Abstract: The directional effect of a static directional microphone is to be improved. In particular shadowing effects of the head for a hearing device worn on the head of the user are to be taken into account when adjusting at the directional microphone. To this end it is proposed that—like the adaptation of an adaptive directional microphone—the energy or power of the directional microphone signal emitted by the directional microphone is minimized, with the difference that in this case extremely long adaptation times are predetermined.

Abstract: A noise reduction system includes multiple transducers that generate time domain signals. A transforming device transforms the time domain signals into frequency domain signals. A signal mixing device mixes the frequency domain signals according to a mixing ratio. Frequency domain signals are rotated in phase to generate phase rotated signals. A post-processing device attenuates portions of the output based on coherence levels of the signals.

Abstract: A microphone for use in windy environments. The microphone uses a transducer to flood the environment outside the microphone with a high frequency (such as an ultrasonic) acoustic field. The sounds desired to be detected are mixed with the high frequency carrier, and can then be received by the microphone with less wind noise. The microphone then demodulates the desired sound signals from the high frequency carrier. The microphone can be configured in a special emitter-receiver configuration that also reduces interference from engine noise.

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 apparatus has a function of correcting an audio signal in response to a noise level. The audio apparatus includes a correction unit that corrects an input audio signal on the basis of a weighting factor, an output unit that produces a played-back audio sound on the basis of the corrected audio signal, a microphone for receiving an external sound that includes the played-back audio sound and noise, a noise-extracting unit that extracts a noise signal from an external sound signal, the noise-extracting unit including a speech-removing unit that removes a speech signal from the noise signal on the basis of noise spectrum data, and a weighting factor calculation unit that calculates the weighting factor on the basis of the extracted noise signal and supplies the calculated weighting factor to the correction unit.

Abstract: A method for noise reduction in a hearing aid device is described, with a signal, which comprises a useful and an interference signal part, being processed in the hearing aid device and with the interference signal part being reduced to the benefit of the useful signal part and with the reduction of the interference signal part being carried out as a function of the input level of the signal, with the interference signal part being more heavily attenuated with a high input level than with a low input level.

Abstract: An acoustic transducer device is provided. The device includes a body, a speaker, a microphone, and a processor. The body has a cavity, a sound exit, and a sound entrance. The cavity interflows with the sound exit and accommodates the speaker. The microphone is disposed within the body beside the speaker. The microphone interflows selectively with the cavity or the sound entrance. The processor is electrically connected to the speaker and the microphone. When the microphone interflows with the cavity, the microphone receives a sound signal generated from the cavity and transmits the sound signal to the processor for cancelling noise in the cavity. When the microphone interflows with the sound exit, the microphone receives an external sound signal and transmits the external sound signal to the processor for cancelling noise from the external.

Abstract: A system and method to detect and remediate unacceptable levels of speech intelligibility evaluates received test audio transmitted across and received in a space or region of interest. Intelligibility is improved by altering the rate, pitch, amplitude and frequency bands energy during presentation of the speech signal.

Abstract: Loudness-based compensation systems and techniques are described which provide audio compensation in noisy environments. Loudness approximations are determined for an audio block both by itself and in the presence of background noise. These approximations utilize compression of audio intensity within frequency bands in order to better reflect listeners' hearing perception. From these loudness approximations, a gain is determined for the audio block and then applied in such a manner that the effect is not jarring to listeners.

Abstract: The present invention provides a leakage detector enabling accurate and stable detection for occurrence and a position of leakage in a buried fluid pipe line and also making it possible for even those not so skilled in the leakage detection work to use the detector. The leakage detector 1 according to the present invention comprises a vibration detector 2 having a pickup 6 incorporating a piezoelectric element; a main body 4 of the detector incorporating voltage amplifiers 31, 37 for voltage-amplifying an output signal and a plurality types of noise removing units 33, 34, 35 for removing noises from the output signal, and a headphone 5. The main body 4 of the detector has a display unit 15 for displaying data for detected vibration sounds on a predefined screen.

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: According to one embodiment, an acoustic correction apparatus includes: a signal obtaining module configured to obtain an acoustic signal from a target space including an object and an external space; a signal output module configured to output to the target space a measurement signal; a coefficient identifying module configured to identify, on the basis of a response acoustic signal, a correction coefficient of a correction filter that reduces a resonance frequency component of a resonance in the object; a filtering module configured to use the correction filter, and filter the signal provided to the object; a noise cancelling module configured to remove, on the basis of the acoustic signal, a noise component comprised in the acoustic signal from the filtered signal; and an output module configured to output the acoustic signal, from which the noise component is removed by the noise cancelling module, to the object.

Abstract: An audio output apparatus includes: an audio codec outputting an analog audio signal corresponding to a digital audio signal from a system controller; a switch unit having a first end coupled to the audio codec through a capacitor, and a grounded second end; and a switch controller triggered by a trigger signal to output a control signal to a control end of the switch unit such that the switch unit couples the capacitor to ground in response to the control signal. The trigger signal is generated by one of the system controller, the audio codec, and a power circuit supplying electric power to the system controller, the audio codec and the switch controller upon occurrence of a condition associated with pop noise, and is outputted to the switch controller before the pop noise is generated, such that the pop noise is conducted to ground via the switch unit.

Abstract: A communication device contains multiple microphones that receive acoustic signals from a user and from the background. The acoustic signals from the user are enhanced using the background acoustic signals to reduce background noise. The enhanced signal are transmitted to an emergency network when an emergency call is made from the communication device. The raw signals are stored in the communication device for later retrieval or are transmitted simultaneously with the enhanced signals. The enhanced signals are transmitted using a circuit-switched voice mode while the raw signals are transmitted using a packet-switched voice mode.

Abstract: Apparatus and a corresponding method for processing speech signals in a noisy reverberant environment, such as an automobile. An array of microphones (10) receives speech signals from a relatively fixed source (12) and noise signals from multiple sources (32) reverberated over multiple paths. One of the microphones is designated a reference microphone and the processing system includes adaptive frequency impulse response (FIR) filters (24) enabled by speech detection circuitry (21) and coupled to the other microphones to align their output signals with the reference microphone output signal. The filtered signals are then combined in a summation circuit (18). Signal components derived from the speech signal combine coherently in the summation circuit, while noise signal components combine incoherently, resulting in composite output signal with an improved signal-to-noise ratio. The composite output signal is further processed in a speech conditioning circuit (20) to reduce the effects of reverberation.

Abstract: A method for processing an audio signal received through a microphone array coupled to an interfacing device is provided. The method is processing at least in part by a computing device that communicates with the interfacing device. The method includes receiving a signal at the microphone array and applying adaptive beam-forming to the signal to yield an enhanced source component of the signal. Also, an inverse beam-forming is applied to the signal to yield an enhanced noise component of the signal. The method combines the enhanced source component and the enhanced noise component to produce a noise reduced signal, where the noise reduced signal is a target voice signal. Then, monitoring an acoustic set-up associated with the audio signal as a background process using the adaptive beam-forming inverse beam-forming to track the target signal component, and periodically setting a calibration of the monitored acoustic set-up.

Abstract: A communication device includes: (1) a wireless adapter at which a wireless headset is communicatively connected to the communication device and at which is received a first acoustic input that includes a speech input and a first ambient noise input; (2) a microphone that receives a second acoustic input, which includes a second ambient noise input; and (3) a dual-channel adaptive noise canceller that utilizes the second ambient noise input to filter the first ambient noise input out of the first acoustic input to generate an acoustic output that primarily comprises the speech input.

Abstract: The adaptive beamformer unit (191) comprises: a filtered sum beamformer (107) arranged to process input audio signals (u 1, u2) from an array of respective microphones (101, 103), and arranged to yield as an output a first audio signal (z) predominantly corresponding to sound from a desired audio source (160) by filtering with a first adaptive filter (f1(-t)) a first one of the input audio signals (u1) and with a second adaptive filter (f2(-t)) a second one of the input audio signals (u2), the coefficients of the first filter (f1(-t)) and the second filter (f2(-t)) being adaptable with a first step size (a1) and a second step size ((x2) respectively; noise measure derivation means (111) arranged to derive from the input audio signals (u1, u2) a first noise measure (x1) and a second noise measure (x2); and an updating unit (192) arranged to determine the first and second step size (a1, (x2) with an equation comprising in a denominator the first noise measure (x1) for the first step size (a1), respectively the

Abstract: A method of suppressing noise in a signal containing speech and noise to provide a noise suppressed speech signal. An estimate is made of the noise and an estimate is made of speech together with some noise. The level of the noise included in the estimate of the speech together with some noise is variable so as to include a desired amount of noise in the noise-suppressed signal.

Abstract: An approach is provided for measuring, identifying, and removing at least one sinusoidal interference signal (Ak·ej(?kt+?k), Ak·ej(?·?k?t+?k)) in a noise signal (w(t), w(?·?t)). A frequency range to be measured is split into a plurality of frequency bands (?) via a Fast Fourier Transform (FFT) filter bank. For each of the frequency bands (?), an autocorrelation matrix ({circumflex over (R)}?) is determined, wherein parameters of the autocorrelation matrices ({circumflex over (R)}?) are variably adjusted based on whether the at least one sinusoidal interference signal (Ak·ej(?kt+?k), Ak·ej(?·?k?t+?k)) is to be measured, identified, or removed and further based on at least one averaging. The autocorrelation matrices ({circumflex over (R)}?) are jointly utilized for one or more of measuring, identifying, or removing the at least one sinusoidal interference signal (Ak·ej(?kt+?k), Ak·ej(?·?k?t+?k)) in the noise signal (w(t), w(?·?t)).

Abstract: This present invention discloses an audio-separating apparatus and operation method thereof. The audio-separating apparatus applies both blind source separation and noise reduction mechanisms. The audio-separating apparatus only uses one microphone to record mixed sound signals. After applying the noise reduction mechanism, noise reduced signals and the mixed sound signals are used as the inputs of the blind source separation. The method may avoid the spatial aliasing effect caused by using a microphone array to record the mixed sound signals. Besides, speech segment losses caused by processing the noise reduction will be effectively recovered, which may help the hearing impaired recognize target speech signals.

Abstract: A method for automatic speech recognition includes determining for an input signal a plurality scores representative of certainties that the input signal is associated with corresponding states of a speech recognition model, using the speech recognition model and the determined scores to compute an average signal, computing a difference value representative of a difference between the input signal and the average signal, and processing the input signal in accordance with the difference value.

Abstract: A broadside small array microphone beamforming unit comprises a first omni-directional microphone to generate a signal X1(t), a second omni-directional microphone to generate a signal X2(t), a first delay unit delaying the signal X1(t) to generate a signal X1(t?T), a second delay unit delaying the signal X2(t) to generate a signal X2(t?T), a first substrator subtracting the signal X1(t?T) from the signal X2(t) to generate a signal R(t)=X2(t)?X1(t?T), a second substrator subtracting the signal X2(t?T) from the signal X1(t) to generate a signal L(t)=X1(t)?X2(t?T), a third delay unit delaying the signal R(t) to generate a signal R?(t)=R(t?D), a gain function unit convoluting the signal L(t) with a gain function G(t) to generate a signal L?(t)=L(t)*G(t?i), and a substrator subtracting the signal L?(t) from the signal R?(t) to generate a signal B?(t)=R?(t)?L?(t).

Abstract: Techniques for introducing background noise segments into signal data are provided. The background noise segments are constructed from a background noise print extracted from the signal data. The background noise print may be user specified, or automatically identified by the signal editing tool. The background noise print may be stored with, and subsequently loaded as part of, the project associated with a signal. The background noise segments that are generated based on the background noise print may have different durations than the background noise print itself.

Abstract: Active noise cancellation is combined with spectrum modification of a reproduced audio signal to enhance intelligibility.

Abstract: A clear, high quality voice signal with a high signal-to-noise ratio is achieved by use of an adaptive noise reduction scheme with two microphones in close proximity. The method includes the use of two omini directional microphones in a highly directional mode, and then applying an adaptive noise cancellation algorithm to reduce the noise.

Abstract: There is provided a voice recognition device and a voice recognition method that enhance the function of noise adaptation processing in voice recognition processing and reduce the capacity of a memory being used. Acoustic models are subjected to clustering processing to calculate the centroid of each cluster and the differential vector between the centroid and each model, model composition between each kind of assumed noise model and the calculated centroid is carried out, and the centroid of each composition model and the differential vector are stored in a memory. In the actual recognition processing, the centroid optimal to the environment estimated by the utterance environmental estimation is extracted from the memory, model restoration is carried out on the extracted centroid by using the differential vector stored in the memory, and noise adaptation processing is executed on the basis of the restored model.

Abstract: A system (10) for beamforming using a microphone array, the system (10) comprising: a beamformer consisting of two parallel adaptive filters (12, 13), a first adaptive filter (12) having low speech distortion (LS) and a second adaptive filter (13) having high noise suppression (SNR); and a controller (14) to determine a weight (?) to adjust a percentage of combining the adaptive filters (12, 13) and to apply the weight to the adaptive filters (12, 13) for an output (15) of the beamformer.

Abstract: Provision to reduce production of musical noise. A noise reduction device includes: means for calculating a rank for each element included in a first region having predetermined sizes in the time axis direction and in the frequency axis direction, depending on a value of the element, in a noise section of an observed signal indicating variation of a frequency spectrum with time; means for calculating a rank for each element included in a second region, depending on a value of the element, the second region having predetermined sizes in the time axis direction and in the frequency axis direction in the observed signal; and means for subtracting, from the values of the respective elements in the second region, values based on the values of the respective elements in the first region whose ranks correspond to ranks of respective elements in the second region.

Abstract: An apparatus and method for active cancellation of interference at a hearing assistive devices are presented. The apparatus includes an interference cancellation circuit for cancelling an interference component of a composite signal, and an activator circuit for enabling interference cancellation circuit. The interference cancellation circuit generates an estimated replica of the interference from an interference profile, inverts the replica to form a cancellation waveform, then adds the cancellation waveform to the composite signal to cancel the interference component. An interference profile can be provided by performing a training sequence on a composite signal to detect a repetitive signal and building a profile using its parameters, retrieving a profile stored in memory, or using an antenna to capture an RF signal.

Abstract: In a wireless communication system, a method and apparatus for noise estimation of a received OFDM communication signal, wherein the signal comprises a data frame with a preamble having at least one long training field (LTF) containing two substantially similar OFDM symbols, comprise examining the LTF for substantially similar OFDM symbols. The noise power in the signal is estimated and the received signal power is measured. The signal to noise ratio is calculated and the signal power is determined by subtracting the noise power from the signal noise.

Abstract: Techniques for introducing background noise segments into signal data are provided. The background noise segments are constructed from a background noise print extracted from the signal data. The background noise print may be user specified, or automatically identified by the signal editing tool. The background noise print may be stored with, and subsequently loaded as part of, the project associated with a signal. The background noise segments that are generated based on the background noise print may have different durations than the background noise print itself.

Abstract: A method for separating a sound source from a mixed signal, includes Transforming a mixed signal to channel signals in frequency domain; and grouping several frequency bands for each channel signal to form frequency clusters. Further, the method for separating the sound source from the mixed signal includes separating the frequency clusters by applying a blind source separation to signals in frequency domain for each frequency cluster; and integrating the spectrums of the separated signal to restore the sound source in a time domain wherein each of the separated signals expresses one sound source.

Abstract: A mobile communication terminal equipped with speaker phone functionality and a method for removing feedback when the speaker phone is in use are disclosed. The mobile communication terminal includes: a first voice input path serving as a default voice input path; a second voice input path serving as an additional voice input path; and a controller for determining whether a speaker phone is in use, wherein the controller selects one of the first and second voice input paths according to whether the speaker phone is in use.

Abstract: A noise estimation unit estimates a noise signal in an input signal. A section decision unit distinguishes a target signal section from a noise signal section in the input signal. A noise suppression unit suppresses the noise signal based on a first suppression coefficient from the input signal. A noise excess suppression unit suppresses the noise signal based on a second suppression coefficient from the input signal. The second suppression coefficient is larger than the first suppression coefficient. A switching unit switches between an output signal from the noise suppression unit and an output signal from the noise excess suppression unit based on a decision result of the section decision unit.

Abstract: An audio signal enhancement device is provided. The device includes a first and a second microphone, placed as close together as possible, the first and second microphone having receiving surfaces facing in opposing directions. The first and second microphones receive a desired target audio signal originating in the proximity of the microphones and undesired noise signals not originating in the proximity of the microphones. The acoustic pressure gradient from the desired target signal between the first and the second microphones is greater than that from the noise signals. Signal processing logic is provided. The signal processing logic is configured to firstly generate a proximity-indicator signal and a pre-target-estimate signal through a combination of output from the first microphone and output of the second microphone.

Abstract: A method of removing noise and interference from a signal by receiving the signal, calculating a joint time-frequency domain of the signal, estimating instantaneous frequencies of the joint time-frequency domain, modifying each estimated instantaneous frequency, if necessary, to correspond to a frequency of the joint time-frequency domain to which it most closely compares, redistributing the elements within the joint time-frequency domain according to the estimated instantaneous frequencies as modified, computing a magnitude for each element in the joint time-frequency domain as redistributed, plotting the results as the time-frequency representation of the signal, identifying in the plot any noise and interference components in the received signal, eliminating from the redistributed joint time-frequency domain elements that correspond to noise and interference, and recovering a signal devoid of noise and interference from the modified redistributed joint time-frequency domain.

Abstract: An audio signal processing apparatus includes a splitting unit for splitting an audio signal of a first system and another audio signal of a second system into pluralities of frequency band components, a level comparing unit for calculating a level ratio or a level difference between each of the frequency bands of the first system and each of the frequency bands of the second systems, and an output control unit for removing frequency band components whose level ratio or level difference calculated by the level comparing unit is equal and substantially equal to a predetermined value from at least one of the first and second systems.

Abstract: An acoustic shock protection method and device are provided. A pattern analysis-based approach is taken to an input signal to perform feature extraction. A parameter space is identified, which is corresponding to the signal space of the input signal. A rule-based decision approach is taken to the parameter space to detect an acoustic shock event. The device may be advantageously implemented using a weighted overlap-add approach to provide low group delay, high-fidelity and a high degree of protection from acoustic shock events.

Abstract: In one aspect of the present invention, a method to reduce noise in a noisy speech signal is disclosed The method comprises: applying at least two versions of the noisy speech signal to a first filter, whereby that first filter outputs a speech reference signal and at least one noise reference signal, applying a filtering operation to each of the at least one noise reference signals, and subtracting from the speech reference signal each of the filtered noise reference signals, wherein the filtering operation is performed with filters having filter coefficients determined by taking into account speech leakage contributions in the at least one noise reference signal.

Abstract: The present invention provides a wireless headset with echo control and noise cancellation. The present invention also provides a method with phase reversion for echo control of a wireless headset wherein the wireless headset comprises closely disposed speakers and acoustic sensors. The present invention further provides a method with beamforming for noise cancellation of a wireless headset wherein the wireless headset comprises two separate units disposed in distance, and wherein each unit comprises an acoustic sensor.

Abstract: Disclosed herein is a noise canceling apparatus including: a microphone configured to pick up ambient sound as noise; a first signal generator configured to receive a signal from the microphone to generate a noise cancel signal that is inverted in phase to the signal received from the microphone and has an amplitude level considered with an attenuation in accordance with a distance from the microphone to an observation point separated away from the microphone; a first loudspeaker configured to be arranged in the proximity of the microphone and output the noise cancel signal; a second signal generator configured to receive the signal from the microphone to generate a positive-phase signal that has the same phase as that of the signal from the microphone; and a second loudspeaker configured to be arranged in the proximity of the microphone and output the positive-phase signal.

Abstract: A method for reducing noise associated with an audio signal received through a microphone sensor array is provided. The method initiates with enhancing a target signal component of the audio signal through a first filter. Simultaneously, the target signal component is blocked by a second filter. Then, the output of the first filter and the output of the second filter are combined in a manner to reduce noise without distorting the target signal. Next, an acoustic set-up associated with the audio signal is periodically monitored. Then, a value of the first filter and a value of the second filter are both calibrated based upon the acoustic set-up. A system capable of isolating a target audio signal from multiple noise sources, a video game controller, and an integrated circuit configured to isolate a target audio signal are included.

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: A signal generating section generates a main signal and a noise reference signal. A determining section determines whether a level ratio is larger than a predetermined value. An adaptive filter section generates a signal indicative of a signal component of a target sound included in the noise reference signal generated by the signal generating section, and learns a filter coefficient only when the determining section determines that the level ratio is larger than the predetermined value. A subtracting section subtracts the signal generated by the adaptive filter section from the noise reference signal. A noise suppressing section suppresses a signal component of noise included in the main signal by using the main signal and the noise reference signal after subtraction by the subtracting section.

Abstract: A stimulus analysis system and artifact rejection method are disclosed. The method may be used in many applications, such as, for example, DPOAE testing, TEOAE testing, BAER testing, ultrasound, MRI, RADAR, GPS, EEG, EKG, or communications. In one embodiment, a system receives a signal, and depending on the noise power of the signal, the signal is placed in one of a plurality of buffers or is discarded. This process is repeated. The combination of buffers that yields the lowest noise power is then selected. The selected combination of buffers may then be used to calculate a signal to noise ratio, which may be used to determine whether the signal received is acceptable, indicating, for example, that a test has been passed or failed.

Abstract: A method and apparatus determine a channel response for an alternative sensor using an alternative sensor signal and an air conduction microphone signal. The channel response is then used to estimate a clean speech value using at least a portion of the alternative sensor signal.

Abstract: A sound collection apparatus according to the present invention comprises: at least one target sound collection means for collecting a sound including a target sound generated from a target sound source, so as to output a collected-sound signal; a plurality of non-target sound collection means, provided at positions different from each other, each forming a dead zone of a sensitivity in a direction of the target sound source so as to collect a sound outside the dead zone and output a collected-sound signal; sensitivity suppression means for generating a sensitivity suppression signal for suppressing a sound collection sensitivity in an overlap region in which a plurality of the dead zones overlap each other, as compared to in a region surrounding the overlap region, by subjecting, to a predetermined signal processing, the collected-sound signal outputted by each of the plurality of non-target sound collection means; and extraction means for removing, from the collected-sound signal outputted by the at least o

Abstract: A playback device includes: a playback portion that plays back content and outputs at least an audio signal; an acquisition portion that acquires an external audio signal; a generating portion that, based on noise collected by a sound collecting device, generates a noise cancellation signal to reduce the noise; a switching portion that, if the acquisition portion has acquired the external audio signal when the playback portion is playing back content, switches an output signal from the audio signal to the external audio signal; and a synthesizing portion that synthesizes the output signal from the switching portion with the noise cancellation signal.

Abstract: A method, medium, and apparatus canceling noise from a mixed sound. The method includes receiving sound source signals including a target sound and noise, extracting at least one feature vector indicating an attribute difference between the sound source signals from the sound source signals, calculating a suppression coefficient considering ratios of noise to the sound source signals based on the at least one extracted feature vector, and canceling the sound source signals corresponding to noise by controlling an intensity of an output signal generated from the sound source signals according to the calculated suppression coefficient. Accordingly, a clear target sound source signal can be obtained.