Dts, Inc.

Daekyoung Noh

Themis George Katsianos

Jean-Marc Jot

James Tracey

Robert C Maling, Iii

Aaron Kube Warner

Oveal Jerome Walker

Michael Johns

Guangji Shi

Antonius A Kalker

Ryan James Cassidy

Pavel Chubarev

Vlad Ionut Ursachi

Twafik Mohamed

Xing He

Xiaoyu Guo

Petronel Bigioi

A method comprises: detecting noise in an environment with a microphone to produce a noise signal; receiving a voice signal to be played into the environment through a loudspeaker; performing multiband correction of the noise signal based on a microphone transfer function of the microphone, to produce a corrected noise signal; performing multiband correction of the voice signal based on a loudspeaker transfer function of the loudspeaker to produce a corrected voice signal; and computing multiband voice intelligibility results based on the corrected noise signal and the corrected voice signal.

Context-aware voice intelligibility enhancement

A loudspeaker system can include a first loudspeaker driver provided in a substantially fixed spatial relationship relative to a microphone. The loudspeaker driver can be tuned, for example automatically and without user input. In an example, the tuning can include receiving transfer function reference information about the first loudspeaker driver and the microphone, and receiving information about a desired acoustic response for the loudspeaker system. The tuning can include determining a simulated response for the loudspeaker system using a first input signal and the transfer function reference information, and can include providing the first input signal to the first loudspeaker driver. In response to the first input signal, an actual response for the loudspeaker driver can be received using the microphone. A compensation filter can be determined for the loudspeaker system based on the determined simulated response and the received actual response for the loudspeaker system.

System and method for real time loudspeaker equalization

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.

Compensating for binaural loudspeaker directivity

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.

Immersive audio reproduction systems

Circuitry can separate a multi-channel input signal into a center signal and a residual signal, apply time-varying center and residual gains to the center and residual signals, and combine the gain-adjusted center and residual signals to form a multi-channel audio signal. The center and residual gains are automatically determined in response to the center and residual signals so as to prevent each channel of the multi-channel audio signal from exceeding a target volume. During first times, the center and residual gains can vary synchronously so as to ensure that amplitude and phase relationships among channels in the multi-channel input signal are retained into the multi-channel audio signal. During second times, the center and residual gains can vary independently so as to reduce the energy of the residual signal compared to the center signal in the multi-channel audio signal.

Center protection dynamic range control

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.

Distributed audio virtualization systems

The methods and apparatus described herein provides technical solutions to the technical problems facing crosstalk cancellation for 3D audio virtualization. One technical solution includes preconditioning audio signals based on crosstalk canceller characteristics and based on characteristics of sound sources at intended locations in 3D space. To provide these technical solutions, the systems and methods described herein include an audio virtualizer and an audio preconditioner. In particular, the audio virtualizer includes a crosstalk canceller, and the audio preconditioner preconditions audio signals based on characteristics of a crosstalk cancellation system and based on characteristics of a binaural synthesis system or intended input source location in space. This solution improves the overall accuracy of virtualization of 3D sound sources and reduces or eliminates audio artifacts such as incorrect localization, inter-channel sound level imbalance, or a sound level that is higher or lower than intended.

System and method for preconditioning audio signal for 3D audio virtualization using loudspeakers

An audio adjustment system is provided that can output a user interface customized by the provider of the audio system instead of the electronic device manufacturer. Such an arrangement can save both field engineers and manufacturers a significant amount of time. Advantageously, in certain embodiments, such an audio adjustment system can be provided without knowledge of the electronic device's firmware. Instead, the audio adjustment system can communicate with the electronic device through an existing audio interface in the electronic device to enable a user to control audio enhancement parameters in the electronic device. For instance, the audio adjustment system can control the electronic device via an audio input jack on the electronic device.

Audio adjustment system

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.

Sweet spot adaptation for virtualized audio

A multi-rate audio processing system and method provides real-time measurement and processing of amplitude/phase changes in the transition band of the lowest frequency subband caused by the audio processing that can be used to apply amplitude/phase compensation to the higher subband(s). Tone signals may be injected into the transition band to provide strong tonal content for measurement and processing. The real-time measurement and compensation adapts to time-varying amplitude/phase changes regardless of the source of the change (e.g. non-linear time-varying linear or user control parameters) and provides universal applicability for any linear audio processing.

Multi-rate system for audio processing

An audio adjustment system is provided that can output a user interface customized by the provider of the audio system instead of the electronic device manufacturer. Such an arrangement can save both field engineers and manufacturers a significant amount of time. Advantageously, in certain embodiments, such an audio adjustment system can be provided without knowledge of the electronic device's firmware. Instead, the audio adjustment system can communicate with the electronic device through an existing audio interface in the electronic device to enable a user to control audio enhancement parameters in the electronic device. For instance, the audio adjustment system can control the electronic device via an audio input jack on the electronic device. The electronic device can also include decoding features for decoding communications sent by the audio adjustment system.

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Huntington Beach, CA, United States of America

Daekyoung Noh