Abstract: Methods, systems, and devices involving estimating pupil dilation or constriction of a person viewing displayed media content due to cognitive load, arousal, or engagement. Some embodiments involve obtaining ambient illuminance data corresponding to illuminance of ambient light in the vicinity of a person, obtaining display screen luminance data associated with a content luminance value of the displayed media content and with display screen brightness, obtaining instantaneous pupil size data corresponding to a pupil size of one or more of the person's pupils, estimating a light-induced pupil dilation or constriction caused by the illuminance of ambient light and the luminance of a display screen being viewed by the person, and estimating the pupil dilation or constriction caused by at least one of engagement, arousal, or cognitive load experienced by the person based, at least in part, on the pupil size data and the light-induced pupil dilation or constriction.

Abstract: An example approach relates to generating images of a first dynamic range from input images of a second dynamic range lower than the first dynamic range. A first array of co-occurrence statistics in connection with an input image of the second dynamic range is generated. A second array of reshaping function index values for selecting reshaping functions from among a plurality of candidate reshaping functions is generated. To generate a reshaped image of the first dynamic range, denoising operations are performed based at least in part on the first array of co-occurrence statistics, whereas reshaping operations are performed based at least in part on the second array of reshaping function index values.

Abstract: A method (1000) of generating at least two audio channels from audio in an immersive audio format comprising at least one height audio channel (1010) and at least two non-height audio channels (1050, 1100), for playing back the at least two audio channels with a non-immersive loudspeaker system of at least two audio loudspeakers (1,2) inside a vehicle. The method comprises applying (1500) a virtual height filter (1300) to the height channel (1010) for, when playing back the height channel by one of the loudspeakers, attenuating spectral components of the height channel directly emanating from said loudspeaker (1;2) and for amplifying spectral components of the height channel reflected from a roof or an area close to the roof inside the vehicle, to generate a virtual height filtered audio signal (1175) and mixing (1700) the virtual height filtered audio signal with at least one of the two non-height audio channels.

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 some embodiments, a method, comprises: dividing, using at least one processor, an audio input into speech and non-speech segments; for each frame in each non-speech segment, estimating, using the at least one processor, a time-varying noise spectrum of the non-speech segment; for each frame in each speech segment, estimating, using the at least one processor, speech spectrum of the speech segment; for each frame in each speech segment, identifying one or more non-speech frequency components in the speech spectrum; comparing the one or more non-speech frequency components with one or more corresponding frequency components in a plurality of estimated noise spectra and selecting the estimated noise spectrum from the plurality of estimated noise spectra based on a result of the comparing.

Abstract: Neural networks (NNs) are described for intra-coding in video coding when reference samples from neighbor blocks are missing. In one architecture, multiple different neural networks are being used for intra coding, each trained depending on the location of the missing samples. In another architecture, a two-network approach is proposed: one NN to adaptively generate reference samples using an array of multi-layer perceptrons (MLPs), and the second NN to generate predicted pixel values based on the concatenated output values of the MLPs. Intra-coding performance may also be improved using a separate post-processing neural network to take advantage of spatial pixel correlation.

Abstract: Systems and methods for streaming HDR/WCG video content. An example method includes receiving from a client device a request identifying HDR/WCG video content to be streamed. The method also includes generating a mapping Look-Up Table (LUT) based on metadata and display parameters. The metadata are extracted from the identified HDR/WCG video content, and the display parameters of the client device are retrieved from a display panel information database accessible to the cloud-based server. The method also includes encoding the mapping LUT based for transmission through a communication link between the cloud-based server and the client device.

Abstract: Systems and methods for force-controlled protection of a loudspeaker. One system includes a driver, an audio source that outputs an audio signal for playback by the driver, a power amplifier that provides power to the driver for playing back the audio signal, and a controller including an electronic processor. The controller is configured to receive the audio signal from the audio source, predict an amount of force exerted by the driver based on a voltage level of the audio signal, determine whether the predicted amount of force exceeds a threshold, and limit an amount of voltage provided to the power amplifier when the predicted amount of force exceeds the threshold.

Abstract: The disclosure relates to methods of processing audio scene information relating to a voxel-based audio scene represented by a two-dimensional projection map, for rendering audio from an audio source at a source location to a listener at a listener location. One such method includes obtaining first diffraction information relating to an acoustic path between the source location and the listener location, the listener location associated with a first voxel of the projection map; determining first direction information indicative of a first rendering direction based on the first diffraction information; retrieving prestored second diffraction information relating to an acoustic path between the source location and a second voxel of the projection map in a neighborhood of the first voxel; determining second direction information indicative of a second rendering direction based on the second diffraction information; and outputting the first direction information and the second direction information for rendering.

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: 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: The present disclosure relates to a method and system (1) for suppressing wind noise. The method comprises obtaining an input audio signal (100, 100?) comprising a plurality of consecutive audio signal segments (101, 102, 103, 101?, 102?, 103?) and suppressing wind noise in the input audio signal with a wind noise suppressor module (20) to generate a wind noise reduced audio signal. The method further comprises sing a neural network (10) trained to predict a set of gains for reducing noise in the input audio signal (100, 100?) given samples of the input audio signal (100, 100?), wherein a noise reduced audio signal is formed by applying said set of gains to the input audio signal (100, 100?) and mixing the wind noise reduced audio signal and the noise reduced audio signal with a mixer (30) to obtain an output audio signal with suppressed wind noise.

Abstract: The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular, a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described. The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of ?f.

Abstract: Described herein is a method of processing an audio signal using a neural network or using a first and a second neural network. Described is further a method of training said neural network or of jointly training a set of said first and said second neural network. Moreover, described is a method of obtaining and transmitting a latent feature space representation of a perceptual domain audio signal using a neural network and a method of obtaining an audio signal from a latent feature space representation of a perceptual domain audio signal using a neural network. Described are also respective apparatuses and computer program products.

Abstract: Described herein is a method of processing a 3D audio scene, including: receiving an original 3D audio scene, the original 3D audio scene including first audio data and first metadata for a plurality of first audio elements in 3D audio space; receiving information indicative of modifying the original 3D audio scene; obtaining one or more rendering parameters from the information; and rendering the original 3D audio scene based on the one or more rendering parameters to obtain a modified 3D audio scene. Further described are alternative methods of processing a 3D audio scene, as well as corresponding apparatuses and computer program products.

Abstract: Matching keypoint pairs are generated and identified in original and rectified image spaces between two input images. A teacher neural network is trained based at least in part on the pairs of matching keypoints in the rectified image space. A neural implicit morphing network is trained jointly with the teacher neural network based at least in part on the matching keypoint pairs in the original image spaces in which predictions outputted from the teacher neural network are used to compute a loss function designated to train the neural implicit morphing network. The neural implicit morphing network on its own, after training, is caused to output intermediate images in the view transition between the two input images.

Abstract: Methods and systems are described to encode an event stream. In addition, the device may receive an event stream. The device further may, encode the event stream into a plurality of triplane representations by, binning the events in the event stream into a plurality of event voxels, and encoding the plurality of event voxels into the plurality of triplane representations using an autoencoder.

Abstract: The present disclosure describes techniques for low-complexity split rendering based on a limited number of pre-renditions, thereby significantly reducing the required number of computations on the pre-rendering side, as well as the amount of transmitted metadata. The present disclosure further describes low complexity solutions for post-renderer corrections around one, two or three rotation axes, e.g., for deviations of yaw, pitch and roll.

Abstract: Disclosed is an encoding/decoding method and apparatus related to adaptive deblocking filtering. There is provided an image decoding method performing adaptive filtering in inter-prediction, the method including: reconstructing, from a bitstream, an image signal including a reference block on which block matching is performed in inter-prediction of a current block to be encoded; obtaining, from the bitstream, a flag indicating whether the reference block exists within a current picture where the current block is positioned; reconstructing the current block by using the reference block; adaptively applying an in-loop filter for the reconstructed current block based on the obtained flag; and storing the current block to which the in-loop filter is or is not applied in a decoded picture buffer (DPB).

Abstract: A computing device system including a computing device having a housing and electronic components disposed within the housing, where the electronic components include a controller, a memory, and a power source. A display screen is supported on the housing, and a socket extends into the housing. A removable speaker is selectively received within the socket, where the removable speaker includes a power source that is automatically charged when received within the socket.

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: Described is a method of modelling a Doppler effect when rendering audio content for a 6 degrees of freedom (6DoF) environment on a user side. In particular, the method may comprise obtaining first parameter values of one or more first parameters indicative of an allowable range of pitch factor modification values. The method may further comprise obtaining a second parameter value of a second parameter indicative of a desired strength of the to-be-modelled Doppler effect. The method may yet further comprise determining a pitch factor modification value based on a relative velocity between a listener and an audio source in the audio content, and the first and second parameter values, using a predefined pitch factor modification function. Particularly, the predefined pitch factor modification function may have the first and second parameters and may be a function for mapping relative velocities to pitch factor modification values.

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: Techniques for determining headtracking information are provided. In some embodiments, a method involves receiving, at a mobile device, motion information from a set of earbuds paired with the mobile device. The method may involve determining rotation information associated with rotation to be applied to a sound field in accordance with a headtracking mode of operation. In a first mode, the motion information received from the earbuds corresponds to the determined rotation information. In a second mode, the motion information received from the earbuds comprises initial rotation information, and the initial rotation information is combined with motion information obtained via sensors of the mobile device to determine the rotation information. In a third mode, the motion information received from the earbuds comprises motion sensor data, and the motion sensor data is combined with motion information obtained via sensors of the mobile device to determine the rotation information.

Abstract: Described herein is a method (30) of rendering an audio signal (17) for playback in an audio environment (27) defined by a target loudspeaker system (23), the audio signal (17) including audio data relating to an audio object and associated position data indicative of an object position. Method (30) includes the initial step (31) of receiving the audio signal (17). At step (32) loudspeaker layout data for the target loudspeaker system (23) is received. At step (33) control data is received that is indicative of a position modification to be applied to the audio object in the audio environment (27). At step (38) in response to the position data, loudspeaker layout data and control data, rendering modification data is generated. Finally, at step (39) the audio signal (17) is rendered with the rendering modification data to output the audio signal (17) with the audio object at a modified object position that is between loudspeakers within the audio environment (27).

Abstract: Devices, systems and methods are described for encoding blocks of audio having content into frames. Example methods include receiving an input signal that includes blocks of audio information, The blocks of audio information include a set of block groups for a respective frame. Some methods include obtaining, for each respective block group, a first measure of quality, and obtaining, for each respective block group, a second measure of quality. The first measure of quality indicates a cost associated with a merge of two or more blocks of audio information to form the respective block group. The second measure of quality indicates an estimated distortion associated with a merge of the two or more blocks of audio information to form the respective block group. The methods include merging, based on the first measure of quality and the second measure of quality, at least two block groups to generate an encoded signal.

Abstract: In some implementations, a method of encoding a low-frequency effect (LFE) channel comprises: receiving a time-domain LFE channel signal; filtering, using a low-pass filter, the time-domain LFE channel signal; converting the filtered time-domain LFE channel signal into a frequency-domain representation of the LFE channel signal that includes a number of coefficients representing a frequency spectrum of the LFE channel signal; arranging coefficients into a number of subband groups corresponding to different frequency bands of the LFE channel signal; quantizing coefficients in each subband group according to a frequency response curve of the low-pass filter; encoding the quantized coefficients in each subband group using an entropy coder tuned for the subband group; and generating a bitstream including the encoded quantized coefficients; and storing the bitstream on a storage device or streaming the bitstream to a downstream device.

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: A method, performed by a device with one or more microphones, for generating an encoded bitstream, the method comprising, capturing, by the one or more microphones, one or more audio signals, analyzing the captured audio signals to determine presence of a wake word, upon detecting presence of a wake word, setting a flag to indicate a speech recognition task is to be performed on the captured audio signals, encoding the captured audio signals, assembling the encoded audio signals and the flag into the encoded bitstream.

Abstract: Methods, systems, and computer program products for generating spatial metadata by performers are disclosed. The method includes receiving one or more audio essences and spatial metadata corresponding to one or more audio essences generated by at least one performer in an audio performance. The one or more audio essences and the spatial metadata are generated by the at least one performer simultaneously, and the spatial metadata includes spatial coordinates in a rendering coordinate space. The spatial coordinates are associated with the one or more audio essences and determined by the at least one performer. The method further includes spatially rendering the one or more audio essences to one or more immersive sound mixes according to the spatial metadata, wherein the one or more immersive sound mixes are provided to at least one of a local live audience, the at least one performer, a mixing engineer, or a remote live audience.

Abstract: This disclosure provides systems, methods and apparatuses for a wireless power transmission apparatus to detect a fault of the wireless power reception apparatus. Various implementations relate generally to fault detection and mitigation in a wireless power system. In some aspects, a transmission (TX) controller of the wireless power transmission apparatus can detect a fault in the wireless power reception apparatus based on power transfer measurements or calculations in the wireless power transmission apparatus. The techniques may enable the wireless power transmission apparatus to detect faults such as an open circuit in the wireless power reception apparatus or a thermostat failure, among other examples. During the transmission of wireless power, the wireless power transmission apparatus may detect the fault and cease the transmission of wireless power.

Abstract: Embodiments are disclosed for context aware capture and rendering. In an embodiment, an audio processing method comprises: capturing a multi-channel input audio signal; generating noise-reduced target sound events of interest and environment noise for each channel of the multi-channel input audio signal; determining an event type for rendering; selecting a rendering scheme based on the event type and a loudspeaker layout; and rendering a multichannel output audio signal using the selected rendering scheme.

Abstract: Some disclosed methods involve receiving, by a control system, location control data from a sound source as the sound source emits sound in a plurality of sound source locations within an audio environment. Some such methods involve receiving, by the control system, direction of arrival data from each audio device of a plurality of audio devices in the audio environment. In some examples, each audio device of the plurality of audio devices includes a microphone array and the direction of arrival data corresponding to microphone signals from microphone arrays responsive to sound emitted by the sound source in the plurality of sound source locations. Some such methods involve estimating, by the control system, sound source locations and audio device locations based, at least in part, on the location control data and the direction of arrival data.

Abstract: Some disclosed methods involve receiving sensor data, including microphone data from each of a plurality of devices in the environment, producing an input embedding vector corresponding to each sensor, producing a device-wise context vector corresponding to each device, obtaining ground truth data and comparing each device-wise context vector with the ground truth data, to produce a comparison result. The comparing may involve an attention-based process. Some disclosed methods involve generating one or more current output analytics tokens based, at least in part, on the comparison result and controlling the operation of at least one device based, at least in part, in the one or more current output analytics tokens. The controlling may involve controlling at least one of a speaker operation or a microphone operation.

Abstract: Sample data and metadata related to spatial regions in images may be received from a coded video signal. It is determined whether specific spatial regions in the images correspond to a specific region of luminance levels. In response to determining the specific spatial regions correspond to the specific region of luminance levels, signal processing and video compression operations are performed on sets of samples in the specific spatial regions. The signal processing and video compression operations are at least partially dependent on the specific region of luminance levels.

Abstract: A system and method determine a clock drift and a clock variance of each node in plural nodes of a time-sensitive Ethernet network. An accumulated clock offset along a time-sensitive network path in the time-sensitive network is determined based on the clock drifts and the clock variances. A guard band having a dynamic size is determined based on the accumulated clock offset. The times at which Ethernet frames are communicated through the nodes are restricted by communicating the guard band with the dynamic size to one or more of the nodes.

Abstract: A method (700) for rendering an audio signal of an audio source (211, 212, 213) in a virtual reality rendering environment (180) is described. The method (700) comprises determining (701) whether or not a directivity pattern (232) of the audio source (211, 212, 213) is to be taken into account for a listening situation of a listener (181) within the virtual reality rendering environment (180). Furthermore, the method (700) comprises rendering (702) an audio signal of the audio source (211, 212, 213) without taking into account the directivity pattern (232) of the audio source (211, 212, 213), if it is determined that the directivity pattern (232) of the audio source (211, 212, 213) is not to be taken into account for the listening situation of the listener (181).

Abstract: A method of coding at least one image comprising the steps of splitting the image into a plurality of blocks, of grouping said blocks into a predetermined number of subsets of blocks, of coding each of said subsets of blocks in parallel, the blocks of a subset considered being coded according to a predetermined sequential order of traversal. The coding step comprises, for a current block of a subset considered, the sub-step of predictive coding of said current block with respect to at least one previously coded and decoded block, and the sub-step of entropy coding of said current block on the basis of at least one probability of appearance of a symbol.

Abstract: A wireless power transmission device for a vehicle and a method are provided. The wireless power transmission device for a vehicle may transmit a signal for detection of a wireless power reception device by using a lower frequency band that is different from an operating frequency band used to control the vehicle. The wireless power transmission device may receive a response signal for the transmitted signal and a power control signal from the wireless power reception device. The wireless power transmission device may control an operating frequency or a voltage (or both) in the wireless power transmission device for the vehicle according to the power control signal. The wireless power transmission device may transmit wireless power to the wireless power reception device.

Abstract: Methods and systems are described for video coding and decoding using cross-component prediction tools. The proposed methods include: inter-luma based modeling for chroma, cross component residue prediction, and convolutional cross-component intra prediction.

Abstract: The present disclosure relates to a method and system for performing speech classification for an audio signal. The method comprises obtaining the audio signal comprising a sequence of audio frames and determining, for each audio frame, a first speech confidence metric using a first speech classifier. For each given audio frame of at least a subset of the sequence of audio frames the method comprises classifying each respective audio frame of a first context window associated with the given audio frame as a speech-frame or non-speech frame by comparing the first speech confidence metric of the respective audio frame to a first predetermined threshold, determining an adaptive threshold based on the number of speech frames of the first context window and determining a first binary speech classification indicator for the given audio frame based on the first speech confidence metric and the adaptive threshold.

Abstract: The invention proposes a method and a device for arithmetic encoding of a current spectral coefficient using preceding spectral coefficients. Said preceding spectral coefficients are already encoded and both, said preceding and current spectral coefficients, are comprised in one or more quantized spectra resulting from quantizing time-frequency-transform of video, audio or speech signal sample values.

Abstract: Systems, devices, and methods are described for determining a “coupled” pair of Left/Right ear Head Related Transfer Functions (HRTFs) that are adapted from an original pair of Left/Right ear HRTFs, wherein the inter-aural delay of the coupled HRTFs is formed using all-pass filters that provide the correct inter-aural delay at low frequencies. The all-pass filters are adapted to limit the inter-aural phase difference at high frequencies. Furthermore, a low-complexity process is described for rapid generation of suitable all-pass filters.

Abstract: In one example, a method of streaming a multiple perspective video of a scene includes generating two or more streams of decoding units (DUs) to encode the videos of different respective views of the scene by intra-coding a selected region of a respective frame, inter-coding other regions of the respective frame, and changing the selected region in accordance with a gradual decoding refresh period. The method also includes generating a stream of packets to carry the generated streams of DUs. The stream of packets includes source packets and parity packets configured to provide unequal error protection to different sets of the DUs based on the respective importance ranks assigned thereto. In some examples, the unequal error protection is implemented using rateless forward error correction coding and a greedy algorithm for optimally distributing the budget of parity packets among the various DUs.

Abstract: An aspect of the present disclosure relates to processing audio comprising decoding a first bitstream (b1) to obtain decoded immersive audio content (A), decoding a second bitstream (bp) to obtain pose information (P, V, V?) associated with a user of a lightweight processing device, determining a first head-pose (P?) based on the pose information, providing a downmix representation (Dmx) of the immersive audio content (A) corresponding to the first head pose (P?), rendering a set of binaural representations (BINn) of the immersive audio content (A), wherein the binaural representations correspond to a second set of head poses (Pn), computing reconstruction metadata (M) to enable reconstruction of the set of binaural representations from the downmix representation (Dmx), the metadata (M) including the first head pose (P?), and encoding the downmix representation (Dmx) and the reconstruction metadata (M) in a third bitstream (b2).

Abstract: A method for generating personalized head-related transfer functions (pHRTFs) for a user of a media playback device comprising acquiring a feature set, x, including anthropometric features acquired with an image capturing system, estimating an initial parameter set, y?, including pHRTF model parameters for the user based on a statistical relationship between the pHRTF model parameters, y, and the feature set, x, estimating a set of final model parameters, y?, and generating a set of pHRTFs from the set of final model parameters. The set of final model parameters are based on the initial parameter set, y?, a demographic prior distribution describing expected variation of pHRTF model parameters, and an accuracy prior distribution describing expected errors in the initial parameter set, y. The accuracy prior distribution is derived from accuracy statistics associated with the image capture system.

Abstract: Method for encoding scene-based audio is provided. In some implementations, the method involves determining, by an encoder, a spatial direction of a dominant sound component in a frame of an input audio signal. In some implementations, the method involves determining rotation parameters based on the determined spatial direction and a direction preference of a coding scheme to be used to encode the input audio signal. In some implementations, the method involves rotating sound components of the frame based on the rotation parameters such that, after being rotated, the dominant sound component has a spatial direction that aligns with the direction preference of the coding scheme. In some implementations, the method involves encoding the rotated sound components of the frame of the input audio signal using the coding scheme in connection with an indication of the rotation parameters or an indication of the spatial direction of the dominant sound component.

Abstract: Methods and systems are described for intra-prediction using template matching (TM) in video coding. The proposed methods include adaptive fusion when using template-based intra mode derivation using the most probable modes (TIMD), and fusion in intra mode prediction with template matching (Intra TMP).

Abstract: Disclosed are an apparatus and a method of encoding/decoding a video, particularly a method and an apparatus for storing a quantization parameter differential value in a largest coding unit (LCU) based on quadtree splitting and adaptively predicting a quantization parameter value based on context information on neighboring CUs. Quadtree-based quantization parameter encoding and decoding methods and apparatuses effectively show information on a block having a quantization parameter differential value based on splitting information on a CU and adaptively predict a quantization parameter value using context information including a block size, block partition and a quantization parameter of a neighboring CU.