The invention relates to Higher-Order Ambisonics (HOA) Content Loudness Level Adjustment. A method for adjusting a loudness level of a HOA audio signal on a transmitting side comprises steps of measuring loudness of only the W-channel of the HOA audio signal, and leveling HOA master gain for all channels of the HOA signal according to the measured loudness of the W-channel.

Example techniques involve calibration of one or more playback devices. An example implementation involves a playback device playing back audio content using a first calibration via the one or more audio transducers and the one or more amplifiers. The first calibration is based on a response of a listening environment to audio content playback by the playback device. The playback device records, via one or more microphones, at least a portion of the played back audio content. Based on the recorded audio content, the playback device detects a change in the response of the listening environment to audio content playback by the playback device. Responsive to detecting the change in the response, the playback device causes output of a prompt to initiate a calibration procedure for the playback device. The calibration procedure involves a mobile device recording playback by the playback device.

Disclosed are systems and methods for processing an audio signal. In particular, there is provided a method for determining dynamic gain values to be applied on a digital input signal. The digital signal may be arranged in blocks. The dynamic gain values may be used for attenuating input signal values exceeding a clipping threshold. More particularly, the method comprising, for each signal block, passing backwards over the next signal block and the current signal block to produce a preliminary gain contour from the input signal; and passing forwards over the current signal block to produce a final gain contour for the current signal block based on the preliminary gain contour, wherein the gain contours are produced by applying an instant gain ascent and a smooth gain decay to the gain contours.

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.

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.

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.

A device and method for controlling reproduction of an audio signal is provided, wherein the device is operated by means of an energy storage device. The method comprises the steps of deactivating a normal mode and activating an energy saving mode. Power consumption from the energy storage device for reproduction of the audio signal is reduced in the energy saving mode when compared to the normal mode. The method comprises reducing in the energy saving mode, a bass frequency component of a frequency spectrum of the audio signal and outputting the audio signal with reduced bass frequency component. The method further comprises ascertaining a charge state of the energy storage device and controlling the reduction in the bass frequency component based on a decrease in the charge state of the energy storage device.

A method or apparatus for delivering audio programming such as music to listeners may include identifying, capturing and applying a listener's audio perception characteristics (sometimes referred to as the listener's Personal Waveform Profile) to transform audio content so that the listener perceives the content similarly to how the content would be perceived by a different listener. An audio testing tool may be implemented as software application to identify and capture respective listeners' Personal Waveform Profiles. A signal processor may operate an algorithm processing source audio content using the respective listeners' Personal Waveform Profiles to provide audio output based on a difference between different profiles.

Disclosed is a system configured to determine filter parameters of an audio precompensation filter for the compensation of an associated sound generating system having at least two loudspeakers. The system is configured to obtain sound field models describing the sound field in at least one region of space designated the bright zone and the sound field in at least one region of space designated the dark zone. The system is also configured to obtain a target sound field in the bright zone. Further, the system is configured to determine filter parameters of the audio precompensation filter so that a criterion function is optimized under the constraint of causality and stability of the dynamics of the audio precompensation filter to enable reproduction of a desired target sound field in the bright zone, while reproducing as little sound as possible in the dark zone.

A sound processing apparatus includes a processing unit that is configured to acquire an audio signal, perform a correction processing on the acquired audio signal and output the correction-processed audio signal to a sound emitting unit. The correction processing includes an indirect sound adjusting processing in which a given signal processing is performed on an audio signal so as to adjust an influence of an indirect sound to be heard at a sound receiving point, and a frequency characteristic adjusting processing in which a frequency characteristic of an audio signal is adjusted. In the correction processing, a frequency characteristic for the frequency characteristic adjusting processing is determined based on a frequency characteristic of the indirect sound adjusting processing.