Abstract: Systems and methods for rendering audio signals are disclosed. In some embodiments, a method may receive an input signal including a first portion and the second portion. A first processing stage comprising a first filter is applied to the first portion to generate a first filtered signal. A second processing stage comprising a second filter is applied to the first portion to generate a second filtered signal. A third processing stage comprising a third filter is applied to the second portion to generate a third filtered signal. A fourth processing stage comprising a fourth filter is applied to the second portion to generate a fourth filtered signal. A first output signal is determined based on a sum of the first filtered signal and the third filtered signal. A second output signal is determined based on a sum of the second filtered signal and the fourth filtered signal.

Abstract: A method of processing an audio signal is disclosed. According to embodiments of the method, magnitude response information of a prototype filter is determined. The magnitude response information includes a plurality of gain values, at least one of which includes a first gain corresponding to a first frequency. The magnitude response information of the prototype filter is stored. The magnitude response information of the prototype filter at the first frequency is retrieved. Gains are computed for a plurality of control frequencies based on the retrieved magnitude response information of the prototype filter at the first frequency, and the computed gains are applied to the audio signal.

Abstract: Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. For each sound source of a plurality of sound sources in the virtual environment, a respective first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the respective sound source in the virtual environment, and the respective first intermediate audio signal is associated with a first bus. For each of the sound sources of the plurality of sound sources in the virtual environment, a respective second intermediate audio signal is determined. The respective second intermediate audio signal corresponds to a reverberation of the input audio signal in the virtual environment.

Abstract: Systems and methods discussed herein can provide three-dimensional audio virtualization with sweet spot adaptation. In an example, an audio processor circuit can be used to update audio signals for sweet spot adaptation based on information from at least one depth sensor or camera about a listener position in a listening environment.

Abstract: Systems and methods can provide an elevated, virtual loudspeaker source in a three-dimensional soundfield using loudspeakers in a horizontal plane. In an example, a processor circuit can receive at least one height audio signal that includes information intended for reproduction using a loudspeaker that is elevated relative to a listener, and optionally offset from the listener's facing direction by a specified azimuth angle. A first virtual height filter can be selected for use based on the specified azimuth angle. A virtualized audio signal can be generated by applying the first virtual height filter to the at least one height audio signal. When the virtualized audio signal is reproduced using one or more loudspeakers in the horizontal plane, the virtualized audio signal can be perceived by the listener as originating from an elevated loudspeaker source that corresponds to the azimuth angle.

Abstract: A method of presenting an audio signal to a user of a mixed reality environment is disclosed. According to examples of the method, an audio event associated with the mixed reality environment is detected. The audio event is associated with a first audio signal. A location of the user with respect to the mixed reality environment is determined. An acoustic region associated with the location of the user is identified. A first acoustic parameter associated with the first acoustic region is determined. A transfer function is determined using the first acoustic parameter. The transfer function is applied to the first audio signal to produce a second audio signal, which is then presented to the user.

Abstract: Systems and methods for rendering audio signals are disclosed. In some embodiments, a method may receive an input signal including a first portion and the second portion. A first processing stage comprising a first filter is applied to the first portion to generate a first filtered signal. A second processing stage comprising a second filter is applied to the first portion to generate a second filtered signal. A third processing stage comprising a third filter is applied to the second portion to generate a third filtered signal. A fourth processing stage comprising a fourth filter is applied to the second portion to generate a fourth filtered signal. A first output signal is determined based on a sum of the first filtered signal and the third filtered signal. A second output signal is determined based on a sum of the second filtered signal and the fourth filtered signal.

Abstract: Examples of the disclosure describe systems and methods for presenting an audio signal to a user of a wearable head device. According to an example method, a source location corresponding to the audio signal is identified. An acoustic axis corresponding to the audio signal is determined. For each of a respective left and right ear of the user, an angle between the acoustic axis and the respective ear is determined. For each of the respective left and right ear of the user, a virtual speaker position, of a virtual speaker array, is determined, the virtual speaker position collinear with the source location and with a position of the respective ear. The virtual speaker array includes a plurality of virtual speaker positions, each virtual speaker position of the plurality located on the surface of a sphere concentric with the user's head, the sphere having a first radius.

Abstract: A method of presenting an audio signal to a user of a mixed reality environment is disclosed. According to examples of the method, an audio event associated with the mixed reality environment is detected. The audio event is associated with a first audio signal. A location of the user with respect to the mixed reality environment is determined. An acoustic region associated with the location of the user is identified. A first acoustic parameter associated with the first acoustic region is determined. A transfer function is determined using the first acoustic parameter. The transfer function is applied to the first audio signal to produce a second audio signal, which is then presented to the user.

Abstract: Systems and methods for rendering audio signals are disclosed. In some embodiments, a method may receive an input signal including a first portion and the second portion. A first processing stage comprising a first filter is applied to the first portion to generate a first filtered signal. A second processing stage comprising a second filter is applied to the first portion to generate a second filtered signal. A third processing stage comprising a third filter is applied to the second portion to generate a third filtered signal. A fourth processing stage comprising a fourth filter is applied to the second portion to generate a fourth filtered signal. A first output signal is determined based on a sum of the first filtered signal and the third filtered signal. A second output signal is determined based on a sum of the second filtered signal and the fourth filtered signal.

Abstract: Systems and methods for rendering audio signals are disclosed. In some embodiments, a method may receive an input signal including a first portion and the second portion. A first processing stage comprising a first filter is applied to the first portion to generate a first filtered signal. A second processing stage comprising a second filter is applied to the first portion to generate a second filtered signal. A third processing stage comprising a third filter is applied to the second portion to generate a third filtered signal. A fourth processing stage comprising a fourth filter is applied to the second portion to generate a fourth filtered signal. A first output signal is determined based on a sum of the first filtered signal and the third filtered signal. A second output signal is determined based on a sum of the second filtered signal and the fourth filtered signal.

Abstract: Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. For each sound source of a plurality of sound sources in the virtual environment, a respective first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the respective sound source in the virtual environment, and the respective first intermediate audio signal is associated with a first bus. For each of the sound sources of the plurality of sound sources in the virtual environment, a respective second intermediate audio signal is determined. The respective second intermediate audio signal corresponds to a reflection of the input audio signal in a surface of the virtual environment.

Abstract: Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. For each sound source of a plurality of sound sources in the virtual environment, a respective first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the respective sound source in the virtual environment, and the respective first intermediate audio signal is associated with a first bus. For each of the sound sources of the plurality of sound sources in the virtual environment, a respective second intermediate audio signal is determined. The respective second intermediate audio signal corresponds to a reverberation of the input audio signal in the virtual environment.

Abstract: Systems and methods for providing accurate and independent control of reverberation properties are disclosed. In some embodiments, a system may include a reverberation processing system, a direct processing system, and a combiner. The reverberation processing system can include a reverb initial power (RIP) control system and a reverberator. The RIP control system can include a reverb initial gain (RIG) and a RIP corrector. The RIG can be configured to apply a RIG value to the input signal, and the RIP corrector can be configured to apply a RIP correction factor to the signal from the RIG. The reverberator can be configured to apply reverberation effects to the signal from the RIP control system. In some embodiments, one or more values and/or correction factors can be calculated and applied such that the signal output from a component in the reverberation processing system is normalized to a predetermined value (e.g., unity (1.0)).

Abstract: A system and method for providing low interaural coherence at low frequencies is disclosed. In some embodiments, the system may include a reverberator and a low-frequency interaural coherence control system. The reverberator may include two sets of comb filters, one for the left ear output signal and one for the right ear output signal. The low-frequency interaural coherence control system can include a plurality of sections, each section can be configured to control a certain frequency range of the signals that propagate through the given section. The sections may include a left high-frequency section for the left ear output signal and a right high-frequency section for the right ear output signal. The sections may also include a shared low-frequency section that can output signals to be combined by combiners of the left and right high-frequency sections.

Abstract: A method of processing an audio signal is disclosed. According to embodiments of the method, magnitude response information of a prototype filter is determined. The magnitude response information includes a plurality of gain values, at least one of which includes a first gain corresponding to a first frequency. The magnitude response information of the prototype filter is stored. The magnitude response information of the prototype filter at the first frequency is retrieved. Gains are computed for a plurality of control frequencies based on the retrieved magnitude response information of the prototype filter at the first frequency, and the computed gains are applied to the audio signal.

Abstract: Embodiments of systems and methods are described for reducing undesired leakage energy produced by a non-front-facing speaker in a multi-speaker system. For example, the multi-speaker system can include an array of forward-facing speakers, one or more upward-facing speakers, and/or one or more side-facing speakers. Filters coupled to any two of the speakers in the multi-speaker system can generate audio signals output by the coupled speakers to reduce, attenuate, or cancel a portion of an audio signal output by one or more non-front-facing speakers that acoustically propagates along a direct path from the respective non-front-facing speaker to a listening position in a listening area in front of the multi-speaker system.

Abstract: An audio processing system for processing a single channel audio signal includes a a processor configured to derive a synthetic difference component from the single channel audio input signal a filtering module configured to apply a first filter to the sum signal represented by the single channel signal and to apply a second filter to the synthetic difference signal; and a control module configured to crossfade to control the amount of the resulting audio signal effect by respectively scaling the sum signal and the difference signal.

Abstract: A method of presenting an audio signal to a user of a mixed reality environment is disclosed. According to examples of the method, an audio event associated with the mixed reality environment is detected. The audio event is associated with a first audio signal. A location of the user with respect to the mixed reality environment is determined. An acoustic region associated with the location of the user is identified. A first acoustic parameter associated with the first acoustic region is determined. A transfer function is determined using the first acoustic parameter. The transfer function is applied to the first audio signal to produce a second audio signal, which is then presented to the user.

Abstract: Systems, devices, and methods are described herein for adjusting a relationship between dialog and non-dialog signals in an audio program. In an example, information about a long-term dialog balance for an audio program can be received. The long-term loudness dialog balance can indicate a dialog-to-non-dialog loudness relationship of the audio program. A dialog loudness preference can be received, such as from a user, from a database, or from another source. A desired long-term gain or attenuation can be determined according to a difference between the received long-term dialog balance for the audio program and the received dialog balance preference. The long-term gain or attenuation can be applied to at least one of the dialog signal and the non-dialog signal of the audio program to render an audio program that is enhanced according to the loudness preference.