Abstract: Psychoacoustic Bass Enhancement (PBE) is integrated with one or more other audio processing techniques, such as active noise cancellation (ANC), and/or receive voice enhancement (RVE), leveraging each technique to achieve improved audio output. This approach can be advantageous for improving the performance of headset speakers, which often lack adequate low-frequency response to effectively support ANC.

Abstract: An acoustic device is provided with first and second one or more acoustic elements to generate a first signal that includes mostly undesired audio and substantially void of desired audio, and a second signal that includes desired as well undesired audio respectively. The first one or more acoustic elements are designed and arranged to generate a Cardioid beam with a null at an originating direction of the desired audio. The second one or more acoustic elements are designed and arranged to generate a complementary beam that includes the desired audio. A system is provided with an appropriate signal processing logic to recover the desired audio using the first and second signals. The signal processing logic may practice echo cancellation like techniques or blind signal separation techniques.

Abstract: Methods and systems for masking audio noise are disclosed. One apparatus includes a silence detector configured to detect a period of substantial silence in an audio signal; a masking noise source operably coupled to the silence detector, the masking noise source configured to generate a noise signal in response to the silence detector detecting the period of substantial silence; and at least one combining device operably coupled to the masking noise source, the at least one combining device configured to contribute to combining the audio signal and the noise signal. A method includes detecting a period of substantial silence in an audio signal; and combining masking noise with the audio signal during the period of substantial silence.

Abstract: A method and device are provided for the objective evaluation of voice quality of a speech signal. The device includes: a module for extracting a background noise signal, referred to as a noise signal, from the speech signal; a module for calculating the audio parameters of the noise signal; a module for classifying the background noise contained in the noise signal on the basis of the calculated audio parameters, according to a predefined set of background noise classes; and a module for evaluating the voice quality of the speech signal on the basis of at least the resulting classification relative to the background noise in the speech signal.

Abstract: Quality of industrial products is evaluated by evaluating non-stationary operation sound, which is a kind of operation sound, from an aspect of tone, using closely simulated evaluation levels of evaluation of non-stationary sound by used of a human sense of hearing. An operation sound of a conforming product sample is converted into sound waveform data by a sound collecting unit, and the sound waveform data is input into a computer via an A-D converter, and then converted into psychoacoustic parameters. Data of pseudo conforming products is additionally obtained from the psychoacoustic parameters of a plurality of conforming product samples by making use of deviation in the data of the conforming product samples. Threshold data is obtained using thresholds and masking data for evaluation calculated from psychoacoustic parameters of data of the conforming product samples and the pseudo conforming products by a statistical technique.

Abstract: Enhanced speech is produced from a mixed signal including noise and the speech. The noise in the mixed signal is estimated using a vector-Taylor series. The estimated noise is in terms of a minimum mean-squared error. Then, the noise is subtracted from the mixed signal to obtain the enhanced speech.

Abstract: A method and a device for reducing snore annoyances include determining a snore sound pattern of a snoring person to predict an upcoming snore sound level. A faked snore sound is played to flatten the resulting snore sound level.

Abstract: In response to a first envelope within a kth frequency band of a first channel, a speech level within the kth frequency band of the first channel is estimated. In response to a second envelope within the kth frequency band of a second channel, a noise level within the kth frequency band of the second channel is estimated. A noise suppression gain for a time frame n is computed in response to the estimated speech level for a preceding time frame, the estimated noise level for the preceding time frame, the estimated speech level for the time frame n, and the estimated noise level for the time frame n. An output channel is generated in response to multiplying the noise suppression gain for the time frame n and the first channel.

Abstract: A microphone capable of canceling vibration noise caused by mechanical vibration is provided with, in capsules, a pair of diaphragms and a pair of back plates opposite to the respective diaphragms. A printed circuit board is disposed at the middle of capsules. A pair of diaphragms is disposed close and opposite to the surfaces of the printed circuit board with the printed circuit board disposed therebetween. The difference in distance from a vibration source to the two diaphragms is made small. The microphone has a high canceling effect for canceling vibration noise caused by mechanical vibration.

Abstract: Techniques are described for processing an audio signal to reduce the total harmonic distortion caused when it is reproduced by a loudspeaker, which is located within an audio reproduction device having an enclosure with an associated resonant frequency. After receiving the input audio signal, which includes the resonant frequency of the enclosure, the level of the input audio signal at the resonant frequency is reduced, thereby producing a first processed signal. In addition, the level of said input audio signal is reduced at all frequencies, producing a second processed signal. The first and second processed signals are combined to produce an output audio signal. The degree to which the level of the audio signal at both the resonant frequency and at all frequencies is reduced may be dependent upon the current volume level.

Abstract: The present invention relates to a multi-microphone system and method adapted to determine phase angle differences between a first microphone and a second microphone signal to detect presence of wind noise.

Abstract: An audio output circuit includes an on-chip left channel amplifier module, an on-chip center channel amplifier module, and an on-chip right channel amplifier module. A left channel IC pin is operably coupled to an output of the on-chip left channel amplifier module. A right channel IC pin is operably coupled to an output of the on-chip right channel amplifier module. A center channel IC pin is operably coupled to an output of the on-chip center channel amplifier module. A center channel feedback IC pin is operably coupled to an input of the on-chip center channel amplifier module to provide a feedback loop. A left jack connection is operably coupled to the left channel IC pin. A right jack connection is operably coupled to the right channel IC pin. A jack return connection coupled to the center feedback IC pin. An inductor has a first node coupled to the jack return connection and a second node coupled to the center channel IC pin.

Abstract: Some embodiments provide a media-editing application for identifying a set of background audio segments of a composite presentation, generating additional background audio segments based on several identified background audio segments, detecting regions of the composite presentation that lack background audio, and inserting generated background audio into the detected regions of the composite presentation.

Abstract: A sound processing apparatus includes a target sound emphasizing unit configured to acquire a sound frequency component by emphasizing target sound in input sound in which the target sound and noise are included, a target sound suppressing unit configured to acquire a noise frequency component by suppressing the target sound in the input sound, a gain computing unit configured to compute a gain value to be multiplied by the sound frequency component using a gain function that provides a gain value and has a slope that are less than predetermined values when an energy ratio of the sound frequency component to the noise frequency component is less than or equal to a predetermined value, and a gain multiplier unit configured to multiply the sound frequency component by the gain value computed by the gain computing unit.

Abstract: A method of compensating for noise in a receiver having a first receiver unit and a second receiver unit, the method includes receiving a first transmission at the first receiver unit, the first transmission having a first signal component and a first noise component; receiving a second transmission at the second receive unit, the second transmission having a second signal component and a second noise component; determining whether the first noise component and the second noise component are incoherent and; only if it is determined that the first and second noise components are incoherent, processing the first and second transmissions in a first processing path, wherein the first processing path is configured to compensate for incoherent noise.

Abstract: A method for reducing parasitic vibrations of a loudspeaker environment while maintaining the perception of the low frequencies of an original electric sound signal for broadcast after processing by the loudspeaker having a cut-off frequency. A vibration frequency band is identified that causes the loudspeaker to vibrate. A low frequency band of the original sound signal having a frequency close to the cut-off frequency of the loudspeaker is identified as an upper limit. At least one harmonic signal from the isolated low frequency band of the original sound signal is generated. The original sound signal and the harmonic signal are combined to obtain a recombined signal. The vibration frequency band is removed from the recombined signal to obtain a signal for broadcast by the loudspeaker.

Abstract: The present invention relates to a method of providing a compensation service for characteristics of a sound system using a smart device comprises: (1) receiving information of the sound system from the smart device; (2) retrieving the inputted sound characteristics of the sound system; and (3) transmitting a compensation signal according to the sound characteristics to the smart device. A user can be provided with a sound compensation service according to the sound characteristics of his/her sound system, i.e. a speaker, an earphone or a headphone, just by allowing a service providing server to receiving information about the sound system from the smart device, retrieving the sound characteristics, and then transmitting a compensation signal according to the sound characteristics to the smart device.

Abstract: Provided is a sound emission and collection device capable of estimating the azimuth of a sound source (such as a main utterer) precisely without any processing load. The sound emission and collection device (1) is connected with another sound emission and collection device via a network or the like. The sound emission and collection device (1) receives a sound signal from another sound emission and collection device, as a sound emission signal (FE), and emits the same from a speaker (SP). The sound emission and collection device (1) collects the sound at microphones (MIC1 to MIC3), and produces sound collection beam signals (NE1 to NE3) of different azimuths. The sound emission and collection device down-samples the individual sound collection beam signals (NE1 to NE3), and filters out the echoes of the down-sampled sound collection beam signals (DNE1 to DNE3).

Abstract: A signal processing method for enhancing the dynamic range of a signal is disclosed. The method comprises: a) forming an attenuated signal from an input signal; b) filtering each of the input and the attenuated signals such that the sum of their bandwidths is less than or equal to the bandwidth of a transmission channel; c) modulating a first one of the filtered input signal and the filtered attenuated signal, whereby the filtered input signal and the filtered attenuated signal occupy respective non-overlapping frequency ranges within the bandwidth of the transmission channel; and d) combining the modulated signal with the second one of the filtered input signal and the filtered attenuated signal to form a composite output signal.

Abstract: Methods and systems to produce audio output signals from audio input signals. In one embodiment, a first portion of the audio input signals can be pre-processed, with the output used to modulate ultrasonic carrier signals, thereby producing modulated ultrasonic signals. The modulated ultrasonic signals can be transformed into a first portion of the audio output signals, which is directional. Based on a second portion of the audio input signals, a standard audio speaker can output a second portion of the audio output signals. Another embodiment further produces distortion compensated signals based on the pre-processed signals. The distortion compensated signals can be subtracted from the second portion of the audio input signals to generate inputs for the standard audio speaker to output the second portion of the audio output signals. In yet another embodiment, noise can be added during pre-processing of the first portion of the audio input signals.

Abstract: Systems and methods for adaptive power control are provided. In exemplary embodiments, a primary signal is received. A noise power level of the primary signal is then estimated. The noise power level may then be compared to at least one power threshold. Subsequently, a large power consuming system is controlled based on the comparison of the noise power level to the power threshold.

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: An input audio signal is detected to provide a fast time constant control signal which is clamped for immediate response when the input signal drops below a threshold level and filtered to provide a modified control signal with a slow time constant. As the modified control signal level decreases toward the control signal level, a time constant differential control signal decreases and slows the rate of change of the decreasing modified control signal which, as it nears the control signal level, reverts to a slow response. The resulting control signal has an exponential release response which can be applied to an audio dynamics processor. The time constant differential control signal may also be detected to provide an inverted output when it is significantly less than the modified control signal so as to result in the control signal also having an exponential attack response to be applied to the processor.

Abstract: An electronic device measures noise variability of background noise present in a sampled audio signal, and determines whether the measured noise variability is higher than a high threshold value or lower than a low threshold value. If the noise variability is determined to be higher than the high threshold value, the device categorizes the background noise as having a high degree of variability. If the noise variability is determined to be lower than the low threshold value, the device categorizes the background noise as having a low degree of variability. The high and low threshold values are between a high boundary point and a low boundary point. The high boundary point is based on an analysis of files including noises that exhibit a high degree of variability, and the low boundary point is based on an analysis of files including noises that exhibit a low degree of variability.

Abstract: Methods and system are described for cancelling interference in a microphone system. A positive bias voltage is applied to a first microphone diaphragm and a negative bias voltage is applied to a second microphone diaphragm. The diaphragms are configured to exhibit substantially the same mechanical deflection in response to acoustic pressures received by the microphone system. A differential output signal is produced by combining a positively-biased output signal from the first microphone diaphragm and a negatively-biased output signal from the second microphone diaphragm. This combining cancels common-mode interferences that are exhibited in both the positively-biased output signal and the negatively-biased output signal.

Abstract: A dual omnidirectional microphone array noise suppression is described. Compared to conventional arrays and algorithms, which seek to reduce noise by nulling out noise sources, the array of an embodiment is used to form two distinct virtual directional microphones which are configured to have very similar noise responses and very dissimilar speech responses. The only null formed is one used to remove the speech of the user from V2. The two virtual microphones may be paired with an adaptive filter algorithm and VAD algorithm to significantly reduce the noise without distorting the speech, significantly improving the SNR of the desired speech over conventional noise suppression systems.

Abstract: Broadly speaking, the embodiments disclosed herein describe an apparatus, system, and method for managing the effects of TDMA noise emitted by a communication device on an audio circuit.

Abstract: There is provided an integrated circuit device having a wiring board 1200?, the wiring board 1200? including: a first vibrating membrane 714-1 which forms a first microphone; a second vibrating membrane 714-2 which forms a second microphone; and a differential signal generating circuit 720 which receives a first voltage signal obtained in the first microphone and a second voltage signal obtained in the second microphone and generates a differential signal indicating a difference between the first and second voltage signals, and a voice input device and an information processing system including the same. Accordingly, it is possible to realize a voice input element having a small size and a noise removal function with high accuracy.

Abstract: An audio interface adapted to reduce a subscriber voice may receive a subscriber voice and a background noise. The subscriber voice may then be compared to the to the background noise. If the received subscriber voice is louder than the received background noise, the audio interface may output a message to the cellular telephone subscriber indicating the subscriber may reduce his speaking volume. Additionally, an audio interface may process a voice waveform that corresponds to the subscriber voice and a background waveform that corresponds to the background noise to generate a substantially opposite voice waveform and a substantially opposite background waveform respectively. The substantially opposite voice waveform and background waveform may be substantially out of phase from the voice waveform and background waveform respectively and output via one or more output ports of the audio interface.

Abstract: The suppression of off-axis audio in an audio environment is provided. Off-axis audio may be considered audio that does not originate from a region of interest. The off-axis audio is suppressed by comparing a phase difference between signals from two microphones to a target slope of the phase difference between signals originating from the region of interest. The target slope can be adapted to allow the region of interest to move with the location of a human speaker such as a driver.

Abstract: The invention relates to an audio device for use proximate a user's ears. The audio device includes first and second audio transmitting/receiving devices that are capable of operating in stereo. The audio device may be used within a system for manipulating audio signals received by the device. The manipulation may include processing received audio signals to enhance their quality. The processing may include applying one or more audio enhancement algorithms such as beamforming, active noise reduction, etc. A corresponding method for manipulating audio signals is also disclosed.

Abstract: A novel adaptive beamforming technique with enhanced noise suppression capability. The technique incorporates the sound-source presence probability into an adaptive blocking matrix. In one embodiment the sound-source presence probability is estimated based on the instantaneous direction of arrival of the input signals and voice activity detection. The technique guarantees robustness to steering vector errors without imposing ad hoc constraints on the adaptive filter coefficients. It can provide good suppression performance for both directional interference signals as well as isotropic ambient noise.

Abstract: A switch controller is provided that uses one or more capacitors to generate a slow turn on/slow turn off switch control signals to suppress audible switching noise in an audio switch. In some embodiments, an analog inverter and a capacitor are used to generate the switch control signals, while in other embodiments two capacitors are used to generate the switch control signals. To conserve power between switching states, routing logic is provided that ties the switch control signals to respective voltage rails and disables selected portions of the switch controller.

Abstract: In order to reduce interference in an audio signal during a call on a mobile communication device, a plurality of transforms of the audio signal is performed, each transform containing phase information and amplitude information of corresponding samples of the audio signal. The results of the transforms are then averaged in order to generating a compensation signal that can be subtracted from the audio signal.

Abstract: A signal processing apparatus includes: a separation processing unit that generates observed signals in the time frequency domain by performing the short-time Fourier transform on mixed signals as outputs, which are acquired from a plurality of sound sources by a plurality of sensors, and generates sound source separation results corresponding to the sound sources by a linear filtering process on the observed signals. The separation processing unit has a linear filtering process section that performs the linear filtering process on the observed signals so as to generate separated signals corresponding to the respective sound sources, an all-null spatial filtering section that applies an all-null spatial filter to generate signals filtered with the all-null spatial filter (spatially filtered signals) in which the acquired sounds in null directions are removed, and a frequency filtering section that performs a filtering process by inputting the separated signals and the spatially filtered signals.

Abstract: The present invention discloses an echo cancellation method. The method includes: dividing an audio signal into a high-band audio signal and a low-band audio signal; performing adaptive filtering on the low-band audio signal, and performing synthesis filtering on a signal obtained after the low-band audio signal undergoes the adaptive filtering and on the high-band audio signal to generate a preliminary echo cancellation signal; performing envelope predication echo suppression on a high-band signal in the preliminary echo cancellation signal, and calculating and outputting a residual echo suppression coefficient; performing echo suppression on a low-band signal in the preliminary echo cancellation signal, and outputting a processing result; and multiplying the output result by the residual echo suppression coefficient, and outputting a signal of which echoes are canceled.

Abstract: A compact noise suppression circuit for a small speaker system and having a circuit board with a left signal input connection and a right signal input connection and a ground connection, and further having a left speaker supply connection and a left speaker return connection, and a right speaker supply connection and a right speaker return connection, and further having a left bifilar coil and a right bifilar coil, and left and right coil connections on the circuit board, and means securing said left and right bifilar coils to respective coil connections on the circuit board. Also disclosed is a compact portable audio player system incorporating such a noise suppression circuit, and a noise suppression circuit for a mon-aural signal system.

Abstract: Coefficient generator generates a crosstalk coefficient that is a predetermined value and that is used to calculate the amount of crosstalk of an echo. Converter uses either the output signal of a sound pickup device or the signal obtained by subtracting the output signal of an echo canceller from the output signal of the sound pickup device as a first signal, corrects the first signal based on the crosstalk coefficient generated in coefficient generator, and produces a near-end signal obtained by removing the echo from the first signal.

Abstract: A recording/reproducing apparatus includes a plurality of unidirectional microphones and a plurality of direction indicator switches. The unidirectional microphones are each disposed in the periphery with a predetermined angle interval therebetween, whereas the direction indicator switches are able to indicate other directions other than a plurality of directions corresponding to the unidirectional microphones. In a normal reproducing mode, a plurality of audio signals which are picked up by the unidirectional microphones and subsequently recorded is read and reproduced in parallel. When any one of the direction indicator switches is operated, only the audio signal emitted in the designated direction is selectively read and reproduced. When another direction other than a plurality of directions corresponding to the unidirectional microphones is designated, audio signals picked up by two unidirectional microphones which are disposed to sandwich the designated direction is selectively read and reproduced.

Abstract: A voice input device includes a first microphone, a second microphone, and a processor. The second microphone has a lower distance decay rate than the first microphone. The processor is configured to acquire noise information of noise by comparing a first signal obtained from the first microphone with a second signal obtained from the second microphone. The processor is further configured to perform noise suppression processing based on the noise information.

Abstract: A signal processing method for converting a signal received via a transmission path or read from a storage medium into a first audible signal, and suppressing a noise other than a desired signal contained in the first audible signal based on predetermined audio quality adjustment information, comprising steps of: in suppressing a noise other than a desired signal contained in the first audible signal to generate an enhanced signal, receiving audio quality adjustment information for adjusting audio quality; and adjusting audio quality of the enhanced signal using the audio quality adjustment information.

Abstract: Decomposition of a multi-source signal using a basis function inventory and a sparse recovery technique is disclosed.

Abstract: A motherboard includes a motherboard power supply connector and a time delay circuit. The motherboard power supply connector connects a power supply unit. The motherboard power supply connector has a power supply on pin and a power good pin. The power good pin is configured for receiving a power good signal from the power supply unit. The time delay circuit has an input terminal and an output terminal. The input terminal is configured for receiving a power supply on signal. The output terminal is connected to the power supply on pin and is configured for sending the power supply on signal to the power supply on pin after a time delay determined by the time delay circuit.

Abstract: A content reproduction device including: a microphone that collects noise in the surroundings of a casing; a feature amount extractor that extracts a plurality of feature amounts; a distance calculator that calculates an intervector distance between the extracted feature amount vector and a feature amount vector with the same dimensions which is set in advance as a feature amount of a waveform of a music signal; a determinator that determines whether or not music is included in the sounds collected by the microphone; a processor that processes the signal of the sounds collected by the microphone to change the volume or frequency characteristics of the sounds collected by the microphone; and an adder that adds and outputs the signal of the sounds collected by the microphone and the signal of sounds of reproduced content.

Abstract: According to an embodiment, a method of reducing noise in a signal received at a processing stage of an acoustic system includes, at the processing stage identifying at least one frequency which causes a system gain of the acoustic system to be above an average system gain of the acoustic system; providing a noise attenuation factor for reducing noise in the signal for the at least one frequency, the noise attenuation factor for the at least one frequency based on the system gain for that frequency; and applying the noise attenuation factor to a component of the signal at that frequency.

Abstract: An echo suppression system and method, and a computer-readable storage medium that is configured with instructions that when executed carry out echo suppression. Each of the system and the method includes the elements of a linear echo suppressor having a reference signal path, with a nonlinearity introduced in the reference signal path to introduce energy in spectral bands. Unlike an echo canceller, the echo suppression system and method are relatively robust to errors in the introduced nonlinearity.

Abstract: An adaptive power control monitors the noise level within a primary acoustic signal and compares the noise level to a threshold. If the noise level is lower than the threshold, a noise suppression system is deactivated and bypass filtering and cross fading are enabled. If the noise level is higher than the threshold, the noise suppression system is activated.

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: The present technology minimizes undesirable effects of multi-level noise suppression processing by applying an adaptive equalization. A noise suppression system may apply different levels of noise suppression based on the (user-perceived) signal-to-noise-ratio (SNR). The resulting high-frequency data attenuation may be counteracted by adapting the signal equalization. The present technology may be applied in both transmit and receive paths of communication devices. Intelligibility may particularly be improved under varying noise conditions, e.g. when a cell phone user is moving in and out of noisy environments.

Abstract: An signal processing apparatus, system and software product for audio modification/substitution of a background noise generated during an event including, but not be limited to, substituting or partially substituting a noise signal from one or more microphones by a pre-recorded noise, and/or selecting one or more noise signals from a plurality of microphones for further processing in real-time or near real-time broadcasting.