Abstract: Techniques and corresponding systems for estimating an audiogram for a user of a media playback device including obtaining user hearing threshold data for the user, sample hearing threshold data, at least one of sample calibration data and sample noise data, and determining an estimate of the audiogram for the user based on such data. Related techniques for estimating calibration data for a media playback device, as well as corresponding computing apparatus, computer programs, and computer-readable storage media are also described.

Abstract: Some disclosed methods involve: receiving an observation sequence including a plurality of extracted features, each of which corresponds to a sequential signal of a sequence of sequential signals; determining a lattice of posterior possibilities, the lattice including a possibility of each observation sequence corresponding to one label class of a plurality of label classes; and applying a loss function to the lattice of posterior possibilities according to ground truth values, where applying the loss function involves applying both sequential information and cluster boundary information. Some methods involve updating parameters for determining the lattice according to losses determined by the loss function and performing the foregoing operations until one or more convergence criteria are met.

Abstract: Methods, systems, and media for utilizing head tracking data are provided. In some embodiments, a method involves receiving, at each earbud of a pair of communicatively coupled earbuds, sensor data from one or more sensors. The method may involve determining, at each earbud of the pair of communicatively coupled earbuds, head orientation information. The method may involve transmitting the determined head orientation information between the pair of communicatively coupled earbuds such that a leader earbud transmits head orientation information determined by the leader earbud to a follower earbud. The method may involve synchronizing, at each earbud, the determined head orientation data based at least in part on timing information associated with a timestamp at which the head orientation information was transmitted. The method may involve utilizing the synchronized head orientation data to present audio content by each earbud of the pair of communicatively coupled earbuds.

Abstract: Systems, methods, and computer program products for performing gain control on audio signals are provided. An automatic gain control system obtains a downmixed audio signal of an audio signal to be encoded. The system determines that an overload condition has occurred for a frame of the downmixed audio signal. Responsive to the overload condition, the system determines a gain transition function for the frame, wherein the gain transition function is based at least on a gain transition step size. The system applies the gain transition function to the frame to generate a gain adjusted frame of the downmixed audio signal. The system provides the gain adjusted frame and information indicative of the gain transition function for encoding by an encoder.

Abstract: Light projection systems using white light illumination. One embodiment provides a projection system using white light illumination. The projection system includes an illumination assembly configured to receive a white light input. A prism is configured to separate the white light input into color light inputs, redirect the color light inputs to respective modulators, and combine modulated color light inputs from the respective modulators into a white light output. An optical filter is configured to spatially Fourier transform the white light output to generate a filtered white light output. A projection lens assembly is configured to project the filtered white light output.

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: Disclosed herein are techniques for processing streaming data. In some embodiments, the techniques involve obtaining input data representative of a frame of streaming data. The techniques may involve identifying a query transformation, a key transformation, and a value transformation based on the input data. The techniques may involve updating a query buffer, a key buffer, and a value buffer, such that the buffers are each configured to store parameters associated with previous frames of streaming data and the frame of streaming data. The techniques may involve retrieving one or more query frames from the query buffer. The techniques may involve determining a dot product of the query frames and frames in the key buffer to determine a set of weights. The techniques may involve determining a weighted sum between the set of weights and frames in the value buffer, and utilizing the weighted sum to generate a streaming attention vector.

Abstract: An encoder for video encoding and decoding with long term reference picture retention. The encoder generates an encoded bitstream for a compliant decoder. At the decoder, coded pictures are decoded using a list of references pictures, of which at least one is a long term reference picture. A first picture is decoded using the long term reference picture. A second picture is decoded without using the long term reference picture which is inactive but is retained in a buffer. A subsequent picture is then decoded again using the long term reference picture.

Abstract: The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length La, extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length Ls, generating a frame of the output signal.

Abstract: Methods, systems, bitstream syntax, and fixed-point implementations are described for loop filtering using neural networks in image and video processing. Given an input image, a hybrid luma-chroma filter is proposed, wherein luma and chroma components are first processed by a first neural network and the output of the first network is subsequently processed by separate luma and chroma subnetworks. Finally, the outputs of the separate luma and chroma subnetworks are concatenated to generate the filtered output of the input image. Computational efficient methods using CP-decomposition are also described. Methods indicating the position of the neural-net loop filter relatively to other filters, such as the adaptive loop filter (ALF) are also discussed.

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: 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: Methods, systems, and media for identifying source files are provided. In some embodiments, a method for identifying source files may involve receiving a target file. The method may further involve identifying, from a set of candidate source files, a subset of candidate source files by filtering the set of candidate source files based on metadata associated with each candidate source file. The method may further involve identifying a matching source file of the subset of candidate source files based on similarities of embeddings of candidate source files of the subset of candidate source files to embeddings associated with the target file.

Abstract: Enclosed are embodiments for very low bit rate scene-based audio (LBRSBA) coding with combined SPAR and DIRAC. In some embodiments, a method comprises: receiving scene based audio metadata; creating from the scene based audio metadata, Spatial Reconstruction (SPAR) metadata and Directional Audio Coding (DirAC) metadata; forming a group of SPAR metadata bands and a group of DirAC metadata bands; quantizing the group of SPAR metadata bands and the group of DirAC metadata bands; and sending to a decoder a first data frame including the quantized group of DirAC metadata bands and a first portion of the quantized group of SPAR metadata bands, and a second data frame following the first data frame, the second data frame including the quantized DirAC metadata bands and a second portion of the quantized group of SPAR metadata bands.

Abstract: A system utilizing a high throughput coding mode for CABAC in HEVC is described. The system may include an electronic device configured to obtain a block of data to be encoded using an arithmetic based encoder; to generate a sequence of syntax elements using the obtained block; to compare an Absolute-3 value of the sequence or a parameter associated with the Absolute-3 value to a preset value; and to convert the Absolute-3 value to a codeword using a first code or a second code that is different than the first code, according to a result of the comparison.

Abstract: A loudspeaker assembly including a wedge-shaped acoustic waveguide and an audio driver. The wedge-shaped acoustic waveguide includes a first face including a slot opening formed therein, a second face joined at an edge with the first face, the second face tilted relative to the first face by a first angle, and a surface disposed opposite the edge and connected between the first face and the second face. The audio driver is coupled to the second face such that the audio driver is tilted at the first angle relative to the first face. The first face is oriented to face in a first direction and the audio driver is oriented to face in a second direction. A pressure level of sound energy emitted by the audio driver is greater along the second direction than the first direction. A vehicle comprising: a passenger cabin, a pillar extending upwards at a first angle relative to the passenger cabin, and the loudspeaker assembly installed on the pillar.

Abstract: HDR color patches are sampled throughout an HDR color space parameterized by a parameter. Reference SDR color patches, input HDR color patches and reference HDR color patches are generated from the sampled HDR color patches. An optimization algorithm is executed to generate an optimized forward reshaping mapping and an optimized backward reshaping mapping. The optimized forward reshaping mapping is used to forward reshape input HDR images into forward reshaped SDR images, whereas the optimized backward reshaping mapping is used to backward reshape the forward reshaped SDR images into backward reshaped HDR images.

Abstract: Image encoding/decoding methods and apparatuses according to the present invention may generate a prediction block of a current block based on a pre-defined prediction mode, generate a transform block of the current block through predetermined transformation, and reconstruct the current block based on the prediction block and the transform block.

Abstract: Coding methods and apparatus for packing single-channel data into a multi-channel container, e.g., an MP4, TIFF, or JPEG container, to at least achieve good utilization of the container's data capacity. In some examples, a coding method comprises: converting a plurality of scalar values of a received data stream into a corresponding plurality of n-dimensional values, the converting being performed using a mapper; assigning each of the n-dimensional values as a pixel value to a respective pixel of a virtual-image frame, where n is an integer greater than one; and compressing the virtual-image frame according to a type of a container for image data. The mapper is configured to map a scalar value to a corresponding n-dimensional value based on a relationship represented by an n-dimensional curve or by a plurality of 2n-way tree partitions of n-dimensional space.

Abstract: A quantization parameter signaling mechanism for both SDR and HDR content in video coding is described using two approaches. The first approach is to send the user-defined QpC table directly in high level syntax. This leads to more flexible and efficient QP control for future codec development and video content coding. The second approach is to signal luma and chroma QPs independently. This approach eliminates the need for QpC tables and removes the dependency of chroma quantization parameter on luma QP.

Abstract: Methods, systems, and bitstream syntax are described for the carriage of neural network topology and parameters as related to neural-network-based post filtering (NNPF) in image and video coding. Examples of NNPF SEI messaging as applicable to the MPEG standards for coding video pictures are described at the sequence layer and at the picture layer.

Abstract: Enclosed are embodiments for audio processing that combines complementary aspects of Spatial Reconstruction (SPAR) and Directional Audio Coding (DirAC) technologies, including higher audio quality, reduced bitrate, input/output format flexibility and/or reduced computational complexity, to produce a codec (e.g., an Ambisonics codec) that has better overall performance than DirAC or SPAR codecs.

Abstract: This disclosure provides systems, methods and apparatuses for a conditional standby state in a wireless power system. A Power Receiver can communicate a request message to a Power Transmitter to initiate a conditional standby state. During the conditional standby state, the Power Receiver can temporarily discontinue communication, power harvesting, or other operations for a period of time referred to as a sleeping time. At expiration of the sleeping time, the Power Transmitter can wake the Power Receiver so that the Power Receiver can check status of a condition of the appliance. Based on the status of the condition, the Power Receiver might request a transition to the power state for an ON action or might request a further conditional standby state for an OFF action. The wireless power system can use the conditional standby state to implement conditional ON or OFF actions of an appliance.

Abstract: A system is programmed to build a machine learning model that comprises a series of masking blocks. Each masking block receives a certain feature vector of an audio segment. Each masking block comprises a first component that generates a first mask for extracting clean speech and a second component that generates a second mask for extracting residual speech masked by the first mask. Each masking block also generates a specific feature vector based on the first mask and the second mask, which becomes the certain feature vector for the next masking block. The second component, which may comprise a gated recurrent unit layer, is computationally less complex than the first component, which may comprise multiple convolutional layers. Furthermore, the system is programmed to receive an input feature vector of an input audio segment and execute the machine learning model to obtain an output feature vector of an output audio segment.

Abstract: Disclosed is a method for separating audio objects in a mixed audio signal, the mixed audio signal comprising a plurality of audio objects. Further disclosed is a computer-implemented method for training a sparse audio object separation model and a method for separating a sparse audio object from a mixed audio signal, the mixed audio signal comprising at least a sparse audio object, a non-sparse audio object, and at least one further audio object. Further disclosed is a computer-implemented method for processing audio based on a signal-to-noise ratio, SNR and a computer-implemented method for processing audio based on a scene environment classification. Disclosed is a non-transitory computer-readable medium and a system configured to perform one or more of the methods.

Abstract: The video encoding/decoding method and device according to the present invention can determine an intra prediction mode of a chroma block, specify a luminance area for inter-component reference of the chroma block, downsample the specified luminance area, derive parameters for the inter-component reference of the chroma block, apply the parameters to the downsampled luminance area, and thereby predict the chroma block.

Abstract: Systems, methods and computer program products for direction of arrival (DOA) based head-tracked split rendering and head-related transfer function (HRTF) personalization are described. Head-tracked audio rendering is split between two devices. A first device receives a main bitstream representation of encoded audio. A second device tracks head pose information. The first device decodes the main bitstream using a main decoder, and encodes the decoded bitstream into pre-rendered binaural signals and post-render metadata. The second device decodes the pre-rendered binaural signals and post-renderer metadata from the intermediate bitstream, and provides the decoded pre-rendered binaural signals and post-renderer metadata to a lightweight renderer. The lightweight renderer renders the pre-rendered binaural signals into binaural audio based on the post-renderer metadata, the head pose information, generic HRTF, and personalized HRTF.

Abstract: Methods and systems are described for video coding and decoding using intra affine prediction. Two or three control point best vectors (CPBV) to be used to derive motion vectors of affine predictions are generated based on creating for each CPBV a list of candidate best vectors, and selecting according to a criterion the two or three best ones. Methods to generate the list of the candidate best vectors include a first method based on neighboring coded units and a distance criterion among two consecutive best vectors in the list, and a second method based on template matching between a coded unit to be encoded or decoded and prior-decoded coded units in a reference area.

Abstract: Systems and methods for adjustable wall reflectance within content viewing environments. One example provides a viewing environment including a content display configured to provide video content, a wall portion having an electrophoretic display, and a controller connected to the electrophoretic display. The controller configured to adjust a reflectivity of the electrophoretic display.

Abstract: Methods, systems, and bitstream syntax are described for a file container that supports the storage and transmission of multi-plane images. Examples are provided for coding texture and opacity information using HEVC or VVC coding and the HEIF container. Examples of carrying coded MPI images according to V3C and an example HEIF-based player are also presented.

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: A method and system for processing an input audio mix, by extracting at least two source audio signals, each audio source signal representing a separate audio source, extracting audio mix information from the input audio mix, the audio mix information including at least one of audio mix semantic properties and audio mix signal properties, determining audio processing parameters based on the audio mix information, and processing the source audio signals based on the audio processing parameters to generate a processed audio mix. The processing of the sources can thus be based on properties of the input audio mix before source separation, thereby allowing a more automated source processing.

Abstract: This disclosure provides systems, methods and apparatuses for a safety mechanism and related operations in an appliance having a Power Receiver (PRx) of a wireless power system. Aspects of this disclosure enable the Power Receiver to determine the status of the disconnect switch based on an electrical measurement of a relay coil of the disconnect switch. The Power Receiver can verify that the disconnect switch is in the closed position before transitioning to a power state. In some aspects, the Power Receiver can determine whether a harvested bias power is sufficient to close the disconnect switch before transitioning to the power state.

Abstract: A method for enhancing audio signals is provided. In some implementations, the method involves (a) obtaining a training set comprising a plurality of training samples, each training sample comprising a distorted audio signal and a clean audio signal. In some implementations, the method involves (b), for a training sample of the plurality of training samples: obtaining a frequency-domain representation of the distorted audio signal; providing the frequency-domain representation to a convolutional neural network (CNN) comprising a plurality of convolutional layers and to a recurrent element, wherein an output of the recurrent element is provided to a subset of the plurality of convolutional layers; generating a predicted enhancement mask, wherein the CNN generates the predicted enhancement mask; generating a predicted enhanced audio signal based on the predicted enhancement mask; and updating weights associated with the CNN and the recurrent element based on the predicted enhanced audio signal.

Abstract: Some implementations involve determining a noise metric and/or a speech intelligibility metric and determining a compensation process corresponding to the noise metric and/or the speech intelligibility metric. The compensation process may involve altering a processing of audio data and/or applying a non-audio-based compensation method. In some examples, altering the processing of the audio data does not involve applying a broadband gain increase to the audio signals. Some examples involve applying the compensation process in an audio environment. Other examples involve determining compensation metadata corresponding to the compensation process and transmitting an encoded content stream that includes encoded compensation metadata, encoded video data and encoded audio data from a first device to one or more other devices.

Abstract: A Method and Apparatus for measurement and conditional reconfiguration in non-terrestrial network is provided. The method includes receiving a measurement configuration IE and a conditional reconfiguration IE, performing conditional reconfiguration evaluation based on the measurement configuration IE and a conditional reconfiguration IE, considering the first cell as a triggered cell and applying the stored condRRCReconfig IE of the first cell. In the evaluation, at least two comparisons are performed based on a distance hysteresis and distance between the terminal and one or more reference positions.

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: Described herein is a computer-implemented deep-learning-based system for determining an indication of an audio quality of an input audio frame. The system comprises at least one inception block configured to receive at least one representation of an input audio frame and to map the at least one representation of the input audio frame into a feature map; and at least one fully connected layer configured to receive a feature map corresponding to the at least one representation of the input audio frame from the at least one inception block, wherein the at least one fully connected layer is configured to determine the indication of the audio quality of the input audio frame. Described are further respective methods of operating and training said system.

Abstract: Given a sequence of images in a first codeword representation, methods, processes, and systems are presented for image reshaping using rate distortion optimization, wherein reshaping allows the images to be coded in a second codeword representation which allows more efficient compression than using the first codeword representation. Syntax methods for signaling reshaping parameters are also presented.

Abstract: A computer-implemented method for playback of game audio by use of representative audio objects at runtime, comprising frame-by-frame the steps of, identifying candidate game audio objects based on non-silent in-game audio emitters, calculating a perceptual importance of each of the candidate game objects based on associated metadata, and generating a pre-set number of representative gaming audio objects based on the perceptual importance for playback of the game audio scene at runtime.

Abstract: A projection system for pixel shifting comprising a light source configured to emit light and a spatial light modulator configured to receive the light and generate a modulated light. The spatial light modulator includes a plurality of micromirrors. The projection system includes a wobulation device configured to shift the modulated light by fractional pixels. The projection system includes a controller configured to, for each of a plurality of subperiods, control the light source to emit the light onto the spatial light modulator, and between each of the plurality of subperiods and with the wobulation device, shift the modulated light by a partial pixel distance greater than a half-pixel distance.

Abstract: A method of processing audio, comprising receiving audio data for playback on a playback device, the audio data including target audio components, receiving masking audio data, obtaining hearing loss data associated with the listener, obtaining playback level data, calculating a target excitation pattern based on the target audio components and the playback level data, calculating a masking excitation pattern based on the masking audio data, calculating a signal gain based on the target excitation pattern, the masking excitation pattern, and the hearing loss data, and applying the signal gain to the target audio components to produce compensated target audio components.

Abstract: Some methods may involve receiving one or more audio signals, making an estimation of the direct sound level for each loudspeaker of the set of loudspeakers in a listening area, making an estimation of the loudness of each loudspeaker in the listening area and rendering one or more audio signals for reproduction via the set of loudspeakers. The rendering may involve determining, for each audio signal, a relative activation of each loudspeaker based, at least in part, on the estimation of the direct sound level for each loudspeaker in the listening area. A combined level of activations across loudspeakers may be based, at least in part, on the estimation of the loudness of each loudspeaker in the listening area.

Abstract: Soundfield signals such as e.g. Ambisonics carry a representation of a desired sound field. The Ambisonics format is based on spherical harmonic decomposition of the soundfield, and Higher Order Ambisonics (HOA) uses spherical harmonics of at least 2nd order. However, commonly used loudspeaker setups are irregular and lead to problems in decoder design. A method for improved decoding an audio soundfield representation for audio playback comprises calculating a panning function (W) using a geometrical method based on the positions of a plurality of loudspeakers and a plurality of source directions, calculating a mode matrix (?) from the loudspeaker positions, calculating a pseudo-inverse mode matrix (?+) and decoding the audio soundfield representation. The decoding is based on a decode matrix (D) that is obtained from the panning function (W) and the pseudo-inverse mode matrix (?+).

Abstract: An aspect of the present disclosure relates to a method for synthesizing an audio waveform from a text description indicating a desired sound, the method comprising: determining a text embedding from the text description; determining an image embedding from the text embedding; synthesizing a spectrogram by inputting the image embedding to a generative model trained to synthesize a spectrogram given an input image embedding; and converting the synthesized spectrogram to the audio waveform.

Abstract: The present disclosure relates to a method and system for concealing a lost or corrupt frame of audio content in a sequence of audio frames The method comprises obtaining at least one first frame associated with the frame sequence and identifying whether a second frame, subsequent to the at least one first frame, is valid or invalid. When the second frame is identified as invalid, the method comprises generating a set of Time-Frequency (TF) tile values for a replacement frame to replace the second frame by applying a sinusoidal extension process to generate the set of TF tile values for the replacement frame or applying a linear prediction process to generate the set of TF tile values for the replacement frame based on a degree of tonality of at least one preceding frame.

Abstract: This disclosure provides systems, methods and apparatuses for a safety mechanism and related operations in an appliance having a Power Receiver (PRx) of a wireless power system. The safety mechanism includes a fuse configured to fail (referred to as “blowing the fuse”) when a fault condition (such as overvoltage or overcurrent) occurs. The fuse is designed to blow, thereby interrupting the power supply to the appliance. By blowing the fuse, the excess voltage or current is effectively disconnected in the appliance, preventing damage to the internal circuitry and other components of the appliance. The safety mechanism might include a fuse blowing switch controlled by a PRx controller and configured to cause the fuse to intentionally blow when the PRx controller detects a fault condition. An optional capacitance element might increase current through the fuse to enhance the protective capability of the fuse.

Abstract: Methods and systems are described for processing a stereo video stream with one or more post-processing factor that can adjust the sense of depth for a user of the stereo video stream in a head mounted display. In one embodiment, a decoder receives a stream that include the stereo video stream and disparity statistics for the stream. The decoder demultiplexes the stream to recover the stereo video stream and the disparity statistics. Using the disparity statistics, the decoder computes a post-processing factor that can be one or more of a scale and a shift factor. The decoder applies the post-processing factor to the decoder stereo stream, where the resulting stereo video stream is provides the desired level of sense of depth for the user of the head mounted display.

Abstract: Novel methods and systems for creating and using user profiles for dialog boost and sound equalizer adjustment to compensate for various ambient sound situations. For creating profiles when not at the ambient sound, a synthesized/pre-recorded ambient noise can be mixed with the media to simulate the noise conditions.

Abstract: An image encoding and decoding method comprises the steps of: determining an inter-component prediction mode of a chrominance component block on the basis of an inter-component prediction mode list and predetermined index information; determining a reference sample for inter-component prediction of the chrominance component block on the basis of the determined inter-component prediction mode; deriving the parameters of a linear prediction model by using the reference sample; and performing inter-component prediction for the chrominance component block by using the parameters of the linear prediction model.