Abstract: An active vibration noise control device is preferably used for canceling a vibration noise by making a speaker generate a control sound. Concretely, the active vibration noise control device includes an attenuating unit which attenuates a control signal generated by an adaptive notch filter, and provides the attenuated control signal to the speaker, when amplitude of a filter coefficient is larger than a threshold. Therefore, it is possible to suppress continuous ups and downs of the filter coefficient during an occurrence of an abnormality. Hence, it becomes possible to appropriately suppress an occurrence of a cyclic abnormal sound and an increase in the vibration noise during the occurrence of the abnormality.

Abstract: A buzz detecting method and a buzz detecting system are provided for testing whether an under-test sound playing device generates a buzz while playing sound. By an application program module, plural under-test sound signals from the under-test sound playing device are converted into plural under-test frequency-domain signals corresponding to the under-test sound signals through Fourier transform. Moreover, the application program module calculates plural under-test noise ratios corresponding to the frequencies of respective under-test sound signals according to respective under-test frequency-domain signals. After the plural under-test noise ratios are compared with plural standard noise ratios from a standard sound playing device, the application program module may automatically judge whether the under-test sound playing device generates a buzz while playing sound. Since the testing procedure does not need to be implemented by the trained testers, the overall efficiency is largely enhanced.

Abstract: Maximum likelihood bit-stream generation and detection techniques are provided using the M-algorithm and Infinite Impulse Response (IIR) filtering. The M-Algorithm is applied to a target input signal X to perform Maximum Likelihood Sequence Estimation on the target input signal X to produce a digital bit stream B, such that after filtering by an IIR filter, the produced digital stream Y produces an error signal satisfying one or more predefined requirements. The predefined requirements comprise, for example, a substantially minimum error. In an exemplary bit detection implementation, the target input signal X comprises an observed analog signal and the produced digital stream Y comprises a digitized output of a receive channel corresponding to a transmitted bit stream. In an exemplary bit stream generation implementation, the target input signal X comprises a desired transmit signal and the produced digital stream Y comprises an estimate of the desired transmit signal.

Abstract: A method for configuring an electronic device having an equalizer suited for processing a numeric audio signal according to an objective equalization improves the equalization of the numeric audio signal. The equalizer has a transfer function that implements a set of band-pass filters having a constant band-pass and a configurable weight in the transfer function. The method provides a set of objective gains representing the objective equalization, with each objective gain corresponding to a respective frequency. The method also determines values of the weights of the band-pass filters that minimize an error evaluated by comparing the set of objective gains to the transfer function. The set of band-pass filters includes at least one pair of band-pass filters having a common central frequency and distinct band-passes. The method also sets the weights of the band-pass filters to the determined values.

Abstract: An active vibration noise control device cancels vibration noise by making plural speakers generate control sounds. The active vibration noise control device selects one or more speakers which output the control sounds, from plural speakers, based on a relationship between (1) a first phase difference which corresponds to a difference between phase characteristics of the vibration noise from a vibration noise source to an evaluation point and phase characteristics of the vibration noise from the vibration noise source to a pseudo evaluation point and (2) a second phase difference for each of the plural speakers corresponding to a difference between phase characteristics of the control sound from the speaker to the evaluation point and phase characteristics of the control sound from the speaker to the pseudo evaluation point. Therefore, it stably decreases the vibration noise at the pseudo evaluation point independently of a frequency band of the vibration noise.

Abstract: Audible crosstalk can be mitigated in a low-cost three-wire device having audio capability and/or voice applications. In some embodiments, a voltage can be introduced in a microphone power supply that is approximately the same as a measured noise voltage and the resulting voltage appearing at a microphone output can be optimized to mitigate the noise voltage and, thus, the presence of crosstalk. In some embodiments, a microphone within a circuit can be isolated to mitigate crosstalk by introducing current into a circuit that is approximately the same as a measured current, but having a flow in an opposite direction.

Abstract: A space-time adaptive beamformer for reducing noise in a vehicle that includes two or more microphones. A first weighting network is used for adjusting the signal characteristics of at least one output of the microphones while at least one delay network is also used for delaying the output in time of at least one output of the microphones. A second weighting network then adjusts the signal characteristics of the output of each of the delay networks and at sum adder works to combine the output of the first weighting network and the second weighting network. Finally, an output of the sum adder is combined with an artificial noise free reference signal to provide a low distortion noise reduced output. By generating a desired signal that acts as an artificial noise free signal reference, adequate noise reduction to be obtained without the distortion created due to processing non-linearity.

Abstract: An echo path change detector may be used to control the rate of adaptation in an acoustic echo canceller. When echo path change is declared, the rate of adaptation may be increased. However, echo path change should not be declared in the presence of double talk, because rapid adaptation during double talk is undesirable. Accordingly, various features are disclosed for detecting echo path changes while avoiding the declaration of such changes in the presence of double talk.

Abstract: Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.

Abstract: There is provided a noise cancellation system, comprising: an input for a digital signal, the digital signal having a first sample rate; a digital filter, connected to the input to receive the digital signal; a decimator, connected to the input to receive the digital signal and to generate a decimated signal at a second sample rate lower than the first sample rate; and a processor. The processor comprises: an emulation of the digital filter, connected to receive the decimated signal and to generate an emulated filter output; and a control circuit, for generating a control signal on the basis of the emulated filter output. The control signal is applied to the digital filter to control a filter characteristic thereof.

Abstract: To provide a noise reduction transmitter which can secure clarity of sounds collected in very noisy environments and maintain a quality of sounds without devising a noise insulation cover particularly. A transmission microphone 7 is arranged inside a noise insulation cover 2 worn on and covering at least a user's 1 mouth. A noise detection microphone 9 which detects external noises is arranged outside the noise insulation cover, and a noise component cancellation circuit 11 is provided which generates a noise component cancellation signal based on an output signal from the noise detection microphone. An electroacoustic transducer 8 is arranged in the noise insulation cover to reproduce a noise component cancellation sound based on an output signal from the noise component cancellation circuit 11.

Abstract: An active noise control (ANC) system may be implemented to both sides of a fan, such that both directions of the noise emitting are treated to reduce the overall noise. The impact on airflow is minimal, and the technique is very effective in a broad range of low frequencies. Passive sound-absorbing materials may be included for attenuation of high frequencies. The resulting quiet fan produces a low level of noise compared to any other device based on fan, which produces the same capacity of airflow. The quiet fan may be incorporated in any mechanical system which requires airflow induction such as: computers, air conditioners, machines, and more.

Abstract: The initial values of parameter estimates are set, including reverberation parameter estimates, which includes a regression coefficient used in a linear convolutional operation for calculating an estimated value of reverberation included in an observed signal, source parameter estimates, which includes estimated values of a linear prediction coefficient and a prediction residual power that identify the power spectrum of a source signal, and noise parameter estimates, which include noise power spectrum estimates. Then, the maximum likelihood estimation is used to alternately repeat processing for updating at least one of the reverberation parameter estimates and the noise parameter estimates and processing for updating the source parameter estimates until a predetermined termination condition is satisfied.

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: The present technology provides robust, high quality dereverberation of an acoustic signal which can overcome or substantially alleviate the problems associated with the diverse and dynamic nature of the surrounding acoustic environment. The present technology utilizes acoustic signals received from a plurality of microphones to carry out a multi-faceted analysis which accurately identifies reverberation based on the correlation between the acoustic signals. Due to the spatial distance between the microphones and the variation in reflection paths present in the surrounding acoustic environment, the correlation between the acoustic signals can be used to accurately determine whether portions of one or more of the acoustic signals contain desired speech or undesired reverberation. These correlation characteristics are then used to generate signal modifications applied to one or more of the received acoustic signals to preserve speech and reduce reverberation.

Abstract: A method and an arrangement for processing audio data, and a corresponding computer program and a corresponding computer-readable storage medium, which can be used, in particular, in the field of audio software and sampling. At least a first spectrum with at least one first spectral property is removed from the spectrum of the audio data, the resulting spectrum of the audio data is transformed after removal of the at least one first spectrum, and the at least one first spectrum or at least one of the first spectrum and/or at least one second spectrum with at least one second spectral property are impressed on the transformed spectrum.

Abstract: An active noise control system includes a plurality of adaptive filters. The plurality of adaptive filters each receives an input signal representative of an undesired sound. The adaptive filters may each generate an output signal based on the input signal. The output signals are used to generate an anti-noise signal configured to drive a speaker to produce sound waves to destructively interfere with the undesired sound.

Abstract: A method estimates noise power spectral density (PSD) in an input sound signal to generate an output for noise reduction of the input sound signal. The method includes storing frames of a digitized version of the input signal, each frame having a predefined number N2 of samples corresponding to a frame length in time of L2=N2/sampling frequency. It further includes performing a time to frequency transformation, deriving a periodogram comprising an energy content |Y|2 from the corresponding spectrum Y, applying a gain function G(k,m)=f(?s2(km),?w2l (k,m?1), |Y(k,m)|2), to estimate a noise energy level |?|2 in each frequency sample, where ?s2 is the speech PSD and ?w2 the noise PSD. It further includes dividing spectra into a number of sub-bands, and providing a first estimate |{circumflex over (N)}|2 of the noise PSD level in a sub-band and a second, improved estimate |{circumflex over (N)}|2 of the noise PSD level in a subband by applying a bias compensation factor B to the first estimate.

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

Abstract: Disclosed is a noise suppression device including an input signal analyzer 8 that analyzes the harmonic structure and periodicity of a plurality of input signals on the basis of the power spectra of the plurality of input signals, a power spectrum synthesizer 9 that synthesizes the power spectra of the plurality of input signals to generate a synthesized power spectrum according to the result of the analysis by the input signal analyzer 8, a noise suppression amount calculator 10 that calculates an amount of noise suppression on the basis of the synthesized power spectrum generated by the power spectrum synthesizer 9 and an estimated noise spectrum estimated from the input signals, and a power spectrum suppressor 11 that carries out noise suppression on the synthesized power spectrum generated by the power spectrum synthesizer 9 by using the amount of noise suppression calculated by the noise suppression amount calculator 10.

Abstract: An electroacoustic channel soundfield is altered. An audio signal is applied by an electromechanical transducer to an acoustic space, causing air pressure changes therein. Another audio signal is obtained by a second electromechanical transducer, responsive to air pressure changes in the acoustic space. A transfer function estimate of the electroacoustic channel is established, responsive to the second audio signal and part of the first audio signal. The transfer function estimate is derived to be adaptive to temporal variations in the electroacoustic channel transfer function. Filters are obtained with transfer functions based on the transfer function estimate. Part of the first audio signal is filtered therewith.

Abstract: In an aspect, the invention features an active noise reduction device including an electronic signal processing circuit. The electronic signal processing circuit includes a first input for accepting a first signal, a second input for accepting a second signal, an output for providing a third signal, a feed-forward path from the first input to the output, and a feed-forward controller for determining the control parameter by calculating a control signal using the first signal and the second signal and then using the control signal to determine the control parameter. The feed-forward path includes a fixed compensation linear filter and a variable compensation filter having an input for receiving a control parameter that applies a selected linear filter from a family of linear filters that vary in both gain and spectral shape and are selectable by the control parameter.

Abstract: An active noise reduction earphone. The earphone includes structure for positioning and retaining the earphone in the ear of a user without a headband, active noise reduction circuitry including an acoustic driver with a nominal diameter greater than 10 mm oriented so that a line parallel to, or coincident with, an axis of the acoustic driver and that intersects a centerline of the nozzle intersects the centerline of the nozzle at angle ?>±30 degrees. A microphone is positioned adjacent an edge of the acoustic driver. The earphone is configured so that a portion of the acoustic driver is within the concha of a user and another portion of the acoustic driver is outside the concha of the user when the earphone is in position. An opening coupling the nozzle to the environment includes impedance providing structure in the opening.

Abstract: A headset includes first and second earcups each having a front opening adapted to be adjacent to a respective ear of a user and including an electroacoustic transducer, a headband coupled to each of the earcups, and an active noise reduction circuit coupled to the electroacoustic transducers. The headband is configurable between at least two configurations that each press the earcups against the head of the user with different amounts of force. The active noise reduction circuit is configured to determine which of the at least two configurations the headband may be configured in, to provide a different amount of noise reduction when the headband may be configured in each of the different amounts of force, and to automatically transition between the different amounts of noise reduction in response to a change in the configuration of the headband between the at least two configurations.

Abstract: A method for a multi microphone noise reduction in a complex noisy environment is proposed. A left and a right noise power spectral density for a left and a right noise input frame is estimated for computing a diffuse noise gain. A target speech power spectral density is extracted from the noise input frame. A directional noise gain is calculated from the target speech power spectral density and the noise power spectral density. The noisy input frame is filtered by Kalman filtering method. A Kalman based gain is generated from the Kalman filtered noisy frame and the noise power spectral density. A spectral enhancement gain is computed by combining the diffuse noise gain, the directional noise gain, and the Kalman based gain. The method reduces different combinations of diverse background noise and increases speech intelligibility, while guaranteeing to preserve the interaural cues of the target speech and directional background noises.

Abstract: A method and system are provided in which a device, such as an acoustic echo canceller, may reduce the residual echo that may be heard at the far end of a conversation when an external speaker volume is changed. The device may compute a gain based on an echo estimate produced by a filter and on a near-end signal comprising audio information. The gain may be based on a correlation of the echo estimate and the near-end signal that tracks the changes in volume. Once computed, the gain may be validated to ensure that it is being applied when appropriate. The echo estimate may be adjusted by first applying the gain to an output of the filter and subsequently scaling a value of each of the coefficients of the filter based on the gain. The gain may be smoothed out over consecutive frames based on several adaptation schemes.

Abstract: An apparatus is provided for suppressing an echo signal included in a measured signal corresponding to a measured sound. In the apparatus, the measured signal and a reference signal in a time domain are transformed into a frequency domain, and calculated for obtaining each value of a ratio and a correlation between the measured signal and the reference signal in the frequency domain. With executing a comparison of the values of the ratio and the correlation, a coefficient is derived, where a product of the coefficient and the measured sound in the frequency domain gives an estimated value of the echo signal. The echo in the measured signal is suppressed with subtracting the estimation of the echo signal from the measured signal, respectively in the frequency domain.

Abstract: A multiband dynamics compressor implements a solution for minimizing unwanted changes to the long-term frequency response. The solution essentially proposes undoing the multiband compression in a controlled manner using much slower smoothing times. In this regard, the compensation provided acts more like an equalizer than a compressor. What is applied is a very slowly time-varying, frequency-dependent post-gain (make-up gain) that attempts to restore the smoothed long-term level of each compressor band.

Abstract: An Active Noise Control (ANC) for controlling a noise produced by a noise source may include an acoustic sensor (212) to sense a noise pattern and to produce a noise signal corresponding to the sensed noise pattern, an estimator (202) to produce a predicted noise signal by applying an estimation function to the noise signal, and an acoustic transducer (216) to produce a noise destructive pattern based on the predicted noise signal.

Abstract: An echo cancelling algorithm in a communication device initializes a step size value used in an adaptive echo filter based on a background noise signal power level relative to a power level of a received signal and a power level of an echo estimate relative to an output of an echo canceller. The algorithm then adjusts the step size value. One aspect adjusts the step size based on the detection of large fast fourier transform values at one, or more, disturbing-signal frequencies. Another aspect estimates residual echo energy to adjust an estimated echo energy, which then is used to set a double talk flag if a transmit signal has much more power than the estimated echo signal. Another aspect compares transmit signal power to a decimated version of the transmit signal power and sets the double talk flag if the former exceeds the latter by a predetermined amount.

Abstract: A frequency-domain based echo and NEXT canceller is provided. The canceller uses log2 encoding to precondition the error signal representing the echo. An improved gradient constraint is applied on at least a portion of a full weight vector in a least-mean-square algorithm. The least-mean-square algorithm is used to compute filter coefficients. The filter coefficients are multiplied by a frequency-domain data vector using a frequency-domain multiplier to generate frequency-domain output vector.

Abstract: Secure outsourced aggregation of data using one-way chains is discussed in this application. Each input data source such as a sensor generates a Verifiable Synopsis (“VS”) which includes sensor data, an Inflation Free Proof (“IFP”) generated using a cryptographic function and a Self-Authenticating Value (“SEAL”) chain generated using a one-way function. An aggregator takes a plurality VSs from multiple data sources and aggregates them together into one. Maximum value, top-k, count, count distinct, sum, average, and other aggregate functions may be used. Folded VS provides a concise proof that no value greater than the maximum value was reported by a sensor, thus providing a check against deflation of sensor data. Similarly, the cryptographic function of the IFP provides a mechanism to prevent inflation of the sensor data. Thus it becomes possible at a portal to verify that aggregated data has not been inflated or deflated by the aggregator.

Abstract: An echo canceller for improved recognition and removal of an echo from a communication device. The echo canceller can dynamically reduce echo using an improved energy estimator and an improved adaptive filter. The improved energy estimator can determine if conversation is in a single talk period or a double talk period based on the combined energy of both the near end background noise and speech. The improved adaptive filter can reduce echo by dynamically changing adaptation speed or step size. In double talk, the adaptive filter(s) can dynamically slow-down or stop adaptation. In single talk, the filter can dynamically increase the speed of adaptation to improve accuracy, or decrease adaptation speed for stability.

Abstract: The sound device includes an audio-information output unit, an analysis unit, an audio-division-spectrum output unit, a noise-division-spectrum output unit and a correction unit. The analysis unit receives audio information from the audio-information output unit, and then outputs sound spectrum information. The noise-division-spectrum output unit outputs sound-volume information for each critical band width of a noise, and the audio-division-spectrum output unit outputs the sound-volume information for each critical band width of the sound-spectrum information. The correction unit corrects the information from the audio-division-spectrum output unit based on the information from the noise-division-spectrum output unit. The audio-signal properties can be well corrected corresponding to the auditory-sense properties of the human, and thus the audio sound, in which an uncomfortable feeling to the auditory sense of the human has been adequately controlled, can be transmitted to a user.

Abstract: A sound reproducing device has a loudspeaker arranged to produce sound from an audio signal provided by an audio signal source. A microphone is positioned to pick up ambient noise and generate a microphone signal which comprises the noise. An ambient noise cancellation (ANC) system receives the microphone signal from the microphone and generates anti-noise corresponding to the ambient noise in the microphone signal. An automatic polarity adaptation (AAP) system monitors the ANC system and, when a decision criterion is fulfilled, causes a switch in polarity for the generated anti-noise.

Abstract: An active noise cancellation system that reduces, at a listening position, power of a noise signal radiated from a noise source to the listening position. The system includes an adaptive filter, at least one acoustic actuator and a signal processing device. The adaptive filter receives a reference signal representing the noise signal, and provides a compensation signal. The at least one acoustic actuator radiates the compensation signal to the listening position. The signal processing device evaluates and assesses the stability of the adaptive filter.

Abstract: A method and system for enhancing the frequency response of speech signals are provided. An average speech spectral shape estimate is calculated over time based on the input speech signal. The average speech spectral shape estimate may be calculated in the frequency domain using a first order IIR filtering or “leaky integrators.” Thus, the average speech spectral shape estimate adapts over time to changes in the acoustic characteristics of the voice path or any changes in the electrical audio path that may affect the frequency response of the system. A spectral correction factor may be determined by comparing the average speech spectral shape estimate to a desired target spectral shape. The spectral correction factor may be added (in units of dB) to the spectrum of the input speech signal in order to enhance or adjust the spectrum of the input speech signal toward the desired spectral shape, and an enhanced speech signal re-synthesized from the corrected spectrum.

Abstract: An active noise cancellation system reduces, at a listening position, the power of a noise signal being radiated from a noise source to the listening position. The system includes an adaptive filter that receives a reference signal representing the noise signal, and provides a compensation signal. A bass management unit receives the compensation signal and applies a phase shift to the compensation signal to provide a phase shifted compensation signal. A first acoustic radiator receives the phase shifted compensation signal and radiates audio indicative thereof to the listening position. A second acoustic radiator receives the compensation signal and radiates audio indicative thereof to the listening position. The transfer function characteristics from the input of the bass management system to the listening position approximately matches a desired transfer function.

Abstract: Method and apparatus for entrainment containment in digital filters using output phase modulation. Phase change is gradually introduced into the acoustic feedback canceller loop to avoid entrainment of the feedback canceller filter. Various embodiments employing different output phase modulation approaches are set forth and time and frequency domain examples are provided. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: A method for obtaining coefficients for a non-adaptive controller for use in an active noise control system, the method comprising capturing experimental data from a target system to characterise the plant of the target system; and implementing an adaptive algorithm to compute the coefficients in dependence on the experimental data.

Abstract: A signal-processing circuit for the generation of a loudspeaker signal may include an input for the feeding of a digital input signal and a digital equalizer filter that is coupled with the input and has at least one first recursive filter that is defined by a first adjustable set of coefficients, an amplification device, a current measurement device, a digital filter block that may be configured to generate, through filtering, an estimate signal as a function of the measurement signal, and a matching block that may be configured to determine a new set of coefficients as a function of the measurement signal, the estimate signal, and the filtered input signal.

Abstract: Different sampling rates between a playout unit and a capture unit are compensated for via a system, method and computer program product. The playout unit receives samples from a computational unit, and the capture unit sends samples to the computational unit. A playout FIFO buffer operates in a playout time domain, and a capture FIFO buffer operates in a capture time domain. The computational unit is synchronized to a common clock. A first relationship is calculated between the common clock and a playout fifo buffer read pointer, and a second relationship is calculated between the common clock and a capture FIFO buffer write pointer. For each sample in the playout time domain a corresponding sample in the samples from said computational unit is found and sent to the playout FIFO buffer. For each sample in the common clock time domain the corresponding sample in the capture time domain is found and sent to the computational unit.

Abstract: Various embodiments reduce noise within a particular environment, while isolating and capturing speech in a manner that allows operation within an otherwise noisy environment. In one embodiment, an array of one or more microphones is used to selectively eliminate noise emanating from known, generally fixed locations, and pass signals from a pre-specified region or regions with reduced distortion.

Abstract: Method and apparatus for signal processing an input signal in a hearing assistance device to avoid entrainment, the hearing assistance device including a receiver and a microphone, the system comprising using a gradient adaptive lattice filter including one or more reflection coefficients to measure an acoustic feedback path from the receiver to the microphone of the hearing assistance device.

Abstract: Disclosed herein is a digital filter circuit for producing a noise reduction signal for reducing noise based on a noise signal outputted from a microphone which collects the noise, including: an analog/digital conversion section; a first digital filter section; an arithmetic operation processing section; a second digital filter section; and a digital/analog conversion section. The first digital filter section and/or the second digital filter section are configured such that a predetermined attenuation amount is obtained within a predetermined range in the proximity of a sampling frequency around the sampling frequency.

Abstract: Embodiments of the present disclosure generally include a method and apparatus for minimizing acoustic echo in video conference. In one embodiment, a computer implemented method for minimizing acoustic echo in video conferencing comprises determining position information wherein the position information is derivable from a video stream and processing the position information using a computer to identify at least one direction wherein the one direction minimizes acoustic echo.

Abstract: A time-domain method of adaptively levelling the loudness of a digital audio signal is proposed. It selects a proper frequency weighting curve to relate the volume level to the human auditory system. The audio signal is segmented into frames of a suitable duration for content analysis. Each frame is classified to one of several predefined states and events of perceptual interest is detected. Four quantities are updated each frame according to the classified state and detected event to keep track of the signal. One quantity measures the long-term loudness and is the main criterion for state classification of a frame. The second quantity is the short-term loudness that is mainly used for deriving the target gain. The third quantity measures the low-level loudness when the signal is deemed to not contain important content, giving a reasonable estimate of noise floor. A fourth quantity measures the peak loudness level that is used to simulate the temporal masking effect.

Abstract: A noise control device eliminates a possibility that a noise arriving at a control point is increased as compared to a case where noise control is not performed and has a reduced circuit scale. The device includes a controlling noise detector for detecting a given noise to output a controlling noise signal, a controlling filter section for signal-processing the controlling noise signal, by using a preset fixed filter coefficient, to output a control signal, and a control speaker for reducing the given noise, by emitting toward the control point a control sound based on the control signal. The device also includes a noise determination section for determining whether the given noise is a noise corresponding to the fixed filter coefficient, and an output control section for stopping output of the control signal from the controlling filter section when the given noise does not correspond.

Abstract: An adaptive gain control system and related operating method for digital audio samples is provided. The method is suitable for use with a digital media encoding system that transmits encoded media streams to a remotely-located presentation device such as a media player. The method begins by initializing the processing of a media stream. Then, the method adjusts the gain of a first set of digital audio samples in the media stream using a fast gain adaptation scheme, resulting in a first group of gain-adjusted digital audio samples having normalized volume during presentation. The method continues by adjusting the gain of a second set of digital audio samples in the media stream using a steady state gain adaptation scheme that is different than the fast gain adaptation scheme, resulting in a second group of gain-adjusted digital audio samples having normalized volume during presentation.

Abstract: A method and a system for the active reduction, at a predetermined area, of the energy of a sound signal (dk(n)), also called a diffused noise signal, generated at the area by a primary signal (xk(n)), or noise signal, by the emission of a plurality of counter-noise signals (yk(n)) having an effect antagonistic to the diffused noise signal (dk(n)), each of the counter-noise signals (yk(n)) including a feedback counter-noise signal (yfbkk(n)) and a feed-forward counter-noise signal (yfwdk(n)). The method includes detecting the periodical components of diffused noise signal (dk(n)) for adjusting the feedback counter-noise signal (yfbkk(n)), and modelling the inverse of the secondary path for adjusting the feedback counter-noise (yfbkk(n)) and feed-forward counter-noise (yfwdk(n)) signals. The invention can be implemented to any type of industrial or non-industrial noise and in any location such as working places and relaxation places.