REVERBERATION GAIN NORMALIZATION
    32.
    发明公开

    公开(公告)号:US20230245642A1

    公开(公告)日:2023-08-03

    申请号:US18296901

    申请日:2023-04-06

    CPC classification number: G10K15/08

    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)).

    INTERAURAL TIME DIFFERENCE CROSSFADER FOR BINAURAL AUDIO RENDERING

    公开(公告)号:US20220132264A1

    公开(公告)日:2022-04-28

    申请号:US17516407

    申请日:2021-11-01

    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 first input audio signal is received, the first input audio signal corresponding to a source location in a virtual environment presented to the user via the wearable head device. The first input audio signal is processed to generate a left output audio signal and a right output audio signal. The left output audio signal is presented to the left ear of the user via a left speaker associated with the wearable head device. The right output audio signal is presented to the right ear of the user via a right speaker associated with the wearable head device. Processing the first input audio signal comprises applying a delay process to the first input audio signal to generate a left audio signal and a right audio signal; adjusting a gain of the left audio signal; adjusting a gain of the right audio signal; applying a first head-related transfer function (HRTF) to the left audio signal to generate the left output audio signal; and applying a second HRTF to the right audio signal to generate the right output audio signal. Applying the delay process to the first input audio signal comprises applying an interaural time delay (ITD) to the first input audio signal, the ITD determined based on the source location.

    Reverberation gain normalization
    34.
    发明授权

    公开(公告)号:US10810992B2

    公开(公告)日:2020-10-20

    申请号:US16442359

    申请日:2019-06-14

    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)).

    Index scheming for filter parameters

    公开(公告)号:US10779082B2

    公开(公告)日:2020-09-15

    申请号:US16427315

    申请日:2019-05-30

    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.

    EFFICIENT RENDERING OF VIRTUAL SOUNDFIELDS
    36.
    发明申请

    公开(公告)号:US20190379992A1

    公开(公告)日:2019-12-12

    申请号:US16438358

    申请日:2019-06-11

    Abstract: An audio system and method of spatially rendering audio signals that uses modified virtual speaker panning is disclosed. The audio system may include a fixed number F of virtual speakers, and the modified virtual speaker panning may dynamically select and use a subset P of the fixed virtual speakers. The subset P of virtual speakers may be selected using a low energy speaker detection and culling method, a source geometry-based culling method, or both. One or more processing blocks in the decoder/virtualizer may be bypassed based on the energy level of the associated audio signal or the location of the sound source relative to the user/listener, respectively. In some embodiments, a virtual speaker that is designated as an active virtual speaker at a first time, may also be designated as an active virtual speaker at a second time to ensure the processing completes.

    Efficient rendering of virtual soundfields

    公开(公告)号:US12120499B2

    公开(公告)日:2024-10-15

    申请号:US18486938

    申请日:2023-10-13

    Abstract: An audio system and method of spatially rendering audio signals that uses modified virtual speaker panning is disclosed. The audio system may include a fixed number F of virtual speakers, and the modified virtual speaker panning may dynamically select and use a subset P of the fixed virtual speakers. The subset P of virtual speakers may be selected using a low energy speaker detection and culling method, a source geometry-based culling method, or both. One or more processing blocks in the decoder/virtualizer may be bypassed based on the energy level of the associated audio signal or the location of the sound source relative to the user/listener, respectively. In some embodiments, a virtual speaker that is designated as an active virtual speaker at a first time, may also be designated as an active virtual speaker at a second time to ensure the processing completes.

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