Abstract: The present document describes a method (500) for providing combined audio and video content from a source device (210) to an audio sink device (230) and to a video sink device (220). The method (500) comprises determining (501) latency information regarding the video latency for processing the video content and the audio latency for processing the audio content of the combined audio and video content along the device chain (100) between the source device (210) and the audio and video sink devices (220, 230). Furthermore, the method (5009 comprises delaying (502) the audio content or the video content in dependence of the latency information, and providing (503) the combined audio and video content with the delayed audio content or the delayed video content.

Abstract: A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

Abstract: Methods for echo estimation or echo management (echo suppression or cancellation) on an input audio signal, with at least one of adaptation of a sparse prediction filter set, modification (for example, truncation) of adapted prediction filter impulse responses, generation of a composite impulse response from adapted prediction filter impulse responses, or use of echo estimation and/or echo management resources in a manner determined at least in part by classification of the input audio signal as being (or not being) echo free. Other aspects are systems configured to perform any embodiment of any of the methods.

Abstract: A standard dynamic range (SDR) image and a reference backward reshaping mapping are received. The reference backward reshaping mapping comprises a reference luma backward reshaping mapping. A color preservation mapping function is used with inputs generated from the SDR image and the reference backward reshaping mapping to determine luminance increase for SDR luma histogram bins generated based on luma codewords in the SDR image. A modified backward reshaping mapping is generated and comprises a modified luma backward reshaping mapping generated from the reference backward reshaping function based on the luminance increase for the SDR luma histogram bins. The SDR image and the modified backward reshaping mapping are encoded into an SDR video signal.

Abstract: The present document relates to audio source coding systems. In particular, the present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a first sample (615) of a first subband signal in a first subband of an audio signal is described. The first subband signal of the audio signal is determined using an analysis filterbank (612) comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the audio signal, respectively.

Abstract: Audio objects are associated with positional metadata. A received downmix signal comprises downmix channels that are linear combinations of one or more audio objects and are associated with respective positional locators. In a first aspect, the downmix signal, the positional metadata and frequency-dependent object gains are received. An audio object is reconstructed by applying the object gain to an upmix of the downmix signal in accordance with coefficients based on the positional metadata and the positional locators. In a second aspect, audio objects have been encoded together with at least one bed channel positioned at a positional locator of a corresponding downmix channel. The decoding system receives the downmix signal and the positional metadata of the audio objects. A bed channel is reconstructed by suppressing the content representing audio objects from the corresponding downmix channel on the basis of the positional locator of the corresponding downmix channel.

Abstract: Dual or multi-modulation display systems are disclosed that comprise projector systems with at least one modulator that may employ non-mechanical beam steering modulation. Many embodiments disclosed herein employ a non-mechanical beam steering and/or polarizer to provide for a highlights modulator.

Abstract: In a method to improve the coding efficiency of high-dynamic range (HDR) images, a decoder parses sequence processing set (SPS) data from an input coded bitstream to detect that an HDR extension syntax structure is present in the parsed SPS data. It extracts from the HDR extension syntax structure post-processing information that includes one or more of a color space enabled flag, a color enhancement enabled flag, an adaptive reshaping enabled flag, a dynamic range conversion flag, a color correction enabled flag, or an SDR viewable flag. It decodes the input bitstream to generate a preliminary output decoded signal, and generates a second output signal based on the preliminary output signal and the post-processing information.

Abstract: Based on viewing tracking data, a viewer's view direction to a three-dimensional (3D) scene depicted by a first video image is determined. The first video image has been streamed in a video stream to the streaming client device before the first time point and rendered with the streaming client device to the viewer at the first time point. Based on the viewer's view direction, a target view portion is identified in a second video image to be streamed in the video stream to the streaming client device to be rendered at a second time point subsequent to the first time point. The target view portion is encoded into the video stream with a higher target spatiotemporal resolution than that used to encode remaining non-target view portions in the second video image.

Abstract: A system is configured to receive player multimedia having live coverage of physical expressions or actions of a player in an online activity and activity multimedia of live participation in the online activity associated with one or more play accounts. In a near continuous fashion, the system is configured to select in real time, for each time point, one or more player items being portions of the player multimedia or one or more activity items being portions of the activity multimedia corresponding to the time point to form a composite item. The system is configured to further transmit in real time the composite item to one or more viewer accounts. The system is configured to then receive viewer data in response and produce future composite items based on the viewer data.

Abstract: A method (910) for rendering an audio signal in a virtual reality rendering environment (180) is described. The method (910) comprises rendering (911) an origin audio signal of an audio source (311, 312, 313) from an origin source position on an origin sphere (114) around an origin listening position (301) of a listener (181). Furthermore, the method (900) comprises determining (912) that the listener (181) moves from the origin listening position (301) to a destination listening position (302). In addition, the method (900) comprises determining (913) a destination source position of the audio source (311, 312, 313) on a destination sphere (114) around the destination listening position (302) based on the origin source position, and determining (914) a destination audio signal of the audio source (311, 312, 313) based on the origin audio signal.

Abstract: In one embodiment, a dual modulator display systems and methods for rendering target image data upon the dual modulator display system are disclosed where the display system receives target image data, possible HDR image data and first calculates display control signals and then calculates backlight control signals from the display control signals. This order of calculating display signals and then backlight control signals later as a function of the display systems may tend to reduce clipping artifacts. In other embodiments, it is possible to split the input target HDR image data into a base layer and a detail layer, wherein the base layer is the low spatial resolution image data that may be utilized as for backlight illumination data. The detail layer is higher spatial resolution image data that may be utilized for display control data.

Abstract: Example embodiments disclosed herein relate to orientation-aware surround sound playback. A method for processing audio on an electronic device that includes a plurality of loudspeakers is disclosed, the loudspeakers arranged in more than one dimension of the electronic device. The method includes, responsive to receipt of a plurality of received audio streams, generating a rendering component associated with the plurality of received audio streams, determining an orientation dependent component of the rendering component, processing the rendering component by updating the orientation dependent component according to an orientation of the loudspeakers and dispatching the received audio streams to the plurality of loudspeakers for playback based on the processed rendering component. Corresponding system and computer program products are also disclosed.

Abstract: A method for rendering an audio output based on an audio data stream including M audio signals, side information including a series of reconstruction instances of a reconstruction matrix C and first timing data, the side information allowing reconstruction of N audio objects from the M audio signals, and object metadata defining spatial relationships between the N audio objects. The method includes generating a synchronized rendering matrix based on the object metadata, the first timing data, and information relating to a current playback system configuration, the synchronized rendering matrix having a rendering instance for each reconstruction instance, multiplying each reconstruction instance with a corresponding rendering instance to form a corresponding instance of an integrated rendering matrix, and applying the integrated rendering matrix to the audio signals in order to render an audio output.

Abstract: A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.

Abstract: A control unit of a multipath data transportation system that optimizes the load of the multiple communication paths of this system when the system transmits a data segment over these paths in parallel with forward error correction. The control unit determines an optimized number of packets to send over each path based on a prediction of quality for each path. The transmitted packets include systematic packets and coded packets.

Abstract: In some embodiments, an encoder device is disclosed to receive an input video stream containing images in a first dynamic range including a first image. The device receives a second image representing the first image. The device obtains statistical data for the first and the second images. The device determines, at a first time delay, a scene cut data from the input video stream and storing the scene cut data in a first sliding window. The device determines, at a second time delay, a first smoothing mapping function based on a second sliding window and the determined scene cut data. The device determines, at a third time delay, a second smoothing mapping function based on a third sliding window and the determined scene cut data. The device generates, at the third time delay, a composer metadata for the first image based on the first and second smoothing mapping functions.

Abstract: Embodiments are described for a system of rendering object-based audio content through a system that includes individually addressable drivers, including at least one driver that is configured to project sound waves toward one or more surfaces within a listening environment for reflection to a listening area within the listening environment; a renderer configured to receive and process audio streams and one or more metadata sets associated with each of the audio streams and specifying a playback location of a respective audio stream; and a playback system coupled to the renderer and configured to render the audio streams to a plurality of audio feeds corresponding to the array of audio drivers in accordance with the one or more metadata sets.

Abstract: A method for decoding a video bitstream is disclosed. The method comprises: entropy decoding a first portion of a video bitstream, wherein first portion of video bitstream is associated with a video frame, thereby producing a first portion of decoded data; entropy decoding a second portion of video bitstream, wherein second portion of video bitstream is associated with video frame, thereby producing a second portion of decoded data, wherein entropy decoding second portion of video bitstream is independent of entropy decoding first portion of video bitstream; and reconstructing a first portion of video frame associated with video bitstream using first portion of decoded data and second portion of decoded data.

Abstract: A method for decoding video includes receiving quantized coefficients representative of a block of video representative of a plurality of pixels. The quantized coefficients are dequantized based upon a function of a remainder. The dequantized coefficients are inverse transformed to determine a decoded residue.

Abstract: A handheld imaging device has a data receiver that is configured to receive reference encoded image data. The data includes reference code values, which are encoded by an external coding system. The reference code values represent reference gray levels, which are being selected using a reference grayscale display function that is based on perceptual non-linearity of human vision adapted at different light levels to spatial frequencies. The imaging device also has a data converter that is configured to access a code mapping between the reference code values and device-specific code values of the imaging device. The device-specific code values are configured to produce gray levels that are specific to the imaging device. Based on the code mapping, the data converter is configured to transcode the reference encoded image data into device-specific image data, which is encoded with the device-specific code values.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: The present invention is directed to systems, methods and apparatus for processing media content for reproduction by a first apparatus. The method includes obtaining pose information indicative of a position and/or orientation of a user. The pose information is transmitted to a second apparatus that provides the media content. The media content is rendered based on the pose information to obtain rendered media content. The rendered media content is transmitted to the first apparatus for reproduction. The present invention may include a first apparatus for reproducing media content and a second apparatus storing the media content. The first apparatus is configured to obtain pose information indicative and transmit the pose information to the second apparatus; and the second apparatus is adapted to: render the media content based on the pose information to obtain rendered media content; and transmit the rendered media content to the first apparatus for reproduction.

Abstract: A system and method of adjusting an audio output in one location so that its propagation into another location is reduced. As a first device in a first location generates sound, a second device in a second location detects the propagated sound. The first device then adjusts its output based on the detected sound.

Abstract: In some embodiments, an encoder device is disclosed to generate single-channel standard dynamic range/high dynamic range content predictors. The device receives a standard dynamic range image content and a representation of a high dynamic range image content. The device determines a first mapping function to map the standard dynamic range image content to the high dynamic range image content. The device generates a single channel prediction metadata based on the first mapping function, such that a decoder device can subsequently render a predicted high dynamic range image content by applying the metadata to transform the standard dynamic range image content to the predicted high definition image content.

Abstract: The precision of up-sampling operations in a layered coding system is preserved when operating on video data with high bit-depth. In response to bit-depth requirements of the video coding or decoding system, scaling and rounding parameters are determined for a separable up-scaling filter. Input data are first filtered across a first spatial direction using a first rounding parameter to generate first up-sampled data. First intermediate data are generated by scaling the first up-sampled data using a first shift parameter. The intermediate data are then filtered across a second spatial direction using a second rounding parameter to generate second up-sampled data. Second intermediate data are generated by scaling the second up-sampled data using a second shift parameter. Final up-sampled data may be generated by clipping the second intermediate data.

Abstract: A method for interactive and user guided manipulation of multichannel audio content, the method including the steps of: providing a content preview facility for replay and review of multichannel audio content by a user; providing a user interface for the user selection of a segment of multichannel audio content having an unsatisfactory audio content; processing the audio content to include associated audio object activity spatial or signal space regions, to create a time line of activity where one or more spatial or signal space regions are active at any given time; matching the user's gesture input against at least one of the active spatial or signal space regions; signal processing the audio emanating from selected active spatial or signal space region using a number of differing techniques to determine at least one processed alternative; providing the user with an interactive playback facility to listen to the processed alternative.

Abstract: Computer-implemented methods for training a neural network, as well as for implementing audio encoders and decoders via trained neural networks, are provided. The neural network may receive an input audio signal, generate an encoded audio signal and decode the encoded audio signal. A loss function generating module may receive the decoded audio signal and a ground truth audio signal, and may generate a loss function value corresponding to the decoded audio signal. Generating the loss function value may involve applying a psychoacoustic model. The neural network may be trained based on the loss function value. The training may involve updating at least one weight of the neural network.

Abstract: The present document relates to a method of layered encoding of a compressed sound representation of a sound or sound field. The compressed sound representation comprises a basic compressed sound representation comprising a plurality of components, basic side information for decoding the basic compressed sound representation to a basic reconstructed sound representation of the sound or sound field, and enhancement side information including parameters for improving the basic reconstructed sound representation.

Abstract: A projector system comprising a laser light source, a collimating lens, a fly-eye lens, an integrating rod and a first modulator is disclosed. The light from a laser light source/fiber illuminates a collimator to substantially collimate the light and then is transmitted through a fly's-eye lens. The fly's-eye lens provides a desired angular/spatial light distribution for further processing to a first modulator of the projector system.

Abstract: A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag. The high frequency regeneration is performed as a post-processing operation with a delay of 3010 samples per audio channel.

Abstract: To support lossless mode at the block level when in-loop reshaping (LMCS) is enabled, the following changes are proposed to the existing LMCS pipeline. In intra mode, encode lossless blocks in the original domain, thus bypassing inverse mapping after reconstruction in the decoder. In inter mode, encode lossless blocks in the original domain, thus bypassing both forward mapping after motion compensation and inverse mapping after reconstruction in the decoder. In both modes, disable any LMCS-related color scaling.

Abstract: The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described.

Abstract: A system for decoding a video bitstream includes receiving a bitstream and a plurality of enhancement bitstreams together with receiving a video parameter set and a video parameter set extension. The system also receives an output layer set change message including information indicating a change in at least one output layer set.

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: The quantization parameter QP is well-known in digital video compression as an indication of picture quality. Digital symbols representing a moving image are quantized with a quantizing step that is a function QSN of the quantization parameter QP, which function QSN has been normalized to the most significant bit of the bit depth of the digital symbols. As a result, the effect of a given QP is essentially independent of bit depth a particular QP value has a standard effect on image quality, regardless of bit depth. The invention is useful, for example, in encoding and decoding at different bit depths, to generate compatible, bitstreams having different bit depths, and to allow different bit depths for different components of a video signal by compressing each with the same fidelity (i.e., the same QP).

Abstract: A method for generating metadata for use by a video decoder for displaying video content encoded by a video encoder includes: (1) accessing a target tone mapping curve; (2) accessing a decoder tone curve corresponding to a tone curve used by the video decoder for tone mapping the video content; (3) generating a plurality of parameters of a trim-pass function used by the video decoder to apply after applying the decoder tone curve to the video content, wherein the parameters of the trim-pass function are generated to approximate the target tone curve with the combination of the trim-pass function and the decoder tone curve, and (4) generating the metadata for use by the video decoder, including said plurality of parameters of the trim-pass function.

Abstract: Methods, processes, and systems are presented for inter-layer scaling and mapping to perform dynamic-range scalability in video coding using single-loop video decoders. Two approaches are proposed: a) A mapping function is applied to each of the color components, b) a mapping function is applied to the luma component and luma-based scaling is applied to the chroma components. Both schemes may be combined with existing schemes for in-loop reshaping, such as luma mapping and chroma residual scaling employed in the versatile video codec (VVC).

Abstract: While a viewer is viewing a first stereoscopic image comprising a first left image and a first right image, a left vergence angle of a left eye of a viewer and a right vergence angle of a right eye of the viewer are determined. A virtual object depth is determined based at least in part on (i) the left vergence angle of the left eye of the viewer and (ii) the right vergence angle of the right eye of the viewer. A second stereoscopic image comprising a second left image and a second right image for the viewer is rendered on one or more image displays. The second stereoscopic image is subsequent to the first stereoscopic image. The second stereoscopic image is projected from the one or more image displays to a virtual object plane at the virtual object depth.

Abstract: Embodiments relate to an audio processing unit that includes a bitstream payload deformatter and a decoding subsystem. The decoding subsystem is coupled to the bitstream payload deformatter and configured to decode at least a portion of a block of an encoded audio bitstream. The block includes a fill element with an identifier indicating a start of the fill element and fill data after the identifier. The fill data includes at least one flag identifying whether a base form of spectral band replication or an enhanced form of spectral band replication is to be performed on audio content of the block. The identifier is a three bit unsigned integer transmitted most significant bit first and having a value of 0x6.

Abstract: Methods, systems, and computer program products for network-based processing and distribution of multimedia content of a live performance are disclosed. In some implementations, recording devices can be configured to record a multimedia event (e.g., a musical performance). The recording devices can provide the recordings to a server while the event is ongoing. The server automatically synchronizes, mixes and masters the recordings. The server performs the automatic mixing and mastering using reference audio data previously captured during a rehearsal. The server streams the mastered recording to multiple end users through the Internet or other public or private network. The streaming can be live streaming.

Abstract: Methods and systems for controlling judder are disclosed. Judder can be introduced locally within a picture, to restore a judder feeling which is normally expected in films. Judder metadata can be generated based on the input frames. The judder metadata includes base frame rate, judder control rate and display parameters, and can be used to control judder for different applications.

Abstract: Systems, methods, and computer program products of audio processing based on Adaptive Intermediate Spatial Format (AISF) are described. The AISF is an extension to ISF that allows spatial resolution around an ISF ring to be adjusted dynamically with respect to content of incoming audio objects. An AISF encoder device adaptively warps each ISF ring during ISF encoding to adjust angular distance between objects, resulting in increase in uniformity of energy distribution around the ISF ring. At an AISF decoder device, matrices that decode sound positions to the output speaker take into account the warping that was performed at the AISF encoder device to reproduce the true positions of sound sources.

Abstract: A method (600) for decoding an encoded audio signal (102) is described. The encoded audio signal (102) comprises a sequence of frames. Furthermore, the encoded audio signal (102) is indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. Different subsets of DRC profiles from the plurality of DRC profiles are comprised within different frames of the sequence of frames, such that two or more frames of the sequence of frames jointly comprise the plurality of DRC profiles.

Abstract: Directional image sensor data may be acquired with one or more directional image sensors. A light source and illumination image may be generated based on the directional image sensor data. A number of operations may be caused to be performed for an image based at least in part on light source information in the light source image. The operations may include display management operations, device positional operations, augmented reality superimposition operations, ambient light control operations, etc.

Abstract: The invention discloses rendering sound field signals, such as Higher-Order Ambisonics (HOA), for arbitrary loudspeaker setups, where the rendering results in highly improved localization properties and is energy preserving. This is obtained by rendering an audio sound field representation for arbitrary spatial loudspeaker setups and/or by a a decoder that decodes based on a decode matrix (D). The decode matrix (D) is based on smoothing and scaling of a first decode matrix {circumflex over (D)} with smoothing coefficients. The first decode matrix {circumflex over (D)} is based on a mix matrix G and a mode matrix {tilde over (?)}, where the mix matrix G was determined based on L speakers and positions of a spherical modelling grid related to a HOA order N, and the mode matrix {tilde over (?)} was determined based on the spherical modelling grid and the HOA order N.

Abstract: Audio perception in local proximity to visual cues is provided. A device includes a video display, first row of audio transducers, and second row of audio transducers. The first and second rows can be vertically disposed above and below the video display. An audio transducer of the first row and an audio transducer of the second row form a column to produce, in concert, an audible signal. The perceived emanation of the audible signal is from a plane of the video display (e.g., a location of a visual cue) by weighing outputs of the audio transducers of the column. In certain embodiments, the audio transducers are spaced farther apart at a periphery for increased fidelity in a center portion of the plane and less fidelity at the periphery.

Abstract: An apparatus and method of blind detection of binauralized audio. If the input content is detected as binaural, a second binauralization may be avoided. In this manner, the user experience avoids audio artifacts introduced by multiple binauralizations.