Abstract: A method, medium, and apparatus for extracting a target sound from mixed sound. The method includes receiving a mixed signal through a microphone array, generating a first signal whose directivity is emphasized toward a target sound source and a second signal whose directivity toward the target sound source is suppressed based on the mixed signal, and extracting a target sound signal from the first signal by masking an interference sound signal, which is contained in the first signal, based on a ratio of the first signal to the second signal. Therefore, a target sound signal can be clearly separated from a mixed sound signal which contains a plurality of sound signals and is input to a microphone array.

Abstract: Techniques are described herein that use sensors (e.g., microphones) for noise reduction in a mobile communication device. For example, one technique enables a first sensor that is initially configured to be a speech sensor to be used as a noise reference sensor. This technique also enables a second sensor that is initially configured to be a noise reference sensor to be used as a speech sensor. Another technique enables a primary sensor and/or a secondary sensor in a handset of a mobile communication device to be used as a speech sensor while a sensor in a headset of the mobile communication device is used as a noise reference sensor, or vice versa. In yet another technique, a secondary sensor in a mobile communication device is configured to be a directional sensor.

Abstract: A data communication apparatus includes a band-elimination filter configured to perform a process of reducing a specific frequency component included in transmission data, a packet producer configured to produce packet data including therein data outputted from the band-elimination filter, and an antenna configured to output the packet data produced by the packet producer, and further, the band-elimination filter is configured to reduce a frequency component corresponding to the reciprocal of a packet transmission cycle inherent in the packet data.

Abstract: This specification describes technologies relating to editing digital audio data. In general, one aspect of the subject matter described in this specification can be embodied in methods that include the actions of receiving an audio signal including audio data in multiple channels; identifying noise in the audio signal including identifying panning information for the audio data in the signal at each of multiple frequency bands; and attenuating the audio data at one or more frequency bands to generate an edited audio signal when the panning exceeds a specified threshold for each of the one or more frequency bands. Other embodiments of this aspect include corresponding systems, apparatus, and computer program products.

Abstract: A method and system for masking pure tones within sound generated from a noise generating source. The method includes detecting one or more pure tones within sound being generated from the noise generating source, and generating one or more masking sounds capable of masking only the one or more pure tones detected within the sound.

Abstract: An amplification circuit for driving an audio signal diffuser that includes a generation circuit of a first pre-charging signal, the generation circuit including an amplifier provided with an input terminal for receiving the first pre-charging signal and provided with an output terminal for providing a second pre-charging signal as a function of the first pre-charging signal, and a decoupling capacitor of the amplifier from the diffuser, the capacitor connected to the output terminal for charging by the second pre-charging signal.

Abstract: A system for enhancing the sound signal produced by an audio system in a listening environment by compensating for ambient noise in the listening environment is provided. The system receives an electrical sound signal and generates a sound output therefrom. A total sound signal is sensed representative of the total sound level in the environment, where the total sound level includes both the sound output from the audio system and the ambient noise within the environment. The system extracts an ambient noise signal representative of the ambient noise in the environment from the total sound signal in response to the total sound signal and to a reference signal derived from the electrical sound signal. The system extracts the ambient noise signal using an adaptive filter with an adaptive step size. The system generates a control signal in response to the ambient noise signal and adjusts the sound output of the audio system to compensate for the ambient noise level in response to the control signal.

Abstract: An audio communications device has a handset in which a touch sensing ear piece region is coupled to an acoustic leakage analyzer. The acoustic leakage analyzer is to analyze signals from the touch sensing ear piece region and on that basis adjust an audio processing parameter. The latter configures an audio processor which generates an audio receiver input signal for the device. Other embodiments are also described and claimed.

Abstract: A first differential value is acquired between first current data and first previous data in an i number (i being a natural number) of sampling periods before the current data. A second differential value is acquired between second current data and second previous data in a j number (j being a natural number) of sampling periods before the current data. Both first data and both second data are of a first and a second digital audio signal, respectively, having a sound level of a digital stereo audio signal in the left and right channels, respectively. A first and a second correction coefficient are acquired by adding the first and second differential values at a first and a second ratio, respectively. The first signal is corrected by multiplying the first signal by the first correction coefficient. The second signal is corrected by multiplying the second signal by the second correction coefficient.

Abstract: A surge protection circuit acquires a surge signal from a left channel (LC) signal line and a right channel (RC) signal line. After the surge signal being transmitted on the LC signal line and the RC signal line is removed, an audio signal outputted from a signal input device is transmitted to an audio output device via the LC signal line and the RC signal line.

Abstract: An audio signal is received that might include keyboard noise and speech. The audio signal is digitized and transformed from a time domain to a frequency domain. The transformed audio is analyzed to determine whether there is likelihood that keystroke noise is present. If it is determined there is high likelihood that the audio signal contains keystroke noise, a determination is made as to whether a keyboard event occurred around the time of the likely keystroke noise. If it is determined that a keyboard event occurred around the time of the likely keystroke noise, a determination is made as to whether speech is present in the audio signal around the time of the likely keystroke noise. If no speech is present, the keystroke noise is suppressed in the audio signal. If speech is detected in the audio signal or if the keystroke noise abates, the suppression gain is removed from the audio signal.

Abstract: Methods and apparatuses for detection and reduction of wind noise in audio devices are disclosed. In an embodiment, a method includes acquiring and transforming the audio signals. Correlations from the transformed audio signals are computed. A cross correlation index is compared to a predetermined value to determine if a wind noise spectral content is present. In another embodiment, an apparatus includes an audio processing unit to receive non-decomposed audio signals, and an audio decomposition unit to receive the non-decomposed audio signals and to generate decomposed audio signals. A wind noise spectrum estimation unit receives non-decomposed audio signals and decomposed audio signals and identifies wind noise spectral components in at least one of the non-decomposed and decomposed audio signals. A wind noise spectrum reduction unit receives the wind noise spectral components and removes the wind noise spectral components from at least one of the non-decomposed and the decomposed audio signals.

Abstract: A communications system, e.g., a wireless microphone, incorporates a quadrature modulator system to reduce power consumption with respect to traditional approaches and is general in nature to support any two-dimensional digital technique. The quadrature modulator system comprises different subsystems, including a digital-analog transformation circuit, a baseband filter, and a quadrature modulator. The digital-analog transformation circuit converts discrete time samples to a continuous time signal, and further includes an oversampling noise-shaping modulator such as a sigma-delta modulator. The baseband filter then removes out-of-band energy including sampling images and quantization noise. Some of the circuit components may comprise discrete devices that may result in a reduction of power consumption for the quadrature modulator system. Alternatively, some or all of the circuit components may be incorporated in a single electronic device.

Abstract: A signal processing device includes a sound pickup unit configured to pick up sound and convert the sound into a sound signal; a signal recording unit including an actuator that is driven by a current supplied from a power source and that has a possibility of generating audible noise caused by the driving, the signal recording unit being configured to record the sound signal therein; a current detecting unit configured to detect the magnitude of the current and a temporal change in the current; and a signal processing unit configured to determine, on the basis of at least one of the magnitude of the current and the temporal change in the current, whether the audible noise has been generated and perform, when it is determined that the audible noise has been generated, noise removal processing for removing the audible noise from the sound signal, which contains the audible noise.

Abstract: A method of reducing noise in an environment where the noise source is in a fixed location relative to a pair of microphones, such as in a camera with a zoom motor, involves receiving signals x1(t), x2(t) from the respective microphones, and filtering each of the signals x1(t), x2(t) with respective first and second linear filters having filter coefficients obtained by computing eigenfilters corresponding to data samples from the respective microphones for noise only and signal only conditions.

Abstract: According to one embodiment, a compensation filtering device includes an impulse response calculator, a group delay compensator, and an extractor. The impulse response calculator calculates an impulse response of a reproduction system comprising a sound field. The group delay compensator compensates for group delay characteristics in a low frequency range lower than a predetermined frequency for a finite impulse response (FIR) filter having reverse characteristics of the impulse response based on group delay characteristics in a middle to high frequency range higher than the predetermined frequency. The extractor extracts a predetermined number of taps from the FIR filter that has been compensated for by the group delay compensator.

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

Abstract: An active noise reduction system using adaptive filters. A method of operation the active noise reduction system includes smoothing a stream of leakage factors. The frequency of a noise reduction signal may be related to the engine speed of an engine associated with the system within which the active noise reduction system is operated. The engine speed signal may be a high latency signal and may be obtained by the active noise reduction system over audio entertainment circuitry.

Abstract: Incoming audio from mobile devices can be centrally processed, where a server can filter background noise in real time, such as by using an XOR function. Instead of discarding the filtered noise, however, it can be processed in parallel to dynamically construct an acoustic map of the environment. The acoustic map can be generated from an aggregation of sound data from multiple devices positioned in a geographic environment. The acoustic map can be linked to a configurable set of rules, conditions, and events, which can cause dynamic adjustments to be made to a workforce task management system. For example, employee availability can be assessed using the acoustic map and workforce tasks can be assigned based in part upon this availability.

Abstract: The present invention discloses a microphone array structure able to reduce noise and improve speech quality and a method thereof. The method of the present invention comprises steps: using at least two microphone to receive at least two microphone signals each containing a noise signal and a speech signal; using FFT modules to transform the microphone signals into frequency-domain signals; calculating an included angle between a speech signal and a noise signal of the microphone signal, and selecting a phase difference estimation algorithm, a noise reduction algorithm or both to reduce noise according to the included angle; if the phase difference estimation algorithm is used, calculating phase difference of the microphone signals to obtain a time-space domain mask signal; and multiplying the mask signal and the average of the microphone signals to obtain the speech signals of the microphone signals. Thereby is eliminated noise and improve speech quality.

Abstract: Method, user terminal and computer program product for controlling audio signals at the user device during a communication session between the user device and a remote node, in which a primary audio signal is received at audio input means of the user device for transmission to the remote node in the communication session. It is determined whether the user device is operating in (i) a first mode in which secondary audio signals output from the user device are likely to disturb the primary audio signal received at the audio input means, or (ii) a second mode in which secondary audio signals output from the user device are not likely to disturb the primary audio signal received at the audio input means.

Abstract: Architecture that enables wireless narrowband devices (e.g., wireless microphones) and white space devices to efficiently coexist on the same telecommunications channels, while not interfering with the usability of the wireless narrowband device. The architecture provides interference detection, strobe generation and detection and, power ramping and suppression (interference-free coexistence with spectrum efficiency). The architecture provides the ability of the white space device to learn about the presence of the microphone. This can be accomplished i using a geolocation database, reactively via a strober device, and/or proactively via the strober device. The strober device can be positioned close to the microphone receiver and signals the presence of a microphone to white space devices on demand.

Abstract: Disclosed herein are conferencing products implementing an acoustic echo cancellation system that utilizes converging coefficients and a detector of turn-off and/or turn-on events of push-to-talk microphones, and, further, that mitigates against divergence and/or drift of coefficients and other variables of an echo canceller. A push-to-talk detector may be used that includes a high-pass filter, a transient detector, or an adjustable high-pass filter. An echo canceller may be disabled as to a push-to-talk microphone that has been turned off.

Abstract: An apparatus processes an acoustic input signal to provide an output signal with reduced noise. The apparatus weights the input signal based on a frequency-dependent weighting function. A frequency-dependent threshold function bounds the weighting function from below.

Abstract: A pop noise suppression apparatus for eliminating popping noise generated upon initiation or shutdown of an audio output circuit comprises a switch component and a control circuit. The switch component allows the audio output circuit to provide audio through the output of the audio output circuit. The control circuit provides a mute signal for a first period of time in to response initiation or shutdown of the audio circuit. The control circuit comprises a capacitor to be charged upon initiation of the audio output circuit or to be discharged upon shutdown of the audio output circuit. A length of the first period of time during which the mute signal is provided depends on a second period of time to charge or discharge the capacitor.

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 sound monitoring system and method. The system can include a plurality of sound pressure level meters, a plurality of sound level indicators and a server connected by a network. The sound pressure level meters measuring a sound level at their location, and the sound level indicators providing a visual indication of the sound level measured by at least one of the sound pressure level meters. The system devices can be powered, as well as monitored and controlled remotely over the network. A user interface enables constructing and monitoring multiple zones and groups in a monitored area, as well as reviewing real-time and historical sound data. The system can also control lighting in the monitored area, and use lighting as a visual indicator of noise level.

Abstract: A “speaker” operating mode is established by a signal processor of a hearing aid for tracking and selecting an acoustic speaker source in an ambient sound. Electric acoustic signals are generated by the hearing aid from the ambient sound that has been picked up, from which signals an electric speaker signal is selected by the signal processor by a database of speech profiles of preferred speakers. The electric speech signal is selectively taken into account in an output sound of the hearing aid in such a way that it will for the hearing-aid wearer acoustically at least be prominent compared with another acoustic source and consequently be better perceived by the hearing-aid wearer.

Abstract: 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.

Abstract: A system and method for providing call privacy for a wireless communication device. A voice communication is received from a user. The voice communication is processed to determine a response signal in response to receiving the voice signal. The response signal is broadcast about the periphery of the user as the voice input is received. The response signal is operative to cause the voice communication to be less discernible by one or more bystanders.

Abstract: An active noise reduction system using adaptive filters. A method of operation the active noise reduction system includes smoothing a stream of leakage factors. The frequency of a noise reduction signal may be related to the engine speed of an engine associated with the system within which the active noise reduction system is operated. The engine speed signal may be a high latency signal and may be obtained by the active noise reduction system over audio entertainment circuitry.

Abstract: Systems and methods for utilizing inter-microphone level differences (ILD) to attenuate noise and enhance speech are provided. In exemplary embodiments, primary and secondary acoustic signals are received by omni-directional microphones, and converted into primary and secondary electric signals. A differential microphone array module processes the electric signals to determine a cardioid primary signal and a cardioid secondary signal. The cardioid signals are filtered through a frequency analysis module which takes the signals and mimics a cochlea implementation (i.e., cochlear domain). Energy levels of the signals are then computed, and the results are processed by an ILD module using a non-linear combination to obtain the ILD. In exemplary embodiments, the non-linear combination comprises dividing the energy level associated with the primary microphone by the energy level associated with the secondary microphone.

Abstract: To reliably and consistently detect desirable sounds, a system detects the presence of wind noise based on the power levels of audio signals. A first transducer detects sound originating from a first direction and a second transducer detects sound originating from a second direction. The power levels of the sound are compared. When the power level of the sound received from the second transducer is less than the power level of the sound received from the first transducer by a predetermined value, wind noise may be present. A signal processor may generate an output from one or a combination of the audio signals, based on a wind noise detection.

Abstract: Systems and methods for providing single microphone noise suppression fallback are provided. In exemplary embodiments, primary and secondary acoustic signals are received. A single microphone noise estimate may be generated based on the primary acoustic signal, while a dual microphone noise estimate may be generated based on the primary and secondary acoustic signals. A combined noise estimate based on the single and dual microphone noise estimates is then determined. Using the combined noise estimate, a gain mask may be generated and applied to the primary acoustic signal to generate a noise suppressed signal. Subsequently, the noise suppressed signal may be output.

Abstract: An audio processing apparatus generates a suppression coefficient sequence that is composed of coefficient values corresponding to frequency components of an audio signal, the frequency components being multiplied by the corresponding coefficient values to suppress noise components of the audio signal. In the audio processing apparatus, a characteristic value calculation unit calculates a noise characteristic value depending on a shape of a magnitude distribution of the audio signal. An intensity setting unit variably sets a suppression intensity of the noise components based on the noise characteristic value. A coefficient sequence generation unit generates the suppression coefficient sequence based on the audio signal and the suppression intensity.

Abstract: A system reduces noise or other external signals that may affect communication. A device converts sound from two or more microphones into an operational signal. Based on one or both signals, a beamformer generates an intermediate signal. Reflected or other undesired signals may be estimated or measured by an echo canceller. Interference may be measured or estimated by processing the echo-reduced signal or estimate by a blocking matrix. An interference canceller may reduce the interference that may modify or disrupt a signal based on the output of the blocking matrix and the intermediate signal.

Abstract: An audio device includes a first audio path with a loudspeaker for reproducing an audio signal, and a second audio path. The second audio path includes in series a band-pass filter for filtering an audio signal, a detector for detecting the amplitude of the band-pass filtered audio signal, a multiplier for multiplying a periodic signal by the amplitude of the band-pass filtered audio signal, and a vibration device for reproducing the multiplied periodic signal. The frequency of the periodic signal is substantially equal to the resonance frequency of the vibration device.

Abstract: A sound collection apparatus includes a target sound collection unit that collects a sound including a target sound and outputs a collected-sound signal, a non-target sound collection unit, provided at positions different from each other, forms dead zones of sensitivity in a direction of the target sound source so as to collect a sound outside the dead zones and outputs a collected-sound signal. A sensitivity suppression unit generates a sensitivity suppression signal for suppressing a sound collection sensitivity in an overlap region in which dead zones overlap, as compared to a region surrounding the overlap region, by subjecting, to a predetermined signal processing, the collected-sound signal outputted by the non-target sound collection unit. An extraction unit removes, from the collected-sound signal, the sensitivity suppression signal generated, so as to extract a signal of a sound generated in the overlap region in which the dead zones overlap.

Abstract: In an audio-accompanying moving-image taking apparatus, a noise period setting unit sets a second period with respect to an audio signal acquired by image-taking. The second period is a period between the endpoint of a first period which is a predetermined period starting from the timing when a drive instruction is made to a drive unit for driving a lens and the point in time when the drive unit stops driving according to the drive instruction. A noise level estimation unit estimates a noise level using the signal present in the second period set by the noise period setting unit. A noise suppression unit suppresses noise from the signal present in the second period using the noise level estimated by the noise estimation unit.

Abstract: A howling canceller applied to an acoustic system having a speaker and a microphone comprises: a filter insertion unit for inserting a notch filter at a frequency of an audio signal picked up by the microphone; a setting unit for setting the insertion time of the notch filter on the basis of the frequency at which the notch filter is inserted; and a releasing unit for, when the insertion time set by the setting unit has elapsed, releasing the notch filter, the insertion time of which has elapsed. The setting unit sets the insertion time of the notch filter to be shorter as the frequency at which the notch filter is inserted increases.

Abstract: A system and method facilitating signal enhancement utilizing an adaptive filter is provided. The invention includes an adaptive filter that filters an input based upon a plurality of adaptive coefficients, the adaptive filter modifying at least one of the adaptive coefficients based on a feedback output. The invention further includes a feedback component that provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output. The invention further provides a noise statistics component that stores noise statistics associated with a noise portion of an input signal and a signal+noise statistics component that stores signal+noise statistics associated with a signal and noise portion of the input signal.

Abstract: The present invention relates to an audio amplification circuit comprising a first preamplifier for receipt of an audio input signal and a second preamplifier comprising a first differential input for receipt of an attenuated audio input signal. The attenuated audio input signal is generated by an attenuator coupled to the audio input signal. A non-linear element is coupled to a first input of the first preamplifier thereby distorting the audio input signal at the first input at large signal levels. A distortion compensation network is adapted to supply a distortion compensation signal from the first input of the first preamplifier to a second differential input of the second preamplifier such that distortion in the output signal of the second preamplifier is cancelled or attenuated. The invention further relates to a corresponding method of compensating an audio amplification circuit for distortion induced by a non-linear element.

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract: A signal processor uses input devices to detect speech or aural signals. Through a programmable set of weights and/or time delays (or phasing) the output of the input devices may be processed to yield a combined signal. The noise contributions of some or each of the outputs of the input devices may be estimated by a circuit element or a controller that processes the outputs of the respective input devices to yield power densities. A short-term measure or estimate of the noise contribution of the respective outputs of the input devices may be obtained by processing the power densities of some or each of the outputs of the respective input devices. Based on the short-term measure or estimate, the noise contribution of the combined signal may be estimated to enhance the combined signal when processed further. An enhancement device or post-filter may reduce noise more effectively and yield robust speech based on the estimated noise contribution of the combined signal.

Abstract: A system improves speech detection or processing by identifying registration signals. The system encodes a limited frequency band by varying the amplitude of a pulse width modulated signal between predefined values. The signal is separated into frequency bins that identify amplitude and phase. The registration signal is measured by comparing a difference in average acoustic power in a plurality of adjacent bins over time.

Abstract: A howling suppressing apparatus includes: a detecting unit configured to detect howling of input audio signals; a plurality of filters configured to apply a filter process sequentially to the audio signals to be output; and a setting unit configured to set a filter coefficient for suppressing the howling detected by the detecting unit for a filter among the plurality of filters, in which filter no filter coefficient for suppressing howling is set, and set a filter coefficient for suppressing the howling detected by the detecting unit for any one of the plurality of filters, if filter coefficients for suppressing howling are set in all of the plurality of filters, based on the detection result from the detecting unit.

Abstract: A noise eliminating apparatus includes a first microphone, a second microphone and a signal processing unit, the signal processing unit includes a linear prediction filter and a noise resynthesis filter, the linear prediction filter receives an output signal of the first microphone, predicts the output signal of the first microphone by linear prediction and generates a prediction signal, and the noise resynthesis filter is an adaptive filter which receives, as a main input signal, a first difference signal obtained by subtracting one of the output signal of the first microphone and the prediction signal from the other, receives, as an error signal, a second difference signal obtained by subtracting one of an output signal of the second microphone and an output signal of the noise resynthesis filter itself from the other, and updates a filter coefficient so that the error signal is minimized.

Abstract: A system and method control the level of applied noise reduction, the level of noise reduction being influenced by monitoring audio signals, analyzing audio signal components, bandwidth of background noise, user preferences and other factors.

Abstract: The present invention relates to a suppressing noise system applied in a mobile device, comprises: at least two microphones are used for respectively transmitting a first audio signal with noise and a second audio signal with noise; a pre-processing unit is coupled to the at least two microphones for oversampling the first and second audio signals, downsampling the sampled first and second audio signals, and then generating a first adjusted signal and a second adjusted signal; and a suppressing noise device is coupled to the pre-processing device for filtering noise in the first and second adjusted signals once again, and outputting a third audio signal.

Abstract: A howling canceller is adapted to an acoustic system having a speaker and first and second microphones. The speaker and the first microphone form a first acoustic feedback loop; the speaker and the second microphone form a second acoustic feedback loop. The howling canceller includes a howling suppressing unit for performing suppression processing in such a way that: frequency components at which howling is possibly occurring are detected in each of the sound signals picked up by the first and second microphones; the detected frequency components of the sound signals picked up by the first and second microphones are compared with each other on a per-frequency basis and a frequency component having larger power is detected; and based on the comparison results, the larger power frequency component of at least one of the sound signals picked up by the first and second microphones is suppressed.