A system and method for providing a dynamic audio environment within a vehicle that includes receiving data associated with at least one audio stream that is being played through an audio system of the vehicle. The system and method also include determining a selected vehicle mode of the vehicle. The system and method additionally include selecting at least one audio effect that is associated with the selected vehicle mode of the vehicle to alter the at least one audio stream. The system and method further include controlling at least one audio source of the vehicle to provide the at least one audio stream according to the at least one audio effect.

A method for controlling a mobile terminal, the method including generating a first sound based on first information related to audio output; generating second information related to the audio output based on the first information and features of the first sound; and adjusting a volume or a frequency of a second sound based on the second information and outputting the second sound with adjusted volume or adjusted frequency, in which the first sound comprises a first sub-sound output through a speaker, a second sub-sound output from a case of the mobile terminal that is vibrated by the first sub-sound, and a third sub-sound output by a resting surface that comes in contact with the case of the mobile terminal and is vibrated by vibration of the case of the mobile terminal.

An adaptive analog parallel combiner circuit for receiver data recovery from a communication signal is provided. The circuit includes a summer that sums outputs of a plurality of filter taps in parallel, including zeroth and first through Nth filter taps, each filter tap having as input the communication signal or a version thereof, wherein N is a finite integer greater than or equal to two. The zeroth filter tap has an amplifier with gain controlled by a zeroth adaptive gain control coefficient, and each of the first through Nth filter taps having an all pass filter and gain controlled amplification, with gain controlled by a corresponding one of a first through Nth adaptive gain control coefficients and the all pass filter implementing a transfer function having a zero and a pole equaling each other and at a base frequency divided by a corresponding integer from one through N.

An electronic audio device has a microchip has embedded therein, circuitry for enhancing the sound outputted therefrom and which is associated, such as a radio, television, headphones, earbuds and the like.

A head-mounted wearable device (HMWD) provides audio output from a first speaker that is driven with a first signal and a second speaker that is driven with a second signal. Based on a volume level setting, an equalization profile and inversion frequency are determined. The equalization profile selectively amplifies or attenuates particular frequencies or ranges of frequencies. Those frequencies in the second signal that are above the inversion frequency have their amplitude inverted, relative to the first signal. When driven by the first signal and the second signal, the first speaker and the second speaker operate as acoustic dipoles below the inversion frequency and acoustic quadrupoles above the inversion frequency. Sound from the first and second speakers with frequencies above the inversion frequency exhibits destructive interference. As a result, the user wearing the HMWD is able hear audio output while audio amplitude perceived by the bystanders is significantly reduced.

Apparatus and methods for processing an audio signal. The apparatus may include a digital signal processor (DSP) configured to receive an audio signal from a vehicular audio signal line. The audio signal may include a flat component and a compensatory component. The flat component may correspond to the audio signal in a state before combination with the compensatory component. The compensatory component may include an OEM EQ component. The apparatus may include a microprocessor in electronic communication with the DSP. The microprocessor may be configured to retrieve from the memory a restorative signal component. The microprocessor may be configured to instruct the DSP to apply the restorative signal component to the audio signal to reduce the audio signal to the flat component.

A method, apparatus and computer program product are provided in order to present an intuitive visual user interface for audio equalization. In the context of a method includes defining at least two locations upon the visual user interface. Each location is associated with a predefined audio equalization characteristic. The method also causes first and second configurable elements to be provided by the visual user interface. Further, the method includes associating an audio equalization characteristic with the first configurable element based upon a location of the first configurable element relative to the at least two locations. The second configurable element provides a visual representation of the audio equalization characteristic based upon the location of the first configurable element.

A method or apparatus for delivering audio programming such as music to listeners may include identifying, capturing and applying a listener's audiometric profile to transform audio content so that the listener hears the content similarly to how the content was originally heard by a creative producer of the content. An audio testing tool may be implemented as software application to identify and capture the listener's audiometric profile. A signal processor may operate an algorithm used for processing source audio content, obtaining an identity and an audiometric reference profile of the creative producer from metadata associated with the content. The signal processor may then provide audio output based on a difference between the listener's and creative producer's audiometric profiles.

An audio post-processor for post-processing an audio signal having a time-variable high frequency gain information as side information includes: a band extractor for extracting a high frequency band of the audio signal and a low frequency band of the audio signal; a high band processor for performing a time-variable modification of the high frequency band in accordance with the time-variable high frequency gain information to obtain a processed high frequency band; and a combiner for combining the processed high frequency band and the low frequency band. Furthermore, a pre-processor is illustrated.

Some embodiments of the invention are directed to enabling a user to easily identify the frequency range(s) at which sound masking occurs, and addressing the masking, if desired. In this respect, the extent to which a first stem is masked by one or more second stems in a frequency range may depend not only on the absolute value of the energy of the second stem(s) in the frequency range, but also on the relative energy of the first stem with respect to the second stem(s) in the frequency range. Accordingly, some embodiments are directed to modeling sound masking as a function of the energy of the stem being masked and of the relative energy of the masked stem with respect to the masking stem(s) in the frequency range, such as by modeling sound masking as loudness loss, a value indicative of the reduction in loudness of a stem of interest caused by the presence of one or more other stems in a frequency range.