The directivity of a loudspeaker describes how sound produced by the speaker varies with angle and frequency. Low-frequency sound tends to be relatively omnidirectional, while high-frequency sound tends to be more strongly directional. Because the two ears of a listener are in different spatial positions, the direction-dependent performance of the speakers can produce unwanted differences in volume or spectral content between the two ears. For example, high-frequency sounds may appear to be muffled in one ear, compared to the other. A multi-speaker sound system can employ binaural directivity compensation, which can compensate for directional variations in performance of each speaker, and can reduce or eliminate the difference in volume or spectral content between the left and right ears of a listener. The binaural directivity compensation can optionally be included with spatial audio processing, such as crosstalk cancellation, or can optionally be included with loudspeaker equalization.

An apparatus for providing a contrast mode and a front optimized mode for audio in a vehicle is provided. An audio controller is programmed to transmit first audio content in a first zone seating area and to transmit second audio content in a second zone seating area. The audio controller receives a first indication to transmit the first audio content in the first zone seating area and the second audio content in the second zone seating area in the contrast mode to provide an equal listening experience. The audio controller receives a second indication to transmit the first audio content in the first zone seating area and the second audio content in the second zone seating area in the front optimized mode to increase a quality of sound in the first zone seating area and to decrease a quality of sound in the second zone seating area.

Embodiments provide methods, apparatuses, and systems for performing crosstalk cancellation and/or generation of virtual speakers. An audio processor may include a crosstalk cancellation circuit and a linearization circuit. The linearization circuit may offset the frequency response of the crosstalk cancellation circuit to provide an overall frequency response that is flat. A virtual speaker circuit may receive an input signal associated with an output channel and pass the input signal to the output channel unmodified. The virtual speaker circuit generates a virtualization signal based on the input signal and passes the virtualization signal to another physical channel. The virtualization signal may be generated further based on an ipsilateral head-related transfer function (HRTF) and a contralateral HRTF that correspond to a virtual speaker location of a virtual speaker generated by the virtual speaker circuit. Other embodiments may be described and/or claimed.

An acoustic crosstalk canceller is determined for an asymmetric audio playback device, by determining a transfer function of an acoustic stereo playback path having asymmetries defined by speakers of the playback device. The transfer function is inverted to determine an inverse transfer function. The inverse transfer function is regularised by applying frequency dependent regularisation parameters to obtain an acoustic crosstalk canceller. Also, the inverse transfer function could be regularised for symmetric playback paths by applying aggregated frequency dependent regularisation parameters to obtain an acoustic crosstalk canceller without band branching.

Apparatus for managing and/or reducing harmonic distortion arising with an audio signal including a phase generator for generating at least one phase-difference signal or a reference audio signal generated by the phase generator, wherein the or each constant phase difference is adapted to provide cancellation of the harmonic distortion components arising along the signal path. Respective amplifier channels for receiving and separately amplifying the audio signal acting as a reference audio signal are also provided as are respective loudspeaker channels for receiving and separately producing sound corresponding to the amplified audio wherein each loudspeaker channel has substantially equal performance parameters and is adapted to radiate the sound relative to other loudspeaker channels to produce a combined sound that corresponds to the audio signal with harmonic distortion components that are reduced compared to the harmonic distortion components arising along the signal path.

An apparatus for providing a contrast mode and a front optimized mode for audio in a vehicle is provided. An audio controller is programmed to transmit first audio content in a first zone seating area and to transmit second audio content in a second zone seating area. The audio controller receives a first indication to transmit the first audio content in the first zone seating area and the second audio content in the second zone seating area in the contrast mode to provide an equal listening experience. The audio controller receives a second indication to transmit the first audio content in the first zone seating area and the second audio content in the second zone seating area in the front optimized mode to increase a quality of sound in the first zone seating area and to decrease a quality of sound in the second zone seating area.

A dual microphone signal processing arrangement for reducing reverberation is described. Time domain microphone signals are developed from a pair of sensing microphones. These are converted to the time-frequency domain to produce complex value spectra signals. A binary gain function applies frequency-specific energy ratios between the spectra signals to produce transformed spectra signals. A sigmoid gain function based on an inter-microphone coherence value between the transformed spectra signals is applied to the transformed spectra signals to produce coherence adapted spectra signals. And an inverse time-frequency transformation is applied to the coherence adjusted spectra signals to produce time-domain reverberation-compensated microphone signals with reduced reverberation components.

The present invention relates to methods for generating a conversion filter (KF) for converting a multidimensional original audio signal (AA) into a two-dimensional listening audio signal (HA), comprising the following steps: 1. Transformation of a time-based original audio signal (PAA) into a frequency-based original audio signal (FAA) 2. Sequential optimization of a basis conversion matrix (BKM) for converting the frequency-based original audio signal (FAA) into a frequency-based listening audio signal (FHA) using a first optimization algorithm (KA1), preferably starting from low frequencies and ascending at least up to a switching frequency (UF) 3. Sequential optimization of the basis conversion matrix (BKA) for converting the frequency-based original audio signal (FAA) into a frequency-based listening audio signal (FHA) using a second optimization algorithm (KA2), preferably starting from the switching frequency (UF) and ascending to high frequencies 4. Storing the optimized basis conversion matrix (BKA) of the correlation between the frequency-based original audio signal (FAA) and the frequency-based listening audio signal (FHA) in a frequency-based conversion matrix (FKM) 5. Transforming the frequency-based conversion matrix (FKM) into a time-based conversion matrix (PKM) as a conversion filter (KF).

A system may include an input configured to receive an audio signal, a filter having a negative group delay within a range of frequencies which are human-audible, the filter configured to receive the audio signal and filter the audio signal to generate a filtered audio signal, and a modulator configured to receive the filtered audio signal and modulate the filtered audio signal to generate a modulated filtered audio signal for communication over a digital interface.

A method and apparatus for audio signal processing in an audio chain to correct a non-linearity of the electroacoustic transducers in the audio chain by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain using an approximation of the quadratic function. The method accommodates the psychoacoustical characteristics of the human ear by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain approximating by a non-linear fifth degree polynomial function for a pressure change by the human ear up to p_. The method and apparatus reduce non-linearities of the entire audio chain with the human ear, by adding non-linearities in the audio chain so that an audio chain characteristic reduces the non-linearity of the human ear polynomial approximation to the pressure change p_=1 Pa.