing the effects of low frequency noise such as wind noise in communications applications is described. In one embodiment, the noise reduction circuit features a high pass filter formed by exploiting the existing off-chip AC coupling capacitances in making the connection to the source of audio signals. The filter may be adaptive to environmental low frequency noise level through programming the shunt resistances. A low-noise wide dynamic range programmable gain amplifier is also described. Adaptive equalization of the audio signal is also described through the utilization of programmable front-end resistors and a back-end audio equalizer.

Abstract: A synthesis filter 106 synthesizes a plurality of wide-band speech signals by combining wide-band phoneme signals and sound source signals from a speech signal code book 105, and a distortion evaluation unit 107 selects one of the wide-band speech signals with a minimum waveform distortion with respect to an up-sampled narrow-band speech signal output from a sampling conversion unit 101. A first bandpass filter 103 extracts a frequency component outside a narrow-band of the wide-band speech signal and a band synthesis unit 104 combines it with the up-sampled narrow-band speech signal.

Abstract: The invention provides a method for determining a signal component for reducing noise in an input signal, which comprises a noise component, comprising the steps of: estimating the noise component in the input signal, estimating a reverberation component in the noise component, and removing the estimated reverberation component from the estimated noise component to obtain a modified estimate of the noise component.

Abstract: Systems and methods are described by which microphones comprising a mechanical filter can be accurately calibrated to each other in both amplitude and phase.

Abstract: A method and system to reduce the noise floor of a communications system is disclosed. The system may be incorporated into any device that provides binary samples from a datastream, such as a cordless telephone system. The system is configured to determine a number of bits of the binary samples that are affected by noise. The system is then able to remove the noise by setting those bits to a fixed value. The fixed value may depend on whether the sample is positive or negative. The value to set may be chosen so that the least significant bits of each sample come as close as possible to 0 for that particular numerical representation system. The system can be integrated with other known signal processing methods.

Abstract: Systems and methods of limiting peak audio power in mobile devices may include a high pass filter and a burst module to detect a burst load condition in a mobile device. The burst module can also apply the high pass filter to an audio signal of the mobile device in response to the burst load condition to obtain a filtered signal, and transmit the filtered audio signal to a speaker of the mobile device.

Abstract: Noise suppression is performed based on null coherence between sub-band signals of a primary acoustic signal and a secondary acoustic signal. The null coherence of a signal refers to portions of the signal that have high coherence and can be nullified by a null processor. The nullified component corresponds to target sources, such as an individual speaking into a phone. The coherence values indicate the presence of a target source and are used to suppress noise in portions of a signal that are not dominated by a desired target source. The inter-microphone level difference may be used in combination with the null coherence to provide noise suppression.

Abstract: A method and a device for suppressing noise in the passenger compartment of a vehicle, which include at least one transducer, a programmable computer, at least one acoustic sensor, the computer being configured such as to apply an electro-acoustic model of the passenger compartment to a correcting system model including a central controller with fixed coefficients joined to a block of variable coefficients, including a Youla parameter in the form of a Youla block Q. The first phase includes determining and calculating the electro-acoustic model and the control law for at least one predetermined noise frequency. In a second phase, in real time, the computer applies the control law to the electro-acoustic model in accordance with the current frequency of the noise to be suppressed.

Abstract: A noise cancellation system for an audio system such as a mobile phone handset, or a wireless phone headset has a first input for receiving a first audio signal from one or more microphone positioned to receive ambient noise, and a second input for receiving a second audio signal from a microphone positioned to detect the user's speech, as well as a third input for receiving a third audio signal for example representing the speech of a person to whom the user is talking. A first noise cancellation block receives the first audio signal and generates a first noise cancellation signal, and this is combined with the third audio signal to form a first audio output signal. A second noise cancellation block receives at least a part of the first audio signal and said second audio signal and applying noise cancellation to generate a second audio output signal.

Abstract: Microphone arrays (MAs) are described that position and vent microphones so that performance of a noise suppression system coupled to the microphone array is enhanced. The MA includes at least two physical microphones to receive acoustic signals. The physical microphones make use of a common rear vent (actual or virtual) that samples a common pressure source. The MA includes a physical directional microphone configuration and a virtual directional microphone configuration. By making the input to the rear vents of the microphones (actual or virtual) as similar as possible, the real-world filter to be modeled becomes much simpler to model using an adaptive filter.