A method of equalising an audio signal derived from a microphone, the method comprising: receiving the audio signal; applying an order-statistic filter to the audio signal in the frequency domain to generate a statistically filtered audio signal; equalising the received audio signal based on the statistically filtered audio signal to generate an equalised audio signal.

In accordance with embodiments of the present disclosure, an adjustable equalization filter may have a response that generates an equalized source audio signal from a source audio signal to account for effects of changes in an electro-acoustical path of the source audio signal to a transducer. An equalizer coefficient control block may adapt the response of the adjustable equalization filter in response to changes in a response of a secondary path estimate filter for modeling the electro-acoustical path of a source audio signal through the transducer, wherein a response of the secondary path estimate filter is adapted in conformity with an error microphone signal indicative of the acoustic output of the transducer.

A method for recording a playback setting of sound is disclosed. The method is applied to an electronic device having an equalizer and includes the following steps of: acquiring an archival data of a sound file played by the electronic device; receiving a set record command; acquiring a gain setting data of the equalizer according to the set record command; pairing the archival data with the gain setting data to generate a pairing data; and saving the pairing data.

Processes and devices for equalizing an audio system that is adapted to use a loudspeaker to transduce test audio signals into test sounds. The processes and devices can involve the use of infrared signals to convey information in one or both directions between the audio system and a portable computer device that captures test sounds, calculates audio parameters that can be used in the equalization process, and transmits these audio parameters back to the audio system for its use in equalizing audio signals that are played by the audio system.

The present invention provides methods and systems for digitally processing audio signals. Some embodiments receive an audio signal and converting it to a digital signal. The gain of the digital signal may be adjusted a first time, using a digital processing device located between a receiver and a driver circuit. The adjusted signal can be filtered with a first low shelf filter. The systems and methods may compress the filtered signal with a first compressor, process the signal with a graphic equalizer, and compress the processed signal with a second compressor. The gain of the compressed signal can be adjusted a second time. These may be done using the digital processing device. The signal may then be output through an amplifier and driver circuit to drive a personal audio listening device. In some embodiments, the systems and methods described herein may be part of the personal audio listening device.

Provided are methods, systems, and apparatuses for recording a three-dimensional (3D) sound field using a vertically-oriented cylindrical array with multiple circular arrays at different heights. The design of the cylindrical array is well-suited to providing a high-resolution in azimuth and a reduced resolution in elevation, and offers improved performance over existing 3D sound reproduction systems. The methods, systems, and apparatuses provide a larger vertical aperture than horizontal aperture, as opposed to a spherical array, which has the same aperture for all dimensions, and further provides an alternative format to mixed-order spherical decomposition.

A technique for determining one or more equalization parameters includes acquiring, via one or more electrodes, auditory brainstem response (ABR) data associated with a first audio sample and determining, via a processor, one or more equalization parameters based on the ABR data. The technique further includes reproducing a second audio sample based on the one or more equalization parameters, acquiring, via the one or more electrodes, complex auditory brainstem response (cABR) data associated with the second audio sample, and comparing, via the processor, the cABR data to at least one representation of the second audio sample to determine at least one measure of similarity. The technique further includes modifying the one or more equalization parameters based on the at least one measure of similarity.

Processes and devices for equalizing an audio system that is adapted to use a loudspeaker to transduce test audio signals into test sounds. The processes and devices can involve the use of infrared signals to convey information in one or both directions between the audio system and a portable computer device that captures test sounds, calculates audio parameters that can be used in the equalization process, and transmits these audio parameters back to the audio system for its use in equalizing audio signals that are played by the audio system.

Systems and methods are described for dynamically suppressing non-linear distortion for a device, such as a speakerphone. A device may receive a signal, where the device has non-linear distortion at a predetermined frequency. The received signal may be analyzed to compute a tone strength parameter and a band level. The received signal may be filtered such that a spectrum of the input signal is dynamically limited by reducing suppression of the non-linear distortion when the tone strength parameter is in a lower portion of a predetermined range and increasing suppression of the non-linear distortion when the tone strength parameter is in an upper portion of the predetermined range, the predetermined range of the tone strength parameter corresponding to a loudness range of the device.

Systems and methods are described for dynamically suppressing non-linear distortion for a device, such as a speakerphone. A device may receive a signal, where the device has non-linear distortion at a predetermined frequency. The received signal may be analyzed to compute a tone strength parameter and a band level. The received signal may be filtered such that a spectrum of the input signal is dynamically limited by reducing suppression of the non-linear distortion when the tone strength parameter is in a lower portion of a predetermined range and increasing suppression of the non-linear distortion when the tone strength parameter is in an upper portion of the predetermined range, the predetermined range of the tone strength parameter corresponding to a loudness range of the device.