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: In accordance with methods and systems of the present disclosure, an integrated circuit for implementing at least a portion of a personal audio device may include an output including an anti-noise signal, a reference microphone input, an error microphone input, and a processing circuit. The processing circuit may implement an adaptive filter having a response that generates the anti-noise signal from the reference microphone signal to reduce the presence of the ambient audio sounds heard by the listener, wherein the processing circuit may implement a coefficient control block that shapes the response of the adaptive filter in conformity with the error microphone signal and the reference microphone signal by adapting the response of the adaptive filter in accordance with a calculated narrow-band-to-full-band ratio, wherein the narrow-band-to-full-band ratio is a function of a narrow-band power of the reference microphone signal divided by a full-band power of the reference microphone signal.

Abstract: A personal audio device, such as a wireless telephone, includes an adaptive noise canceling (ANC) circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone is also provided proximate the speaker to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit determines a degree of coupling between the user's ear and the transducer and adjusts the adaptive cancellation of the ambient sounds to prevent erroneous and possibly disruptive generation of the anti-noise signal if the degree of coupling lies either below or above a range of normal operating ear contact pressure.

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: A system for actively reducing noise at a listening point, includes an earphone housing, a transmitting transducer, a receiving transducer and a controller. The transmitting transducer converts a first electric signal into a first acoustic signal, and radiates the first acoustic signal along a first acoustic path having a first transfer characteristic and along a second acoustic path having a second transfer characteristic. The receiving transducer converts the first acoustic signal and ambient noise into a second electrical signal. The controller compensates for the ambient noise by providing a noise reducing electrical signal to the transmitting transducer. The noise reducing electrical signal is derived from a filtered electrical signal that is provided by filtering the second electrical signal with a third transfer characteristic. The second and the third transfer characteristics together model the first transfer characteristic.

Abstract: An audio processor has an active noise cancellation (ANC) processor to compute an S-filter, which estimates a transfer function from an earpiece speaker, that is to be worn by a user, to an error microphone. The ANC processor adaptively produces an anti noise signal for reducing ambient noise that can be heard by the user. An adaptive equalization (EQ) processor has an EQ filter controller that adjusts an EQ filter, while audio user content is being filtered in accordance with the EQ filter before being fed to the speaker and then heard by the user. Other embodiments are also described and claimed.

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: In accordance with systems and methods of this disclosure, a method may include generating a feedforward anti-noise signal component from a result of measuring with the reference microphone countering the effects of ambient audio sounds at an acoustic output of a transducer by filtering an output of the reference microphone, adaptively generating a feedback anti-noise signal component from a result of measuring with an error microphone for countering the effects of ambient audio sounds at the acoustic output of the transducer by adapting a response of a feedback adaptive filter that filters a synthesized reference feedback to minimize the ambient audio sounds in the error microphone signal, wherein the synthesized reference feedback is based on a difference between the error microphone signal and the feedback anti-noise signal component.

Abstract: A processing circuit may comprise an adaptive filter having a response generating an anti-noise signal from a reference microphone signal, a secondary path estimate filter modeling an electro-acoustic path of a source audio signal, a biasing portion that generates a scaled anti-noise signal by applying a scaling factor and the response of the secondary path estimate filter to the anti-noise signal, and a coefficient control block that shapes the response of the adaptive filter in conformity with the reference microphone signal and a modified playback corrected error signal by adapting the response of the adaptive filter to minimize ambient audio sounds in the error microphone signal, wherein the playback corrected error is based on a difference between the error microphone signal and the source audio signal and the modified playback corrected error signal is based on a difference between the playback corrected error signal and the scaled anti-noise signal.

Abstract: In accordance with methods and systems of the present disclosure, a processing circuit may implement at least one of: a feedback filter having a response that generates at least a portion of an anti-noise component from a playback corrected error, the playback corrected error based on a difference between the error microphone signal and a secondary path estimate; and a feedforward filter having a response that generates at least a portion of the anti-noise signal from a reference microphone signal. The processing circuit may also implement a secondary path estimate filter configured to model an electro-acoustic path of a source audio signal and have a response that generates a secondary path estimate from the source audio signal and a secondary path estimate performance monitor for monitoring performance of the secondary path estimate filter in modeling the electro-acoustic path.

Abstract: A system to eliminate acoustic noise caused by a first MLCC (Multi-Layer Ceramic Capacitor) array positioned on a PCB (printed circuit board) is disclosed. The first MLCC array generates a first vibration responsible for the acoustic noise in response to receiving a varying input voltage. A third MLCC array senses the first vibration and generates a feedback signal. An adaptive filter then uses the feedback signal to generate an output signal that is used by a second MLCC to generate a second vibration that acts as a counter to dampen the first vibration. Because the input voltage signal is varying in time, the adaptive filter continually samples the varying input voltage and the feedback signal to generate the output signal that minimizes the acoustic noise. The second and third MLCC arrays are selectively positioned and oriented on the PCB for optimum performance.

Abstract: In accordance with method and systems of the present disclosure, a processing circuit may implement an adaptive filter having a response that generates the anti-noise signal from the reference microphone signal to reduce the presence of the ambient audio sounds heard by the listener, a coefficient control block that shapes the response of the adaptive filter in conformity with the error microphone signal and the reference microphone signal by adapting the response of the adaptive filter to minimize the ambient audio sounds in the error microphone signal, and a coefficient bias control block which biases coefficients of the coefficient control block towards zero in a range of frequencies outside of a frequency response of the source audio signal.

Abstract: An earpiece for providing an audio signal to two ears of a user, includes an audio interface for providing the audio signal, a bone conduction speaker coupled to the audio interface for outputting an output signal, an adaptive filter with an adjustable impulse response to provide a filtered signal that once played to the second ear of a user, by an in-ear speaker, reduces an audio effect of the leakage signal on the second ear of the user, an in-ear speaker coupled to the audio interface for providing an in-ear audio signal to the second ear of the user, in response to the filtered signal and the second input signal, an in-ear microphone for sensing a sensed audio signal, and a processor for updating an impulse response of the adaptive filter in order to reduce a contribution of the leakage signal to the sensed audio signal.

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: Systems and methods described herein provide for low latency active noise cancellation which alleviate the problems associated with analog filter circuitry. The present technology utilizes low latency digital signal processing techniques which overcome the high latency conventionally associated with conversion between the analog and digital domains. As a result, low latency active noise cancellation is performed utilizing digital filter circuitry which is not subject to the inaccuracies and drift of analog filter components. In doing so, the present technology provides robust, high quality active noise cancellation.

Abstract: Techniques described herein are directed to the enhancement of spectral features of an audio signal via adaptive modulation filtering. The adaptive modulation filtering process is based on observed modulation envelope autocorrelation coefficients obtained from the audio signal. The modulation envelope autocorrelation coefficients are used to determine parameters of an adaptive filter configured to filter the spectral features of the audio signal to provide filtered spectral features. The parameters are updated based on the observed modulation envelope autocorrelation coefficients to adapt to changing acoustic conditions, such as signal-to-noise ratio (SNR) or reverberation time. Accordingly, such acoustic conditions are not required to be estimated explicitly. Techniques described herein also allow for the estimation of useful side information, e.g.

Abstract: Health-sensing and health-action devices and systems are generally described. The health-sensing device may include one or more of a sensor, a filter, and a transmitter. The sensor may be configured to sense one or more factors relating to an indicator of a health related condition or occurrence such as snoring and may include one or more microphone devices, accelerometers, and/or MEMs devices. The filter may be configured to evaluate a signal from the sensor and determine if the indicator has been detected. The transmitter may be arranged for initiating a transmission based on a signal from the filter. The health-action device may be configured for responding to an indicator of a health related condition or occurrence of a user and may include one or more of a receiver, a processor, and a responder. The health-action device may stimulate the user or may cancel the snoring sound.

Abstract: Apparatus and methods for audio noise attenuation are disclosed. An audio signal analyzer can determine whether an input audio signal received from a microphone device includes a noise signal having identifiable content. If there is a noise signal having identifiable content, a content source is accessed to obtain a copy of the noise signal. An audio canceller can generate a processed audio signal, having an attenuated noise signal, based on comparing the copy of the noise signal to the input audio signal. Additionally or alternatively, data may be communicated on a communication channel to a separate media device to receive at least a portion of the copy of the noise signal from the separate media device, or to receive content-identification data corresponding to the content source.

Abstract: A noise cancellation device (NCD) comprises a microphone, an analog-to-digital converter, a digital-to-analog converter, a rechargeable battery, and a processor. The NCD acquires an audio input, from an external device such as a stethoscope or a cell phone, and passes the analog data into an ADC (analog-to-digital converter) for signal conversion. The digitized signals are then passed to the processor for further processing. The processor contains all the processing functions such as preprocessing (divide the input data into frames and apply shaping function to each frame), short term Fourier transform (STFT), adaptive filtering, inverse STFT, and signal synthesis.

Abstract: An adaptive active noise cancellation apparatus performs a filtering operation in a first digital domain and performs adaptation of the filtering operation in a second digital domain.

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: 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 apparatus generally having a first circuit and a second circuit is disclosed. The first circuit may be configured to generate a first sample by filtering an input vector based on (a) a filter vector and (b) a stochastic vector. Each of a plurality of components in the stochastic vector generally has a respective random value. The first circuit may also be configured to generate a second sample as a difference between a third sample and the first sample. The third sample may be received from a network as an echo. The second circuit may be configured to update a subset of a plurality of taps of the filtering where a corresponding one of the components of the stochastic vector has a first value of the random values.

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: 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.

Abstract: An adaptive mode control apparatus and method for adaptive beamforming based on detection of a user direction sound are provided. The adaptive mode control apparatus includes a signal intensity detector that searches for signal intensity of each designated direction to detect signal intensity having a maximum value when a voice signal of each direction is input through at least one microphone; and an adaptive mode controller that compares the signal intensity having the maximum value detected through the signal intensity detector with a threshold value and determines whether to perform an adaptive mode of a Generalized Sidelobe Canceller (GSC) according to the comparison results. Therefore, a lack of control of adaptation of an adaptive filter of the conventional art is solved. That is, as one condition for guaranteeing performance of adaptive beamforming, adaptation of an adaptive filter is not performed when noise of a sound with a high autocorrelation is cancelled.

Abstract: Disclosed is a steerable sensor array that receives input from a target and applies an averaging filter. An adaptive filter is then used if the SNR of the output of the averaging filter reaches a threshold.

Abstract: Various embodiments of the present invention provide systems and methods for data processing. As an example, a data processing circuit is discussed that includes: a noise predictive filter circuit, a mean calculation circuit, a summation circuit, and a filter tap adaptation circuit. The noise predictive filter circuit is operable to perform a noise predictive filtering process on a data input based upon a selector input to yield a noise predictive output. The selector input is derived from a data detector output. The mean calculation circuit is operable to average two or more instances of the noise predictive output to yield a mean output. The summation circuit is operable to subtract the mean output from the noise predictive output to yield a sum output. The filter tap adaptation circuit is operable to adaptively calculate a filter tap based at least in part on a value derived from the data input and a value derived from the sum output.

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

Abstract: An audio processing device comprises a feedback estimation system for estimating feedback from an output transducer to an input transducer, the feedback estimation system comprising an adaptive filter comprising a variable filter part for filtering an input signal according to variable filter coefficients and an algorithm part comprising an adaptive algorithm for dynamically updating filter coefficients, a control unit for controlling the de-correlation unit and the adaptive algorithm, and a correlation detection unit for determining a) the auto-correlation of a signal of the forward path and providing an AC-value and/or b) the cross-correlation between two different signals of the forward path and providing an XC-value.

Abstract: An open air, wide area noise cancellation system and method provides improved identification and characterization of noise sources then generating noise cancelling sound waves.

Abstract: An active vibration noise control apparatus with which vibration noise can be reduced stably and effectively by preventing divergence of a filter coefficient or a delay in convergence due to an effect of a transfer characteristic on a secondary path. The active vibration noise control apparatus according to the present invention calculates an update step size in accordance with a magnitude of a gain in the transfer characteristic corresponding to a frequency of vibration noise, where the transfer characteristic is on the secondary path for propagation of a secondary vibration noise for reducing the vibration noise, and updates the filter coefficient on the basis of the calculated update step size.

Abstract: An active noise cancellation controller for performing noise attenuation in a system over a predetermined frequency rang. The controller comprises several components: a first input for a reference signal indicative of a noise level; a second input for receiving an error signal indicative of a remnant noise level; an output for providing a noise cancellation signal; a fixed feedback controller for processing the error signal; a fixed feed-forward controller for processing the reference signal; and an adaptive feed-forward controller having a digital adaptive finite impulse response filter arranged for operation on the reference signal the error signal.

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 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: 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: A hearing aid system and a method of estimating ambient noise in a listening device includes an input transducer and an output transducer, an electrical forward path between the input transducer and the output transducer providing a forward gain, an electrical feedback path comprising an adaptive filter for estimating the acoustic feedback gain from the output transducer to the input transducer. A method determines the quality of a critical gain measurement for a listening device. The method comprises a) monitoring the energy of the first-difference of the filter coefficients of the adaptive filter over time and b) applying a predefined threshold criterion to the change in energy content from one time instance to another to determine an acceptable impact of the ambient noise. This technique may e.g. be used for the fitting of hearing instruments where background noise is variable.

Abstract: A hearing aid system includes an input transducer, an output transducer, and an electrical signal path between them. The signal path includes a signal processing unit. An electrical feedback cancellation path is formed between the output side and the input side of the signal path and subtracts an estimate of acoustical feedback from a signal on the input side of the amplifier part, the electrical feedback cancellation path including an adaptive filter for providing a variable filtering function. A probe signal generator generates a probe signal characterizing an acoustical feedback path, the probe signal being fed to the adaptive filter of the electrical feedback cancellation path. A second input transducer is located at a position where the acoustical signal is substantially free from acoustic feedback from the output transducer, and the adaptive filter uses a signal derived from the second electrical input signal to influence its filtering function.

Abstract: In one aspect, multiple adaptive W filters and associated adaptive filter controllers are provided that use multiple reference microphone signals to produce multiple, “component” anti-noise signals. These are gain weighted and summed to produce a single anti-noise signal, which drives an earpiece speaker. The weighting changes based on computed measures of the coherence between content in each reference signal and content in an error signal. Other embodiments are also described and claimed.

Abstract: A method of signal processing an input signal in a hearing aid to avoid entrainment, the hearing aid including a receiver and a microphone, the method comprising using an adaptive filter to measure an acoustic feedback path from the receiver to the microphone and adjusting an adaptation rate of the adaptive filter using an output from a filter having an autoregressive portion, the output derived at least in part from a ratio of a predictive estimate of the input signal to a difference of the predictive estimate and the input signal.

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

Abstract: A howling suppression device includes: a subtractor which generates an error signal; an adaptive filter which applies filtering to the error signal; and a coefficient update control unit which controls an update rate of a filter coefficient and includes: a convergence analysis unit which determines whether or not a condition that the degree of convergence of filter characteristics is higher than a criterion value is satisfied; a change amount analysis unit which determines whether or not a condition that the degree of convergence has turned out to be higher is satisfied; and an update rate control unit which sets the update rate to a first rate when both the conditions are satisfied and to a second rate when at least one of the conditions is not satisfied. The adaptive filter updates the filter coefficient at the update rate.

Abstract: An active noise control (ANC) processor has an adaptive filter that uses a reference signal to produce an anti-noise signal, and an error signal to evaluate cancellation performance. An adaptive filter algorithm engine configures the filter coefficients of the adaptive filter, in accordance with pre-shaped versions of the error and reference signals. The pre-shaping filter has a high-pass transfer function and enables the adaptive algorithm engine to increase noise cancellation performance in a high frequency band during the presence of focused or narrow-band noise energy in a low frequency band. Other embodiments are also described and claimed.

Abstract: A system for managing the changing state of an adaptive filter in an active noise control (ANC) system is described. An adaptive filter state storage stores copies of prior states of the adaptive filter. A disturbance detector can detect either normal ambient noise or abnormal ambient noise. An adaptive filter state manager signals that a copy of a current state of the adaptive filter is to be repeatedly written to the state storage, so long as normal ambient noise is being detected. But when abnormal noise is detected, the state manager signals that the adaptive filter be restored to one of its prior states, from the copies stored in the state storage. Other embodiments are also described and claimed.

Abstract: A method comprises processing M subband communication signals and N target-cancelled signals in each subband with a set of beamformer coefficients to obtain an inverse target-cancelled covariance matrix of order N in each band; using a target absence signal to obtain an initial estimate of the noise power in a beamformer output signal averaged over recent frames with target absence in each subband; multiplying the initial noise estimate with a noise correction factor to obtain a refined estimate of the power of the beamformer output noise signal component in each subband; processing the refined estimate with the magnitude of the beamformer output to obtain a postfilter gain value in each subband; processing the beamformer output signal with the postfilter gain value to obtain a postfilter output signal in each subband; and processing the postfilter output subband signals to obtain an enhanced beamformed output signal.

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: An active noise cancellation system includes an adaptive filter, a signal source, an acoustic actuator, a microphone, a secondary path and an estimation unit. The adaptive filter receives a reference signal representing noise, and provides a compensation signal in response to the received reference signal. The signal source provides a measurement signal. The acoustic actuator radiates the compensation signal and the measurement signal to the listening position. The microphone receives a first signal that is a superposition of the radiated compensation signal, the radiated measurement signal, and the noise signal at the listening position, and provides a microphone signal in response to the received first signal. The secondary path includes a secondary path system that represents a signal transmission path between an output of the adaptive filter and an output of the microphone.

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: 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 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.