Abstract: Novel methods and systems for coordinating sound to both internal and external speakers for a device is disclosed. The audio signal is distributed among the internal and external speakers and aligned so that signals going to the internal speakers are aligned with signals going to the external speakers.

Abstract: Systems and methods for performing color mapping operations. One system includes a processor to perform post-production editing of image data. The processor is configured to identify a first region of an image and identify a second region of the image. The first region includes a first white point having a first tone, and the second region includes a second white point having a second tone. The processor is further configured to determine a color mapping function based on the first tone, apply the color mapping function to the second region of the image, and generate an output image.

Abstract: A method and a device for encoding and decoding intra prediction are disclosed. An image decoding method for performing intra prediction comprises the steps of: receiving a bitstream including data on prediction modes of a current block and a block adjacent to the current block; extracting the data from the received bitstream so as to confirm the prediction mode of the adjacent block; determining whether a boundary pixel within the adjacent block can be used as a reference pixel for the current block in consideration of the prediction mode of the adjacent block; obtaining the reference pixel of the current block according to the determined result; generating a prediction block predicted in the frame on the basis of the obtained reference pixel; and decoding the current block by using the generated prediction block.

Abstract: A method for decoding an encoded audio signal is described. The encoded audio signal may comprise a sequence of frames and may be indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. Different subsets of DRC profiles may be comprised within different frames. The method may comprise determining a first rendering mode from the plurality of different rendering modes; determining one or more DRC profiles from a subset of DRC profiles comprised within a current frame; determining whether at least one of the DRC profiles is applicable to the first rendering mode; selecting a default DRC profile as a current DRC profile, if none of the DRC profiles is applicable to the first rendering mode; wherein definition data of the default DRC profile is known at a decoder; and decoding the current frame using the current DRC profile.

Abstract: The disclosure relates to a neural network based system for speech enhancement, comprising a generative network for generating an enhanced audio signal and a conditioning network for generating conditioning information for the generative network. The conditioning network comprises a plurality of layers and is configured to receive an audio signal as input: propagate the audio signal through the plurality of layers; and provide one or more first internal representations of the audio signal or processed versions thereof as the conditioning information, wherein the one or more first internal representations of the audio signal are extracted at respective layers of the conditioning network. The generative network is configured to receive a noise vector and the conditioning information as input; and generate the enhanced audio signal based on the noise vector and the conditioning information. The disclosure further relates to a method of training the system.

Abstract: Camera tracking data with respect to a camera operating in a 3D physical space is received. An image portion depicting one or more visual objects not physically present in the 3D physical space is generated using a camera perspective derived from the camera tracking data. The one or more visual objects is caused to be visually combined with the camera perspective into a personal image taken by the camera.

Abstract: Methods and systems for generating and using dynamic spatial metadata in image and video processing are described. In an encoder, in addition to global metadata, local, spatial metadata for two or more image regions or image objects are generated, smoothed, and embedded as spatial metadata values. In a decoder, the decoder can reconstruct the spatial metadata and use interpolation techniques to generate metadata for specific regions of interest. Examples of generating spatial metadata related to min, mid, and max luminance values in an image are provided.

Abstract: The present document relates to audio coding systems. In particular, the present document relates to efficient methods and systems for parametric multi-channel audio coding. An audio encoding system configured to generate a bitstream indicative of a downmix signal and spatial metadata for generating a multi-channel upmix signal from the downmix signal is described. The system comprises a downmix processing unit configured to generate the downmix signal from a multi-channel input signal; wherein the downmix signal comprises m channels and wherein the multi-channel input signal comprises n channels; n, m being integers with m<n. Furthermore, the system comprises a parameter processing unit configured to determine the spatial metadata from the multi-channel input signal.

Abstract: The present disclosure relates to the field of audio enhancement, and in particular to methods, devices and software for supervised training of a machine learning model, MLM, the MLM trained to enhance a degraded audio signal by calculating gains to be applied to frequency bands of the degraded audio signal. The present disclosure further relates to methods, devices and software for use of such a trained MLM.

Abstract: Methods are described to communicate source color volume information in a coded bitstream using SEI messaging. Such data include at least the minimum, maximum, and average luminance values in the source data plus optional data that may include the color volume x and y chromaticity coordinates for the input color primaries (e.g., red, green, and blue) of the source data, and the color x and y chromaticity coordinates for the color primaries corresponding to the minimum, average, and maximum luminance values in the source data. Messaging data signaling an active region in each picture may also be included.

Abstract: A method of controlling headphones having external microphone signal pass-through functionality may involve controlling a display to present a geometric shape on the display and receiving an indication of digit motion from a sensor system associated with the display. The sensor system may include a touch sensor system or a gesture sensor system. The indication may be an indication of a direction of digit motion relative to the display. The method may involve controlling the display to present a sequence of images indicating that the geometric shape either enlarges or contracts, depending on the direction of digit motion and changing a headphone transparency setting according to a current size of the geometric shape. The headphone transparency setting may correspond to an external microphone signal gain setting and/or a media signal gain setting of the headphones.

Abstract: Some methods may involve receiving a first content stream that includes first audio signals, rendering the first audio signals to produce first audio playback signals, generating first calibration signals, generating first modified audio playback signals by inserting the first calibration signals into the first audio playback signals, and causing a loudspeaker system to play back the first modified audio playback signals, to generate first audio device playback sound. The method(s) may involve receiving microphone signals corresponding to at least the first audio device playback sound and to second through Nth audio device playback sound corresponding to second through Nth modified audio playback signals (including second through Nth calibration signals) played back by second through Nth audio devices, extracting second through Nth calibration signals from the microphone signals and estimating at least one acoustic scene metric based, at least partly, on the second through Nth calibration signals.

Abstract: In a method to improve backwards compatibility when decoding high-dynamic range images coded in a wide color gamut (WCG) space which may not be compatible with legacy color spaces, hue and/or saturation values of images in an image database are computed for both a legacy color space (say, YCbCr-gamma) and a preferred WCG color space (say, IPT-PQ). Based on a cost function, a reshaped color space is computed so that the distance between the hue values in the legacy color space and rotated hue values in the preferred color space is minimized. HDR images are coded in the reshaped color space. Legacy devices can still decode standard dynamic range images assuming they are coded in the legacy color space, while updated devices can use color reshaping information to decode HDR images in the preferred color space at full dynamic range.

Abstract: A digital watermarking method including receiving, by an electronic processor, an original image signal containing a series of original visual images, where the original image signal encoded uses a perceptual quantizer (PQ) luminance level encoding transfer function resulting in PQ luminance steps within the original image signal, and where the PQ luminance steps have varying sizes across a luminance range. The method further includes receiving, by the electronic processor, a watermark image signal including a watermark, and adjusting the strength of the watermark by at least a first weighting factor that is a predetermined first number of PQ luminance steps.

Abstract: A method (200) of processing audio in an immersive audio format comprising at least one height audio channel, comprising: obtaining (250) two height audio signals from at least a portion of the at least one height audio channel: modifying (270) a relative phase between the two height audio signals in frequency bands in which phase differences are predominantly out of phase to obtain two phase modified height audio signals: and playing back (290) the processed audio comprising the two phase modified height audio signals with at least two audio loudspeakers. The phase differences occur as a result of having monaural signals emanated from two audio loudspeakers at one or more listening positions symmetrically off-center with respect to the at least two loudspeakers. laterally spaced with respect to each of said one or more listening positions. The method allows perception of sound height/elevation without using overhead speakers.

Abstract: Decoding of Ambisonics representations for a stereo loudspeaker setup is known for first-order Ambisonics audio signals. But such first-order Ambisonics approaches have either high negative side lobes or poor localisation in the frontal region. The invention deals with the processing for stereo decoders for higher-order Ambisonics HOA.

Abstract: Methods and systems for improving coding efficiency of video.

Abstract: Described is picture segmentation through columns and slices in video encoding and decoding. A video picture is divided into a plurality of columns, each column covering only a part of the video picture in a horizontal dimension. All coded tree blocks (“CTBs”) belonging to a slice may belong to one or more columns. The columns may be used to break the same or different prediction or in-loop filtering mechanisms of the video coding, and the CTB scan order used for encoding and/or decoding may be local to a column. Column widths may be indicated in a parameter set and/or may be adjusted at the slice level. At the decoder, column width may be parsed from the bitstream, and slice decoding may occur in one or more columns.

Abstract: A method for dereverberating audio signals is provided. In some implementations, the method involves obtaining a real acoustic impulse response (AIR); identifying a first portion of the real AIR corresponding to early reflections of a direct sound and a second portion of the real AIR that corresponding to late reflections of the direct sound; generating one or more synthesized AIRs by modifying the first portion of the real AIR and/or the second portion of the real AIR; and using the real AIR and the one or more synthesized AIRs to generate a plurality of training samples, each training sample comprising an input audio signal and a reverberated audio signal, wherein the reverberated audio signal is generated based on the input audio signal and one of the real AIR or one of the one or more synthesized AIRs, which plurality of training samples are used to train a machine learning model.

Abstract: A method of audio processing includes classifying an audio signal as noise or as non-noise using a first model. For a noise signal. the audio signal is classified as user-generated content (UGC) noise or as professionally-generated content (PGC) noise using a second model. For a non-noise signal or PGC noise. the audio signal is processed using a first audio processing process. For UGC noise. the audio signal is processed using a second audio processing process.

Abstract: Methods, systems, and bitstream syntax are described for the fusion of latent features in multi-level, end-to-end, neural networks used in image and video compression. The fused architecture may be static or dynamic based on image characteristics (e.g., natural images versus screen content images) or other coding parameters, such as bitrate constrains or rate-distortion optimization. A variety of multi-level fusion architectures are discussed.

Abstract: Methods are described to communicate source color volume information in a coded bitstream using SEI messaging. Such data include at least the minimum, maximum, and average luminance values in the source data plus optional data that may include the color volume x and y chromaticity coordinates for the input color primaries (e.g., red, green, and blue) of the source data, and the color x and y chromaticity coordinates for the color primaries corresponding to the minimum, average, and maximum luminance values in the source data. Messaging data signaling an active region in each picture may also be included.

Abstract: Described herein is a method of metadata-based dynamic processing of audio data for playback, the method including: receiving, by a decoder, a bitstream including audio data and metadata for dynamic loudness adjustment; decoding, by the decoder, the audio data and the metadata to obtain decoded audio data and the metadata; determining, by the decoder, from the metadata, one or more processing parameters for dynamic loudness adjustment based on a playback condition; applying the determined one or more processing parameters to the decoded audio data to obtain processed audio data; and outputting the processed audio data for playback. Described is further a method of encoding audio data and metadata for dynamic loudness adjustment into a bitstream. Moreover, described are a respective decoder and encoder, a respective system and computer program products.

Abstract: Methods and apparatus for transmission of volumetric images in the MPI format. According to an example embodiment, texture and alpha layers of multiplane images are packed, as tiles, into a sequence of video frames. The sequence of video frames is then compressed to generate a video bitstream, which is transmitted together with a metadata bitstream specifying at least the parameters of the packing arrangement for the tiles in the sequence of video frames. Example packing arrangements include various selectable spatial and temporal arrangements for texture layers, alpha layers, and camera views. In some examples, the metadata bitstream is implemented using a SEI message and includes parameters selected from the group consisting of a size of the reference view, the number of layers in the multiplane image, the number of simultaneous views, one or more characteristics of the packing arrangement, layer merging information, dynamic range adjustment information, and reference view information.

Abstract: Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.

Abstract: Methods and systems for precision rendering in display mapping using neural networks are described. Given an intensity input image, a sequence of neural networks comprising a pyramid-halving sub-network, a pyramid down-sampling sub-network, a pyramid-up-sampling sub-network, and a final-layer generation sub-network generate a base layer image and a detail layer image to be used in display mapping.

Abstract: Described herein is an audio decoder for decoding a bitstream of encoded audio data, wherein the bitstream of encoded audio data represents a sequence of audio sample values and comprises a plurality of frames, wherein each frame comprises associated encoded audio sample values, the audio decoder comprising: a determiner configured to determine whether a frame of the bitstream of encoded audio data is an immediate playout frame comprising encoded audio sample values associated with a current frame and additional information; and an initializer configured to initialize the decoder if the determiner determines that the frame is an immediate playout frame, wherein initializing the decoder comprises decoding the encoded audio sample values comprised by the additional information before decoding the encoded audio sample values associated with the current frame.

Abstract: The present document describes a method for generating a bitstream, wherein the bitstream comprises a sequence of superframes for a sequence of frames of an immersive audio signal. The method comprises, repeatedly for the sequence of superframes, inserting coded audio data for one or more frames of one or more downmix channel signals derived from the immersive audio signal, into data fields of a superframe; and inserting metadata for reconstructing one or more frames of the immersive audio signal from the coded audio data, into a metadata field of the superframe.

Abstract: Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent.

Abstract: A method for delivering media content to one or more clients over a distributed system is disclosed. The method may include generating a plurality of network-coded symbols from a plurality of original symbols representing a first media asset. The method may further include generating an original plurality of coded variants of the first media asset. The method may further include distributing a first coded variant of the original plurality of coded variants to a first cache on a first server device for storage in the first cache. The method may further include distributing a second coded variant of the original plurality of coded variants to a second cache on a second server device for storage in the second cache.

Abstract: An encoding device evaluates a plurality of processing and/or post-processing algorithms and/or methods to be applied to a video stream, and signals a selected method, algorithm, class or category of methods/algorithms either in an encoded bitstream or as side information related to the encoded bitstream. A decoding device or post-processor utilizes the signaled algorithm or selects an algorithm/method based on the signaled method or algorithm. The selection is based, for example, on availability of the algorithm/method at the decoder/post-processor and/or cost of implementation. The video stream may comprise, for example, downsampled multiplexed stereoscopic images and the selected algorithm may include any of upconversion and/or error correction techniques that contribute to a restoration of the downsampled images.

Abstract: The present document describes a method (900) for controlling access to a software asset of a software program, which is executed on an electronic device (110). The method (900) comprises, on one or more software provider servers (120), receiving (902) a request for authentication from a service provider server (210); upon successful authentication, providing (904) an authentication token to the service provider server (210); receiving (906) a request for a feature access token from the service provider server (210); and in reaction to the request, providing (908) the feature access token to the service provider server (210).

Abstract: A method for providing media content within a media distribution network. The method comprises transforming source media content into an interim format, thereby providing transformed content. Furthermore, the method comprises storing the transformed content on at least one core storage unit. In addition, the method comprises receiving a request for the source media content from a client. The method further comprises encoding the transformed content or intermediate coded content derived therefrom into encoded content suitable for transmission over a core network and/or an edge network, as well as sending the encoded content via the core network and/or the edge network to the client.

Abstract: The disclosed embodiments enable converting audio signals captured in various formats by various capture devices into a limited number of formats that can be processed by an audio codec (e.g., an Immersive Voice and Audio Services (IVAS) codec). In an embodiment, a simplification unit of the audio device receives an audio signal captured by one or more audio capture devices coupled to the audio device. The simplification unit determines whether the audio signal is in a format that is supported/not supported by an encoding unit of the audio device. Based on the determining, the simplification unit, converts the audio signal into a format that is supported by the encoding unit. In an embodiment, if the simplification unit determines that the audio signal is in a spatial format, the simplification unit can convert the audio signal into a spatial “mezzanine” format supported by the encoding.

Abstract: Systems and methods for preserving headphone rendering mode (HRM) in object clustering are described. In an embodiment, an object-based audio data processing system includes a processor configured to receive a plurality of audio objects, wherein an audio object of the plurality of audio objects is associated with respective object metadata that indicates respective spatial position information and an HRM; determine a plurality of cluster positions by applying an extended hybrid distance metric to a spatial coding algorithm to calculate a partial loudness for each of the audio objects; render the audio objects to the cluster positions to form a plurality of clusters by applying the extended hybrid distance metric to the spatial coding algorithm to calculate object-to-cluster gains; and transmit the clusters to a spatial reproduction system.

Abstract: A glass part to be made into a two-part freeform optical element is received. The glass part includes a precision freeform surface in a first specific spatial shape and a to-be-corrected surface. A non-glass material layer is added onto the to-be-corrected surface of the glass part to produce an accurate second surface in a second specific spatial shape. An optical coating is applied to the precision freeform surface of the glass part in the two-part freeform optical element.

Abstract: Multiple virtual source locations may be defined for a volume within which audio objects can move. A set-up process for rendering audio data may involve receiving reproduction speaker location data and pre-computing gain values for each of the virtual sources according to the reproduction speaker location data and each virtual source location. The gain values may be stored and used during “run time,” during which audio reproduction data are rendered for the speakers of the reproduction environment. During run time, for each audio object, contributions from virtual source locations within an area or volume defined by the audio object position data and the audio object size data may be computed. A set of gain values for each output channel of the reproduction environment may be computed based, at least in part, on the computed contributions. Each output channel may correspond to at least one reproduction speaker of the reproduction environment.

Abstract: Several embodiments of scalable image processing systems and methods are disclosed herein whereby color management processing of source image data to be displayed on a target display is changed according to varying levels of metadata.

Abstract: Individual loudspeaker dynamics processing configuration data, for each of a plurality of loudspeakers of a listening environment, may be obtained. Listening environment dynamics processing configuration data may be determined, based on the individual loudspeaker dynamics processing configuration data. Dynamics processing may be performed on received audio data based on the listening environment dynamics processing configuration data, to generate processed audio data. The processed audio data may be rendered for reproduction via a set of loudspeakers that includes at least some of the plurality of loudspeakers, to produce rendered audio signals. The rendered audio signals may be provided to, and reproduced by, the set of loudspeakers.

Abstract: Several embodiments of scalable image processing systems and methods are disclosed herein whereby color management processing of source image data to be displayed on a target display is changed according to varying levels of metadata.

Abstract: In a method to improve backwards compatibility when decoding high-dynamic range images coded in a wide color gamut (WCG) space which may not be compatible with legacy color spaces, hue and/or saturation values of images in an image database are computed for both a legacy color space (say, YCbCr-gamma) and a preferred WCG color space (say, IPT-PQ). Based on a cost function, a reshaped color space is computed so that the distance between the hue values in the legacy color space and rotated hue values in the preferred color space is minimized. HDR images are coded in the reshaped color space. Legacy devices can still decode standard dynamic range images assuming they are coded in the legacy color space, while updated devices can use color reshaping information to decode HDR images in the preferred color space at full dynamic range.

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: A speaker having a narrow aperture waveguide for transmitting sound from the rear side of a flat display panel. A transducer of the speaker radiates sound through the waveguide and outwards from the rear of the display panel and around an edge of the display panel to form soundwaves radiating directly or nearly directly to a listener positioned in front of the display panel. The waveguide includes fins that control the directivity of soundwaves exiting the waveguide and around the edge of the display panel.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency Q?+r?0 is generated on the basis of existing components at ? and ?+?0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.

Abstract: A method for multi-band ducking of audio signals is provided. In some implementations, the method involves receiving, at a decoder, an input audio signal, wherein the input audio signal is a downmixed audio signal. In some implementations, the method involves separating the input audio signal into a first set of frequency bands. In some implementations, the method involves determining a set of ducking gains, a ducking gain corresponding to a frequency band of the first set of frequency bands. In some implementations, the method involves generating a broadband decorrelated audio signal, wherein ducking gains of the set of ducking gains are applied to at least one of: 1) a second set of frequency bands prior to generating the at least one broadband decorrelated audio signal: or 2) a third set of frequency bands that separates the at least one broadband decorrelated audio signal.

Abstract: Embodiments feature families of rate allocation and rate control methods that utilize advanced processing of past and future frame/field picture statistics and are designed to operate with one or more coding passes. At least two method families include: a family of methods for a rate allocation with picture look-ahead; and a family of methods for average bit rate (ABR) control methods. At least two other methods for each method family are described. For the first family of methods, some methods may involve intra rate control. For the second family of methods, some methods may involve high complexity ABR control and/or low complexity ABR control. These and other embodiments can involve any of the following: spatial coding parameter adaptation, coding prediction, complexity processing, complexity estimation, complexity filtering, bit rate considerations, quality considerations, coding parameter allocation, and/or hierarchical prediction structures, among others.

Abstract: A method for generating a bitstream indicative of an object based audio program is described. The bitstream comprises a sequence of containers. A first container of the sequence of containers comprises a plurality of substream entities for a plurality of substreams of the object based audio program and a presentation section. The method comprises determining a set of object channels. The method further comprises providing a set of object related metadata for the set of object channels. In addition, the method comprises inserting a first set of object channel frames and a first set of object related metadata frames into a respective set of substream entities of the first container. Furthermore, the method comprises inserting presentation data into the presentation section.