Abstract: Systems and methods to provide an audio spectral correction can be implemented in a variety of applications. Such systems and methods can include an equalizer design that uses efficient filter sections tuned with a closed form algorithm to give an accurate and intuitive frequency response with low complexity and minimal processing overhead. Embodiments can enable efficient and accurate control of reverberation time vs. frequency in artificial reverberators. Embodiments can enable efficient realization of a cascade combination of multi-band equalization functions as a single multi-band equalizer function, in which such functions can be expressed as graphic equalization functions using the same set of center frequencies as the cascade combination of multi-band equalization functions. Additional apparatus, systems, and methods are disclosed.

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: 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: A sound field coding system and method that provides flexible capture, distribution, and reproduction of immersive audio recordings encoded in a generic digital audio format compatible with standard two-channel or multi-channel reproduction systems. This end-to-end system and method mitigates any impractical need for standard multi-channel microphone array configurations in consumer mobile devices such as smart phones or cameras. The system and method capture and spatially encode two-channel or multi-channel immersive audio signals that are compatible with legacy playback systems from flexible multi-channel microphone array configurations.

Abstract: An audio signal processing system can be configured to provide virtualized audio information in a three-dimensional soundfield using at least a pair of loudspeakers or headphones. The system can include an audio input configured to receive audio program information that includes at least N discrete audio signals, a first virtualization processor circuit configured to generate intermediate virtualized audio information by filtering M of the N audio signals, and a second virtualization processor circuit configured to generate further virtualized audio information by differently filtering K of the N audio signals, wherein K, M, and N are integers. The system can include an audio signal combination circuit to combine the intermediate virtualized audio information with at least one of the N audio signals, other than the M audio signals, to render fewer than N audio signals for transmission to a second virtualization processor circuit.

Abstract: A method and apparatus for processing object-based audio signals is provided. The apparatus receives a plurality of object-based audio signals. Each object-based audio signal of the object-based audio signals includes audio waveform data and object metadata associated with the audio waveform data. The object metadata includes at least one of a loudness parameter or a power parameter associated with the audio waveform data. The apparatus determines a loudness metric based on the received object-based audio signals and based on the at least one of the loudness parameter or the power parameter for each object-based audio signal of the received object-based audio signals. In one configuration, the apparatus renders the received object-based audio signals to a set of output signals based on the determined loudness metric. In another configuration, the apparatus transmits (e.g., broadcast, file delivery, or streaming) the received object-based audio signals based on the determined loudness metric.

Abstract: Accurate modeling of acoustic reverberation can be essential to generating and providing a realistic virtual reality or augmented reality experience for a participant. In an example, a reverberation signal for playback using headphones can be provided. The reverberation signal can correspond to a virtual sound source signal originating at a specified location in a local listener environment. Providing the reverberation signal can include, among other things, using information about a reference impulse response from a reference environment and using characteristic information about reverberation decay in a local environment of the participant. Providing the reverberation signal can further include using information about a relationship between a volume of the reference environment and a volume of the local environment of the participant.

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: A sound field coding system and method that provides flexible capture, distribution, and reproduction of immersive audio recordings encoded in a generic digital audio format compatible with standard two-channel or multi-channel reproduction systems. This end-to-end system and method mitigates any impractical need for standard multi-channel microphone array configurations in consumer mobile devices such as smart phones or cameras. The system and method capture and spatially encode two-channel or multi-channel immersive audio signals that are compatible with legacy playback systems from flexible multi-channel microphone array configurations.

Abstract: MIDI-generated audio streams or other input streams of audio events are perceptually associated with specific locations in 3D space with respect to the listener. A conventional pan parameter is redefined so that it no longer specifies the relative balance between the audio being fed to two fixed speaker locations. Instead, the new MIDI pan parameter extension specifies a virtual position of an audio stream in 3D space. Preferably, the relative position of a single audio stream is set along a predefined arc in 3D space.

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: 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 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 sound field coding system and method that provides flexible capture, distribution, and reproduction of immersive audio recordings encoded in a generic digital audio format compatible with standard two-channel or multi-channel reproduction systems. This end-to-end system and method mitigates any impractical need for standard multi-channel microphone array configurations in consumer mobile devices such as smart phones or cameras. The system and method capture and spatially encode two-channel or multi-channel immersive audio signals that are compatible with legacy playback systems from flexible multi-channel microphone array configurations.

Abstract: A digital signal is processed by splitting it into at least two frequency subbands and the two subband signals are downsampled. A filter is applied in at least one of the subband signals. At least one of the phase and magnitude of the subband filtered signals is matched in the transition frequency band between the two subbands.

Abstract: A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.

Abstract: Accurate modeling of acoustic reverberation can be essential to generating and providing a realistic virtual reality or augmented reality experience for a participant. In an example, a reverberation signal for playback using headphones can be provided. The reverberation signal can correspond to a virtual sound source signal originating at a specified location in a local listener environment. Providing the reverberation signal can include, among other things, using information about a reference impulse response from a reference environment and using characteristic information about reverberation decay in a local environment of the participant. Providing the reverberation signal can further include using information about a relationship between a volume of the reference environment and a volume of the local environment of the participant.

Abstract: This invention describes a method for decentralized decoding of a multichannel audio signal by broadcasting the original encoded data and distributing the decoding process between a plurality of receiving units. This allows for the design and manufacture of scalable multichannel audio reproduction systems having an arbitrary number of output channels, composed of a plurality of generic decoder and loudspeaker units each generating fewer output channels. With distributed decoding, a manufacturer can use “off-the-shelf” stereo or mono signal processors, digital-to-analog converters and amplifier components in each generic decoding module, thus reducing manufacturing costs and complexity requirements for each module while offering unlimited scalability in the total number of output channels.

Abstract: Systems and methods to provide an audio spectral correction can be implemented in a variety of applications. Such systems and methods can include an equalizer design that uses efficient filter sections tuned with a closed form algorithm to give an accurate and intuitive frequency response with low complexity and minimal processing overhead. Embodiments can enable efficient and accurate control of reverberation time vs. frequency in artificial reverberators. Embodiments can enable efficient realization of a cascade combination of multi-band equalization functions as a single multi-band equalizer function, in which such functions can be expressed as graphic equalization functions using the same set of center frequencies as the cascade combination of multi-band equalization functions. Additional apparatus, systems, and methods are disclosed.

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.

Abstract: A digital signal is processed by splitting it into at least two frequency subbands and the two subband signals are downsampled. A filter is applied in at least one of the subband signals. At least one of the phase and magnitude of the subband filtered signals is matched in the transition frequency band between the two subbands.