Abstract: Techniques are described for addressing rolling shutter delay and in some cases rolling shutter delay and stabilization. Processing circuits may receive image content in overlapping portions of images, and may adjust the image content until there is overlap in the overlapping portions. Processing circuits may also receive information of deviation of the device from a common reference. Based on the overlapping image content, the deviation of the device from the common reference, and image content in non-overlapping portions, the processing circuits may determine mapping of coordinates to a rectangular mesh for generating an equirectangular image.

Abstract: A wearable display device is described that is connected to a host device. The wearable display device includes one or more sensors configured to generate eye pose data indicating a user's field of view, one or more displays, and one or more processors. The one or more processors are configured to output a representation of the eye pose data to the host device and extract one or more depth values for a rendered frame from depth data output by the host device. The rendered frame is generated using the eye pose data. The one or more processors are further configured to modify one or more pixel values of the rendered frame using the one or more depth values to generate a warped rendered frame and output, for display at the one or more displays, the warped rendered frame.

Abstract: Methods for generating encoded audio programs indicative of N channels of discontinuity-corrected, encoded audio content, including by applying discontinuity correction values to multi-channel audio content, and for rendering such a program (e.g., to generate a discontinuity-corrected M-channel mix of content indicated by the program). Other aspects are systems or devices (e.g., encoders or decoders, or rendering systems) configured to implement any of the methods.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (a) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: A method of encoding adaptive audio, comprising receiving N objects and associated spatial metadata that describes the continuing motion of these objects, and partitioning the audio into segments based on the spatial metadata. The method encodes adaptive audio having objects and channel beds by capturing a continuing motion of a number N objects in a time-varying matrix trajectory comprising a sequence of matrices, coding coefficients of the time-varying matrix trajectory in spatial metadata to be transmitted via a high-definition audio format for rendering the adaptive audio through a number M output channels, and segmenting the sequence of matrices into a plurality of sub-segments based on the spatial metadata, wherein the plurality of sub-segments are configured to facilitate coding of one or more characteristics of the adaptive audio.

Abstract: Methods for detecting whether a rendered version of a specified seamless connection (“SSC”) at a connection point between two audio segment sequences results in an audible discontinuity, and methods for analyzing at least one SSC between audio segment sequences to determine whether a renderable version of each SSC would have an audible discontinuity at the connection point when rendered, and in appropriate cases, for a SSC having a renderable version which is determined to have an audible discontinuity when rendered, correcting at least one audio segment of at least one segment sequence to be connected in accordance with the SSC in an effort to ensure that rendering of the SSC will result in seamless connection without an audible discontinuity. Other aspects are editing systems configured to implement any of the methods, and storage media and rendering systems which store audio data generated in accordance with any of the methods.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: Methods for generating encoded audio programs indicative of N channels of discontinuity-corrected, encoded audio content, including by applying discontinuity correction values to multi-channel audio content, and for rendering such a program (e.g., to generate a discontinuity-corrected M-channel mix of content indicated by the program). Other aspects are systems or devices (e.g., encoders or decoders, or rendering systems) configured to implement any of the methods.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: The disclosed embodiments illustrate method for rendering time-compressed multimedia content on a user-computing device. The method includes determining metadata for one or more frames in multimedia content based on each of one or more time-compression factors and one or more attributes of the multimedia content. Further, the determined metadata comprises a binary value associated with each of the one or more frames of the multimedia content. The method further includes transmitting the multimedia content associated with the determined metadata to the user-computing device, based at least on a time-compression factor in a user request received from the user-computing device. Further, the transmitted multimedia content is rendered on the user-computing device as the time-compressed multimedia content.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: Audio processing methods may involve receiving audio data corresponding to a plurality of audio channels. The audio data may include a frequency domain representation corresponding to filterbank coefficients of an audio encoding or processing system. A decorrelation process may be performed with the same filterbank coefficients used by the audio encoding or processing system. The decorrelation process may be performed without converting coefficients of the frequency domain representation to another frequency domain or time domain representation. The decorrelation process may involve selective or signal-adaptive decorrelation of specific channels and/or specific frequency bands. The decorrelation process may involve applying a decorrelation filter to a portion of the received audio data to produce filtered audio data. The decorrelation process may involve using a non-hierarchal mixer to combine a direct portion of the received audio data with the filtered audio data according to spatial parameters.

Abstract: Some audio processing methods may involve receiving audio data corresponding to a plurality of audio channels and determining audio characteristics of the audio data, which may include transient information. An amount of decorrelation for the audio data may be based, at least in part, on the audio characteristics. If a definite transient event is determined, a decorrelation process may be temporarily halted or slowed. Determining transient information may involve evaluating the likelihood and/or the severity of a transient event. In some implementations, determining transient information may involve evaluating a temporal power variation in the audio data. Explicit transient information may or may not be received with the audio data, depending on the implementation. Explicit transient information may include a transient control value corresponding to a definite transient event, a definite non-transient event or an intermediate transient control value.

Abstract: Methods which uses interpolated primitive matrices to decode encoded audio to recover (losslessly) content of a multichannel audio program and/or to recover at least one downmix of such content, and encoding methods for generating such encoded audio. In some embodiments, a decoder performs interpolation on a set of seed primitive matrices to determine interpolated matrices for use in rendering channels of the program. Other aspects are a system or device configured to implement any embodiment of the method.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: A method of decomposing a matrix of dimension L-by-N, where L is less than or equal to N, into a sequence of N-by-N unit primitive matrices and a permutation matrix comprising a sequence that is the product of the primitive matrices and the permutation matrix, containing L rows that are substantially close to the provided L-by-N matrix, where the choice of the permutation matrix and the indices of the non-trivial rows in the primitive matrices are chosen to limit the coefficient values in the primitive matrices.

Abstract: Audio characteristics of audio data corresponding to a plurality of audio channels may be determined. The audio characteristics may include spatial parameter data. Decorrelation filtering processes for the audio data may be based, at least in part, on the audio characteristics. The decorrelation filtering processes may cause a specific inter-decorrelation signal coherence (“IDC”) between channel-specific decorrelation signals for at least one pair of channels. The channel-specific decorrelation signals may be received and/or determined. Inter-channel coherence (“ICC”) between a plurality of audio channel pairs may be controlled. Controlling ICC may involve at receiving an ICC value and/or determining an ICC value based, at least partially, on the spatial parameter data. A set of IDC values may be based, at least partially, on the set of ICC values. A set of channel-specific decorrelation signals, corresponding with the set of IDC values, may be synthesized by performing operations on the filtered audio data.