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 signal processing method and an audio equalizer, both implemented in an embedded system without incurring excessive computation, use a Kaiser-Bessel-derived (KBD) window and an Overlap-and-Add (OLA) processing to eliminate signal distortion of a time domain audio signal during signal conversion and to generate filters according to audio effects desired by a user.

A clock generator receives first and second clock signals, and input representing a desired frequency ratio. A comparison is made between frequencies of an output clock signal and the first clock signal, and a first error signal represents the difference between the desired frequency ratio and this comparison result. The first error signal is filtered. A comparison is made between frequencies of the output clock signal and the second clock signal, and a second error signal represents the difference between the filtered first error signal and this comparison result. The second error signal is filtered. A numerically controlled oscillator receives the filtered second error signal and generates an output clock signal. As a result, the output clock signal has the jitter characteristics of the first input clock signal over a useful range of jitter frequencies and the frequency accuracy of the second input clock signal.

Systems and methods to provide distortion sensing, prevention, and/or distortion-aware bass enhancement in audio systems can be implemented in a variety of applications. Sensing circuitry can generate statistics based on an input signal received for which an acoustic output is generated. In various embodiments, the statistics can be used such that a multi-notch filter can be used to provide input to a speaker to generate the acoustic output. In various embodiments, the statistics from the sensing circuitry can be provided to a bass parameter controller coupled to bass enhancement circuitry to operatively provide parameters to the bass enhancement circuitry. The bass enhancement circuitry can provide a bass enhanced signal for generation of the acoustic output, based on the parameters. Various combinations of a multi-notch filter and bass enhancement circuitry using statistics from sensing circuitry can be implemented to provide an enhanced acoustic output. Additional apparatus, systems, and methods are disclosed.

Examples involve calibration of a playback device including multiple audio transducers. An example implementation involves forming a particular playback configuration that configures the multiple audio transducers to form two or more channels and receiving a command to initiate calibration of the particular playback configuration. The calibration includes emitting a calibration sound according to a sequence that involves the calibration sound cycling through frequencies of a calibration frequency range. Based on the command, the playback device emits the calibration sound contemporaneously via the two or more channels. Emission of the calibration sound by each channel is staggered such that each emitted calibration sound is delayed relative to a preceding emitted calibration sound. The playback device receives audio processing coefficients that are based on one or more recordings of the emitted calibration sound; and applies the audio processing coefficients when playing back the audio content as part of the particular playback configuration.

A method for dynamic equalization is performed at a system. The system receives an electronic audio signal. The system monitors available headroom based on two or more of: an amplifier of the system, a change in volume of the system, characteristics of a transducer of a loudspeaker associated with the system, and the audio signal. Responsive to a change in the available headroom, the system modifies one or more parameters of an equalizer to at least a portion of the signal in order to extend a physical low-frequency response of the loudspeaker, the one or more parameters comprising a gain for the and a frequency.

A wearable acoustic transducer includes a communication part for performing communication and speaker and control modules. The speaker module includes an enclosure having an inner space, a first opening in which a speaker part is mounted being formed in a first side surface thereof, a second opening in which a passive vibration part is mounted being formed in a second side surface thereof, the speaker part configured to emit sound, and a passive vibration part for relieving air stiffness in the inner space of the enclosure. The control module performs gain correction on an electric signal including sound or voice that is received from the communication part according to prestored gain correction data and applies the gain-corrected electric signal to the speaker part, or generates an electric signal corresponding to the gain correction data and applies the electric signal to the speaker part.

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.

In an embodiment, a method comprises: filtering reference audio content items to separate the reference audio content items into different frequency bands; for each frequency band, extracting a first feature vector from at least a portion of each of the reference audio content items, wherein the first feature vector includes at least one audio characteristic of the reference audio content items; obtaining at least one semantic label from at least a portion of each of the reference audio content items; obtaining a second feature vector consisting of the first feature vectors per frequency band and the at least one semantic label; generating, based on the second feature vector, cluster feature vectors representing centroids of clusters; separating the reference audio content items according to the cluster feature vectors; and computing an average target profile for each cluster based on the reference audio content items in the cluster.

Provided are an electronic device and method for managing the perceived loudness of an input audio signal. The electronic device includes a communication unit configured to communicate with internal units and external devices and receive an audio source signal; a loudness adjustment unit configured to receive the audio source signal from the communication unit and obtain an input audio source signal from the audio source signal; a noise separator configured to separate noise from the input audio source signal; a voice booster configured to boost an output audio signal received from the loudness adjustment unit; a storage unit storing instructions; and at least one processor configured to communicate with the communication unit, the loudness adjustment unit, the noise separator, the voice booster, and the storage unit and control operations thereof by executing the stored instructions.