An active noise cancellation (ANC) system including a selectable decimation rate decimator that receives an oversampled digital input and has an input that selects the decimation rate, a filter that receives an output of the decimator, and a selectable interpolation rate interpolator that receives an output of the filter and has an input that selects the interpolation rate. The selectable decimation rate decimator and the selectable interpolation rate interpolator operate to provide a selectable sample rate for the filter based on the selected decimation and interpolation rates. The filter may be an anti-noise filter, feedback filter, and/or a filter that models an acoustic transfer function of the ANC system. Rate selection may be static, or dynamically controlled based on battery or ambient noise level. A ratio of the decimation rate and the interpolation rate is fixed independent of the dynamically controlled decimation and interpolation rates.

Provided is an active noise cancellation (ANC) method applied for an ANC headphone. The ANC method includes: in a channel estimation mode, estimating a plurality of environment channels by generating, transmitting and capturing a training signal; in the channel estimation mode, tuning a plurality of ANC filters based on the estimated plurality of environment channels; and in a normal mode, performing ANC on an input signal based on the plurality of ANC filters.

Systems and methods for noise cancelation in a listening area include functionality that activates a microphone of the remote control device of a home entertainment system or a microphone of another mobile device to collect a baseline white noise profile of a user's media content listening area, such as the user's family room, living room or other TV viewing space. This baseline white noise profile may be used by the user's set-top box (STB) or other receiving device to generate the corresponding noise canceling audio signal to create a noise-canceling effect and a more immersive and enjoyable listening experience for the user. A plurality of audio data samples from which to select, each representing a different baseline profile of ambient white noise associated with the media content listening area, may be generated and used for different devices that generate white noise.

A filtering device includes a low-pass filter (LPF), a noise estimation circuit and a first combining circuit. The LPF receives and filters a pre-filtering signal to generate an output signal of the filtering device. The noise estimation circuit estimates an estimated noise signal according to the output signal and the pre-filtering signal. The first combining circuit subtracts the estimated noise signal from an input signal of the filtering device to generate the pre-filtering signal.

A voice communication technique by which superior echo canceling can be realized even in use of non-directional microphones is provided. When voice uttered by a near end speaker is defined as speaker's voice; sound obtained by emitting a received speech signal, which is a voice signal of a far end speaker, by a loudspeaker is defined as reproduction sound; a signal obtained by picking up an acoustic signal, which contains the speaker's voice and the reproduction sound, by a first microphone is defined as a first sound pickup signal; and a signal obtained by picking up an acoustic signal, which contains the speaker's voice and the reproduction sound, by a second microphone is defined as a second sound pickup signal, a voice communication device includes: a first subtraction unit that generates a first-subtracted first sound pickup signal that is a difference between a corrected sound pickup signal and the second sound pickup signal or the first sound pickup signal, the corrected sound pickup signal being obtained by correcting either one of the first sound pickup signal and the second sound pickup signal on the basis of an amplitude difference based on a difference between a distance d1 from the loudspeaker to the first microphone and a distance d2 from the loudspeaker to the second microphone (d2d1); and a second subtraction unit that generates a transmitted speech signal, which is to be transmitted to the far end speaker, by subtracting an adaptive-filtered second sound pickup signal, which is obtained by canceling echo, from the first-subtracted first sound pickup signal.

A method and system for operating a signal filter device are provided in the present disclosure. The signal filter device may include M adaptive filters, M first analysis filters, M second analysis filters, and a controller. M may be an integer above two. The method may include generating, by the M first analysis filters, M first sub-band signals based on an input signal. The method may also include generating, by the M second analysis filters, M second sub-band signals based on a reference signal. The method may further include adjusting, by the controller, the working states of the M adaptive filters based on correlations between the M first sub-band signals and the M second sub-band signals.

Arrangements for communication and/or noise attenuation within an aircraft, and aircraft and methods for making aircraft including such arrangements are provided. In one example, an arrangement includes an array of first microphones cooperatively configured to be directed towards a first aircraft operator when disposed in a first cockpit seat to receive a first communication input from the first aircraft operator. An array of first speakers is cooperatively configured to be directed towards the first aircraft operator when disposed in the first cockpit seat to provide a first communication output to the first aircraft operator.

The overall performance of an ANC system may be improved by configuring the ANC system to perform adaption in the frequency domain. The ANC systems may be configured to update an algorithm of an adaptive filter based, at least in part, on the first input signal, the second input signal, and a feedback signal that is based on an output of the adaptive filter. Updating may include changing parameters of the algorithm based on a SDR based, at least in part, on the first input signal. Updating may also include normalizing a step size and processing at least full band information for the input signal in a frequency domain to generate coefficient values for the algorithm. Updating may also include applying a frequency domain magnitude constraint on adaptive filter coefficients.

According to an example embodiment, an apparatus for active cancellation of sound and vibration is provided, the apparatus including sound and vibration generation components for jointly producing vibration and sound under control of a driving signal provided as input thereto, the components being arranged inside a padding to generate mechanical vibration that is perceivable as a vibration and sound on at least one outer surface of the padding and to radiate a sound through the at least one outer surface of the padding, a feedback unit for providing feedback information that is indicative of acoustic energy of sound and vibration inside the padding, and a drivert for generating the driving signal in dependence of the feedback information so as to reduce energy of ambient sound and vibration induced inside the padding due to one or more external sources of sound and vibration.

A processing module for a noise control circuit may include a howling detector configured to receive an input signal and to determine a linearity metric based on the input signal, the linearity metric comprising a measure of the linearity of a logarithmic representation of the energy of the input signal. The processing module may also include a gain adjuster configured to adjust the gain of a noise control unit.