Abstract: A vehicle includes an engine and/or powertrain producing noise that is audible in a passenger compartment of the vehicle when the engine and/or powertrain is running. An active noise control arrangement includes a first loudspeaker disposed within a passenger compartment of the vehicle. A digital signal processor receives audio data and transmits an audio signal to the first loudspeaker dependent upon the audio data. A microphone is disposed within the passenger compartment and converts the sound from the first loudspeaker and the noise within the passenger compartment into a microphone signal. The microphone signal is transmitted to the digital signal processor, and the digital signal processor modifies the audio signal such that the audio signal attenuates the noise in the passenger compartment.

Abstract: A system and method for active noise compensation in a vehicle is provided. The system has at least one apparatus for detecting a noise, a computer unit for calculating a compensation sound, at least one apparatus for generating a compensation sound which interferes with the noise, and a device for detecting at least one driver-specific parameter. The compensation sound is calculated depending on the at least one driver-specific parameter. Also provided is a motorcycle having a system for active noise compensation.

Abstract: A method and a system for calibrating a surround sound system are disclosed. The calibration system can provide a graphical user interface for display comprising a visual representation of the room hosting a multichannel surround sound system. The graphical user interface can permit user input of gestures to place or make changes to the placement of icons representing one or more loudspeakers and a listener. The calibration system can estimate the positions of the one or more loudspeakers or the listener based on the placement of the icons in the model room. A spatial calibration based on the estimated positions can then be performed such that the multichannel surround sound system can render sound scenes more accurately.

Abstract: The disclosure includes a headset comprising one or more earphones including one or more sensing components. The headset also includes one or more voice microphones to record a voice signal for voice transmission. The headset also includes a signal processor coupled to the earphones and the voice microphones. The signal processor is configured to employ the sensing components to determine a wearing position of the headset. The signal processor then selects a signal model for noise cancellation. The signal model is selected from a plurality of signal models based on the determined wearing position. The signal processor also applies the selected signal model to mitigate noise from the voice signal prior to voice transmission.

Abstract: Multi-sensor signal optimization is facilitated for speech communication. A sensor component including acoustic sensors can be configured to detect sound and generate, based on the sound, first sound information associated with a first sensor of the acoustic sensors and second sound information associated with a second sensor of the acoustic sensors. Further, an audio processing component can be configured to generate filtered sound information based on the first sound information, the second sound information, and a spatial filter associated with the acoustic sensors, determine noise levels for the first sound information, the second sound information, and the filtered sound information, and generate output sound information based on a selection of one of the noise levels or a weighted combination of the noise levels.

Abstract: Methods, apparatus, systems and articles of manufacture to implement selective suppression of audio emitted from an audio source are disclosed. Example methods disclosed herein include obtaining, at a portable device, data transmitted at a first time by a transmitter associated with an audio source, the data including reference audio data corresponding to a first audio signal to be output by the audio source at a second time after the first time, the data further specifying a time delay corresponding to a difference between the second time and the first time. Disclosed example methods also include generating, based on the reference audio data and the time delay, a suppression audio signal to provide to an audio output driver in communication with a speaker of the portable device to suppress the first audio signal when the first audio signal is output by the audio source at the second time.

Abstract: Method and system for active reduction of a predefined audio acoustic signal (AAAS), also referred to as “noise”, in a quiet zone, without interfering undefined acoustic noise signals within as well as outside the quiet zone, by generating accurate antiphase AAAS signal. The accuracy of the generated antiphase AAAS is obtained by employing a unique synchronization signal(s) (SYNC) which is generated and combined with the predefined AAAS. The combined signal is electrically transmitted (referred to as the “electric channel”) to a processing “quieting component”. Simultaneously, the generated SYNC signal is acoustically broadcasted near the predefined AAAS and merges with it. A microphone in the quiet zone receives the merged acoustic signals that arrive via the air (referred to as the “acoustical channel”) to the quiet zone and a receiver in the quieting component receives the combined electrical AAAS and SYNC signal that arrive wire or wireless to the quiet zone.

Abstract: A digital circuit arrangement for an ambient noise-reduction system affording a higher degree of noise reduction than has hitherto been possible. The arrangement converts the analog signals into N-bit digital signals at sample rate f0, and then subjects the converted signals to digital filtering. The value of N in some embodiments is 1 but, in any event, is no greater than 8, and f0 may be 64 times the Nyquist sampling rate but, in any event, is substantially greater than the Nyquist sampling rate. This permits digital processing to be used without incurring group delay problems that rule out the use of conventional digital processing in this context. Furthermore, adjustment of the group delay can readily be achieved, in units of a fraction of a micro-second, providing the ability to fine tune the group delay for feed forward applications.

Abstract: The present disclosure discloses a system for attenuating sound produced by a vehicle. The system comprises a navigation device associated with the vehicle for determining location details of the vehicle, a sound reduction unit provisioned in an exhaust assembly of the vehicle. The system further includes an Electronic Control Unit (ECU) of the vehicle communicatively coupled to the navigation device and the sound reduction unit. The ECU is configured to detect sound reduction location, by comparing the location details with a pre-defined location data and operate the sound reduction unit to attenuate sound produced by the vehicle when the sound reduction location is detected. The system of the present disclosure, attenuates sound intensity of the vehicle to a desired level by considering surrounding conditions of the vehicle, thus the vehicle noise may be automatically attenuated in the sound sensitive areas or locations.

Abstract: A system and method receives one or more captured signals through a captured audio path and produces one or more playback signals through a playback audio path. The system and method executes one or more signal processing functions and measures the delays within the playback audio path and captured audio path during operation of the one or more signal processing functions. The system and method stores the measured delays in a memory and compensates the one or more signal processing functions for the playback delay and the capture delay.

Abstract: A voice input/output device is connected to a smart device having a voice signal input/output function is provided. A digital voice signal output by the smart device is output to an output unit, (e.g., ear receiver or a speaker), in the form of a voice, and a voice signal transmitted by the smart device is extracted and input to the smart device as a digital voice signal to generate a sound by the smart device itself in addition to an externally generated sound that is input again with improved quality. Accordingly, a voice signal output by the smart device is extracted using a earphone type device having microphone mounted thereon and the voice signal is input to the smart device again.

Abstract: Personal audio systems and methods are disclosed. A personal audio system includes a processor to generate a personal audio stream by processing an ambient audio stream in accordance with an active processing parameter set, a circular buffer memory to store a most recent snippet of the ambient audio stream, and an event detector to detect a trigger event. In response to detection of the trigger event, a controller may extract audio feature data from the most recent snippet of the ambient audio stream and transmit the audio feature data and associated metadata to a knowledgebase remote from the personal audio system.

Abstract: A signal compensation method and an electronic device adapted to the method are provided. The signal compensation method includes storing filter data related to a hearing aid in a storage unit, recording a first sound signal, and compensating for the first sound signal, based on the filter data.

Abstract: Variable sound decomposition masking techniques are described. In one or more implementations, a mask is generated that incorporates a user input as part of the mask, the user input is usable at least in part to define a threshold that is variable based on the user input and configured for use in performing a sound decomposition process. The sound decomposition process is performed using the mask to assign portions of sound data to respective ones of a plurality of sources of the sound data.

Abstract: The present disclosure relates to a noise reduction device and a noise reduction method capable of reducing noise in a more stable and effective manner. Signal processing is performed to generate a reference signal representing a waveform of noise. Moreover, signal processing is performed for an error signal representing a waveform of an error measured by a microphone in accordance with an amplitude-frequency characteristic. Then, a filter coefficient with which the error signal becomes zero is calculated under adaptive algorithm with reference to the reference signal. The reference signal is filtered by using the filter coefficient to obtain a control signal. The control signal is supplied to the corresponding one of the predetermined number of output units. The present technology is applicable to a noise cancelling system equipped in a closed space such as an interior of a vehicle, for example.

Abstract: Data discovery for sensor data in an M2M network uses probabilistic models, such as Gaussian Mixing Models (GMMs) to represent attributes of the sensor data. The parameters of the probabilistic models can be provided to a discovery server (DS) that respond to queries concerning the sensor data. Since the parameters are compressed compared to the attributes of the sensor data itself, this can simplify the distribution of discovery data. A hierarchical arrangement of discovery servers can also be used with multiple levels of discovery servers where higher level discovery servers using more generic probabilistic models.

Abstract: An apparatus comprising: at least one first input configured to receive at least one first sensor signal, the at least one first sensor signal having a first modality; at least one second input configured to receive at least one second sensor signal, the at least one second sensor signal having a second modality different from the first modality; and a codec configured to generate at least one meta signal comprising and stored using separate resources the at least one first sensor signal and the at least one second sensor signal.

Abstract: The disclosure relates to combined active noise cancellation and noise compensation in a headphone. An audio processing device includes a selector and a noise compensation unit. The selector can select one of a plurality of first transfer functions based on at least one feature of at least one of an external noise and a content audio signal representing a sound to be reproduced through the headphone. The noise compensation unit can compute a second audio signal by applying the selected first transfer function to a first audio signal, and derive gains for the noise compensation at least based on the second audio signal. The at least one feature can be used to distinguish at least two of the first transfer functions. Each of the first transfer functions can transform the first audio signal to the second audio signal which is assumed as representing a version of the sound represented by the first audio signal, which arrives at an eardrum of a listener wearing the headphone.

Abstract: Disclosed is a method for optimizing noise cancellation in a headset, the headset comprising a headphone and a microphone unit comprising at least a first microphone and a second microphone, the method comprising: generating at least a first audio signal from the at least first microphone, where the first audio signal comprises a speech portion from a user of the headset and a noise portion from the surroundings; generating at least a second audio signal from the at least second microphone, where the second audio signal comprises a speech portion from the user of the headset and a noise portion from the surroundings; generating a noise cancelled output by filtering and summing at least a part of the first audio signal and at least a part of the second audio signal, where the filtering is adaptively configured to continually minimize the power of the noise cancelled output, and where the filtering is adaptively configured to continually provide that at least the amplitude spectrum of the speech portion of

Abstract: A Point-Of-Sale (POS) device detects vibrations produced from equipment in its operational environment. The vibrations are identified and one or more filters produced to remove the vibrations when the POS device is used to improve the accuracy of the POS device.

Abstract: The specification and drawings present a use of multiple microphones for increasing acoustic sensing capabilities by processing acoustic signals from the multiple microphones in outdoor luminaire mounted surveillance/sensor systems. For example, various embodiments presented herein describe signal processing means to utilize stereo/multiple microphones in a luminaire (such as an outdoor roadway luminaire) to provide enhanced information regarding the surroundings of the luminaire. The multiple microphone luminaire sensor processing system can provide a more environmentally robust and sensitive approach which can be, for example, resistant to environmental noise such as a wind noise, as well as capable of isolating specific sounds from the surroundings, e.g., in specific directions.

Abstract: A vehicle includes an engine and/or powertrain producing noise that is audible in a passenger compartment of the vehicle when the engine and/or powertrain is running. An active noise control arrangement includes a first loudspeaker disposed within a passenger compartment of the vehicle. A digital signal processor receives audio data and transmits an audio signal to the first loudspeaker dependent upon the audio data. A microphone is disposed within the passenger compartment and converts the sound from the first loudspeaker and the noise within the passenger compartment into a microphone signal. The microphone signal is transmitted to the digital signal processor, and the digital signal processor modifies the audio signal such that the audio signal attenuates the noise in the passenger compartment.

Abstract: A microphone array device includes a first sound reception unit configured to obtain a first sound signal that is input from a first microphone, a second sound reception unit configured to obtain a second sound signal that is input from a second microphone, a noise state evaluation unit configured to compare the first sound signal and the second sound signal and to obtain an evaluation parameter to evaluate an influence of a non-target sound included in the second sound signal on a target sound included in the first sound signal according to a result of the comparison, a subtraction adjustment unit configured to set a suppression amount for the second sound signal based on the evaluation parameter and to generate a third sound signal; and a subtraction unit configured to generate a signal to be output based on the third sound signal and the first sound signal.

Abstract: [Object] To provide a signal processing apparatus which can allow a user to listen to sound assumed by a designer while a difference among individuals is taken into account. [Solution] Provided is a signal processing apparatus including: a characteristics difference calculating unit configured to calculate a difference between reproduction target characteristics of a first sound signal and characteristics of a second sound signal obtained by collecting, at a microphone provided inside a headphone, sound output from a driver which outputs the sound based on a third sound signal obtained by performing signal processing on the first sound signal; and a sound signal processing unit configured to select a parameter to be used at the signal processing based on the difference calculated by the characteristics difference calculating unit, and perform signal processing on the first sound signal.

Abstract: Examples described herein involve identifying one or more error conditions during calibration of one or more playback devices in a playback environment. A microphone of a network device may detect and sample an audio signal while the one or more playback devices in the playback environment plays a calibration tone. A processor of the network device may then receive, from the microphone, a stream of audio data. The audio data may include an audio signal component and a background noise component. As a subset of the audio data is received, the processor may identify based on at least the subset of audio data, the one or more error conditions. The processor may then cause a graphical display to display a graphical representation associated with the identified error condition.

Abstract: A crosstalk canceller for reducing acoustic crosstalk at a time of audio playback is derived by forming a channel frequency response for a nominated playback geometry, and decomposing the channel frequency response to derive a decomposition element such as a singular value decomposition matrix. A value of the decomposition element, such as the smallest singular value, is then adjusted to reduce spectral coloration, and filter coefficients of a crosstalk cancellation filter are derived from the adjusted decomposition element.

Abstract: A method may include determining a desired speech estimate originating from a speech acceptance direction range while reducing a level of interfering noise, determining an interfering noise estimate originating from a noise rejection direction range while reducing a level of desired speech, calculating a ratio of the desired speech estimate to the interfering noise estimate, dynamically computing a set of thresholds based on the speech acceptance direction range, noise rejection direction range, a background noise level, and a noise type, estimating a power spectral density of background noise arriving from the noise rejection direction range, calculating a frequency-dependent gain function based on the power spectral density of background noise and thresholds, and applying the frequency-dependent gain function to at least one microphone signal generated by the plurality of microphones to reduce noise arriving from the noise rejection direction while preserving desired speech arriving from the speech acceptan

Abstract: A sound control module is configured to determine N magnitudes for outputting a predetermined engine sound at N predetermined harmonics of a base frequency, respectively, where N is an integer greater than one. An adjustment module is configured to determine N magnitude adjustments for the N predetermined harmonics, respectively, based on a position of a compression ratio (CR) actuator configured to vary a CR of an engine. The sound control module is further configured to determine N adjusted magnitudes for the N predetermined harmonics based on: the N magnitudes for the N predetermined harmonics, respectively; and the N magnitude adjustments for the N predetermined harmonics, respectively. An audio driver module is configured to apply power to at least one speaker of the vehicle and output the predetermined engine sound based on the N adjusted magnitudes at N frequencies, respectively, corresponding to the N predetermined harmonics of the base frequency.

Abstract: A computing device senses speech or other user input. One or more physical variables pertaining to a user of the computing device are also sensed, and respective signals are analyzed or compared to the user input. The analysis determines if the user input is likely that of an authorized user, and assigns a confidence metric to that determination. The computing device may then perform actions corresponding to the speech or user input content in accordance with the determination.

Abstract: Methods, apparatus, systems and articles of manufacture to implement selective suppression of audio emitted from an audio source are disclosed. Example apparatus disclosed herein include a speaker, an audio output driver in communication with the speaker, memory including computer readable instructions, and a processor. In some disclosed example apparatus, the processor is to execute the instructions to perform operations including obtaining reference audio data corresponding to an audio signal to be output subsequently by the audio source and including identification data identifying the audio source, and in response to an input selection corresponding to the identification data, generating a suppression signal to provide to the audio output driver, the suppression signal based on the reference audio data and a time delay between a first time at which the reference audio data was received and a second time at which the audio signal is to be output by the audio source.

Abstract: In accordance with embodiments of the present disclosure, a processing circuit may implement an adaptive filter, a first signal injection portion which injects a first additional signal into a first frequency range content source audio signal, and a second signal injection portion which injects a second additional signal into a second frequency range content source audio signal, wherein the first additional signal and the second additional signal are substantially different.

Abstract: The technology described in this document can be embodied in a computer-implemented method that includes receiving, by one or more processing devices, a first plurality of values representing a set of coefficients of an adaptive filter disposed in an active noise cancellation system. The method also includes accessing one or more estimates of instantaneous phase values associated with a transfer function representing an effect of a secondary path of the active noise cancellation system, and updating the first plurality of values based on the one or more estimates of the instantaneous phase values to generate a set of updated coefficients for the adaptive filter. The method further includes programming the adaptive filter with the set of updated coefficients to affect operation of the adaptive filter.

Abstract: A time heuristic audio control system, comprises a receiver for receiving time-based data from a personal computing device and a memory storing one or more sets processing parameters comprising instructions for processing the ambient sound based upon the time-based data. The system further includes a processor coupled to the memory and the receiver configured to adjust the ambient sound as directed by a selected set of processing parameters retrieved from the memory to create adjusted audio, the selected set of processing parameters retrieved based upon the time-based data and at least one speaker for outputting the adjusted audio.

Abstract: The disclosure relates to combined active noise cancellation and noise compensation in a headphone. An audio processing device includes a selector and a noise compensation unit. The selector can select one of a plurality of first transfer functions based on at least one feature of at least one of an external noise and a content audio signal representing a sound to be reproduced through the headphone. The noise compensation unit can compute a second audio signal by applying the selected first transfer function to a first audio signal, and derive gains for the noise compensation at least based on the second audio signal. The at least one feature can be used to distinguish at least two of the first transfer functions. Each of the first transfer functions can transform the first audio signal to the second audio signal which is assumed as representing a version of the sound represented by the first audio signal, which arrives at an eardrum of a listener wearing the headphone.

Abstract: A system for audio processing comprising an initial background statistical model system configured to generate an initial background statistical model using a predetermined sample size of audio data. A parameter computation system configured to generate parametric data for the audio data including cepstral and energy parameters. A background statistics computation system configured to generate preliminary background statistics for determining whether speech has been detected. A first speech detection system configured to determine whether speech was present in the initial sample of audio data. An adaptive background statistical model system configured to provide an adaptive background statistical model for use in continuous processing of audio data for speech detection. A parameter computation system configured to calculate cepstral parameters, energy parameters and other suitable parameters for speech detection.

Abstract: A noise filter includes a filter device and a filter retainer. The filter device includes a device unit having at least one capacitor provided therein, an input side lead wire extending from the device unit, an output side lead wire extending from the device unit, and a ground connection portion provided on the device unit. The filter device forms the noise filter that uses a mutual inductance between the input side lead wire and the output side lead wire. The filter retainer has a holding structure configured to maintain an arrangement of the input side lead wire and the output side lead wire. The arrangement forms an overlapping section in which the lead wires are closely opposed to each other. The noise filter may be attached to a harness.

Abstract: A method and apparatus are for determining one or more background noise characteristics, determining one or more incoming audio characteristics; and generating a combined audio signal comprising an active noise cancellation (ANC) component and a modified incoming audio (MIA) component. The ANC component is determined based on at least one of the one or more incoming audio characteristics and the background noise characteristics. Each of a limit of the ANC component and a limit of the MIA component is dynamically controlled to be less than or equal to a system limit, wherein a limit of the combined signal is approximately at the system limit.

Abstract: A transmitter/receiver apparatus including: a transmitter/receiver terminal; a switching amplifier that includes a low-side switching element connected between a ground terminal and a pulse output terminal and a high-side switching element connected between the pulse output terminal and a power supply terminal and that outputs a pulse signal from the pulse output terminal; a filter that passes therethrough and outputs as a transmitted signal a predetermined frequency component of the pulse signal from a transmitter terminal; and a transmit/receive switch unit that switches the connection status between the transmitter/receiver terminal and the transmitter terminal and also switches the connection status between the transmitter/receiver terminal and a receiver terminal.

Abstract: Provided is a noise reduction apparatus, a noise reduction method, and a program for generating voice that is easy to hear while reducing sudden sound. The noise reduction apparatus includes a first microphone configured to collect mainly voice and output the voice as a first signal, a second microphone configured to collect mainly sound other than the voice and output the sound as a second signal, a sudden sound detecting unit configured to detect sudden sound in the first signal and the second signal, a sudden sound information storage unit configured to store the sudden sound as one or a plurality of reference signals, and an adder configured to remove the sudden sound from the first signal based on the reference signal stored in the sudden sound information storage unit.

Abstract: A method that is accomplished in a vehicle engine harmonic modification system. A non-zero target engine harmonic signal that is representative of a target engine harmonic sound level in the vehicle cabin is provided. The target engine harmonic signal is used in an operation of the engine harmonic modification system so as to modify the level of engine harmonic sound in the vehicle cabin, to bring the engine harmonic sound level in the vehicle cabin closer to the target engine harmonic sound level.

Abstract: A method for masking a vehicle noise includes setting a noise-generated driving region in which the vehicle noise is generated while a vehicle is running; measuring the vehicle noise in the noise-generated driving region; determining a masking sound which cancels the vehicle noise based on the measured vehicle noise; and outputting the determined masking sound into an interior of the vehicle.

Abstract: An integrated circuit for digital signal routing. The integrated circuit has analog and digital inputs and outputs, including digital interfaces for connection to other integrated circuits. Inputs, including the digital interfaces, act as data sources. Outputs, including the digital interfaces, act as data destinations. The integrated circuit also includes signal processing blocks, which can act as data sources and data destinations. Signal routing is achieved by means of a multiply-accumulate block, which takes data from one or more data source and, after any required scaling, generates output data for a data destination. Data from a data source is buffered for an entire period of a data sample clock so that the multiply-accumulate block can retrieve the data at any point in the period, and output data of the multiply-accumulate block is buffered for an entire period of the data sample clock so that the data destination can retrieve the data at any point in the period.

Abstract: Methods, systems, and media for ambient background noise modification are provided. In some implementations, the method comprises: identifying at least one noise present in an environment of a user having a user device, an activity the user is currently engaged in, and a physical or emotional state of the user; determining a target ambient noise to be produced in the environment based at least in part on the identified noise, the activity the user is currently engaged in, and the physical or emotional state of the user; identifying at least one device associated with the user device to be used to produce the target ambient noise; determining sound outputs corresponding to each of the one or more identified devices, wherein a combination of the sound outputs produces an approximation of one or more characteristics of the target ambient noise; and causing the one or more identified devices to produce the determined sound outputs.

Abstract: Personal audio systems and methods are disclosed. A personal audio system includes a processor to generate a personal audio stream by processing the digitized ambient audio in accordance with an active processing parameter set, an event detector to detect a trigger event other than manual selection by a user of the personal audio system, and a controller configured to, in response to detection of the trigger event, select a new processing parameter set and instruct the processor to begin generating the personal audio stream by processing the digitized ambient audio in accordance with the new processing parameter set.

Abstract: A method in a communications device includes the following operations. During a call, a process automatically detects that the device has moved from an at-the-ear position to an away-from-the-ear position. Based on the detection, a noise suppressor that operates upon an uplink signal for the call is signaled to change its noise suppression performance. Other embodiments are also described and claimed.

Abstract: According to one embodiment, a noise reduction system for reducing noise including impact noise repetitively generated at a time interval includes the following elements. The error signal generator generates an error signal based on the noise being detected. The delay signal generator has a time delay characteristic and delays a signal, which is generated based on the error signal, to generate a delay signal, the time delay characteristic being determined based on an imaging sequence or pre-scanning by the MRI device and corresponding to the time interval. The control filter generates the first control signal from the delay signal. The loudspeaker unit includes at least one pair of a first filter and a control loudspeaker and a transmission unit.

Abstract: An audio output device controls the application of active noise cancellation (ANC) levels based on a current status of the device. The audio output device includes a communication interface configured to provide a wired or wireless connection to a playback device, a signal detector configured to detect an audio signal provided through the communication interface, an ANC module configured to apply a first-level ANC to the audio signal, and a control unit configured to determine a level of the ANC to be applied by the ANC module. Upon determining that the audio signal is not provided based on information provided from at least one of the communication interface or the signal detector, the control unit instructs the ANC module to perform a second-level ANC.

Abstract: A noise reduction device includes a plurality of noise microphones, noise controller, and control speakers. Noise controller generates a control-sound signal to reduce a noise, at a control center of a control space, with the noise being detected by the plurality of noise microphones. Noise microphone disposed at a shorter distance from the control center than distance “d” indicated by Relational Expression (1) is smaller in number than noise microphones disposed at longer distances than distance “d” where “?” is a wavelength corresponding to control upper-limit frequency “f” in noise microphones, “d0” is a distance from the control center to control speakers, “t” is a control delay time in control speakers, and “v” is a sound speed.

Abstract: An adaptive noise canceling (ANC) circuit adaptively generates an anti-noise signal from a reference microphone signal that is injected into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone proximate the speaker provides an error signal. A secondary path estimating adaptive filter estimates the electro-acoustical path from the noise canceling circuit through the transducer so that source audio can be removed from the error signal. Tones in the source audio, such as remote ringtones, present in downlink audio during initiation of a telephone call, are detected by a tone detector using accumulated tone persistence and non-silence hangover counting, and adaptation of the secondary path estimating adaptive filter is halted to prevent adapting to the tones. Adaptation of the adaptive filters is then sequenced so any disruption of the secondary path adaptive filter response is removed before allowing the anti-noise generating filter to adapt.

Abstract: The technology described herein can be embodied in a computer implemented method that includes detecting, by one or more processing devices, onset of an unstable condition in an active noise control system. The method also includes obtaining, responsive to detecting the onset of the unstable condition, updated filter coefficients for a system-identification filter configured to represent a transfer function of a secondary path of the active noise control system. The updated filter coefficients are generated using a set of multiple subband adaptive filters, wherein filter coefficients of each subband adaptive filter in the set are configured to adapt to changes in a corresponding portion of a frequency range associated with potential unstable conditions in the active noise control system. The method also includes programming the system identification filter with the updated coefficients to affect operation of the active noise control system.