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

Abstract: The present invention relates to a block partitioning structure in video coding technology, and a video encoding and decoding method and apparatus using the same, wherein the video encoding and decoding method includes the steps of acquiring quad-partitioning information of a block; acquiring bi-partitioning information of the block when the acquired quad-partitioning information of the block does not indicate four partitions; acquiring partitioning direction information for bi-partitioning of the block when the acquired bi-partitioning information of the block indicates two partitions; acquiring information on whether to perform any other type of partitioning, when the acquired bi-partitioning information of the block does not indicate two partitions; and acquiring additional information required for the any other type of partitioning, when the acquired information on whether to perform any other type of partitioning indicates that the any other type of partitioning is performed.

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

Abstract: A system and method of modifying a binaural signal using headtracking information. The system calculates a delay, a first filter response, and a second filter response, and applies these to the left and right components of the binaural signal according to the headtracking information. The system may also apply headtracking to parametric binaural signals. In this manner, headtracking may be applied to pre-rendered binaural audio.

Abstract: The invention improves HOA sound field representation compression and decompression. A decoder decodes compressed dominant directional signals and compressed residual component signals so as to provide decompressed dominant directional signals and decompressed time domain signals representing a residual HOA component in a spatial domain. A re-correlator re-correlates the decompressed time domain signals to obtain a corresponding reduced-order residual HOA component. A processor determines a decompressed residual HOA component based on the corresponding reduced-order residual HOA component, and determines predicted directional signals based on at least a parameter. The processor is further configured to determine an HOA sound field representation based on the decompressed dominant directional signals, the predicted directional signals, and the decompressed residual HOA component.

Abstract: An apparatus and method of interfacing between a source media device and a destination media device. A wireless module passes through an audio signal from the source device to an output device, and transmits a wireless signal to a wireless device that outputs the audio signal. In this manner, the number of devices used for the connections may be reduced.

Abstract: A technique including receiving and decoding a coded bitstream encoded with audio content including first audio objects corresponding to a first media content type of two consecutive media content types and second audio objects corresponding to a second media content type of the two consecutive media content types, and audio metadata corresponding to the audio content. The audio metadata including first and second audio object gains, for the first and second audio objects, generated in part based on a first fading curve of the first media content type and a second fading curve of the second media content type, respectively. The technique further includes applying the first and second audio object gains to the first and second audio objects, and rendering a sound field represented by the first audio object with the applied first audio object gain and the second audio object with the applied second audio object gain.

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

Abstract: Described is a method of hosting a teleconference among a plurality of client devices arranged in two or more acoustic spaces, each client device having an audio capturing capability and/or an audio rendering capability, the method comprising: grouping the plurality of client devices into two or more groups based on their belonging to respective acoustic spaces, receiving first audio streams from the plurality of client devices, generating second audio streams from the first audio streams for rendering by respective client devices among the plurality of client devices, based on the grouping of the plurality of client devices into the two or more groups, and outputting the generated second audio streams to respective client devices. Further described are corresponding computation devise, computer programs, and computer-readable storage media.

Abstract: A method for mitigating modal noise includes applying a time-varying mechanical force to a fiber segment of the multimode optical fiber in at least a first direction orthogonal to a fiber axis of the multimode optical fiber within the fiber segment. A modal-noise mitigator for a multimode optical fiber includes an actuator configured to apply a time-varying mechanical force to a fiber segment of the multimode optical fiber in at least a first direction orthogonal to a fiber axis of the multimode optical fiber within the fiber segment.

Abstract: Disclosed is a data transmission system that transmits data by using a relay. The relay selects a transmission terminal from among a plurality of terminals accessing a base station. A base station transmits base station data to the relay during a first time slot, and the transmission terminal transmits terminal data to the relay. The relay transmits terminal data to the base station during a second time slot, and transmits base station data to the transmission terminal.

Abstract: Overlapped block disparity estimation and compensation is described. Compensating for images with overlapped block disparity compensation (OBDC) involves determining if OBDC is enabled in a video bit stream, and determining if OBDC is enabled for one or more macroblocks that neighbor a first macroblock within the video bit stream. The neighboring macroblocks may be transform coded. If OBDC is enabled in the video bit stream and for the one or more neighboring macroblocks, predictions may be made for a region of the first macroblock that has an edge adjacent with the neighboring macroblocks. OBDC can be causally applied. Disparity compensation parameters or modes may be shared amongst views or layers. A variety of predictions may be used with causally-applied OBDC.

Abstract: Computer-implemented methods for speech synthesis are provided. A speech synthesizer may be trained to generate synthesized audio data that corresponds to words uttered by a source speaker according to speech characteristics of a target speaker. The speech synthesizer may be trained by time-stamped phoneme sequences, pitch contour data and speaker identification data. The speech synthesizer may include a voice modeling neural network and a conditioning neural network.

Abstract: There are two representations for Higher Order Ambisonics denoted HOA: spatial domain and coefficient domain. The invention generates from a coefficient domain representation a mixed spatial/coefficient domain representation, wherein the number of said HOA signals can be variable. An aspect of the invention further relates to methods and apparatus decoding multiplexed and perceptually encoded HOA signals, including transforming a vector of PCM encoded spatial domain signals of the HOA representation to a corresponding vector of coefficient domain signals by multiplying the vector of PCM encoded spatial domain signals with a transform matrix and de-normalizing the vector of PCM encoded and normalized coefficient domain signals, wherein said de-normalizing comprises.

Abstract: The present disclosure relates to a method of decoding audio scene content from a bitstream by a decoder that includes an audio renderer with one or more rendering tools.

Abstract: A digital PSF for use in a dual modulation display. The invention allows the use of less than optimal point spread (PSF) functions in the optics between the pre-modulator and primary modulator of a dual modulation projection system. This technique uses multiple halftones per frame in the pre-modulator synchronized with a modified bit sequence in the primary modulator to produce a compensation image that reduces the errors produced by the sub-optimal PSF. The invention includes the application to dual modulation and dual modulated 3D viewing systems.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: Methods, systems, and bitstream syntax are described for canvas size, single layer or multi-layer, scalable decoding, with support for regions of interest (ROI), using a decoder supporting reference picture resampling. Offset parameters for a region of interest in a current picture and offset parameters for an ROI in a reference picture are taken into consideration when computing scaling factors to apply reference picture resampling. Syntax elements for supporting ROI regions under reference picture resampling are also presented.

Abstract: The present disclosure relates to an apparatus for decoding an encoded Unified Audio and Speech stream. The apparatus comprises a core decoder for decoding the encoded Unified Audio and Speech stream. The core decoder includes a fast Fourier transform, FFT, module implementation based on a Cooley-Tuckey algorithm. The FFT module is configured to determine a discrete Fourier transform, DFT. Determining the DFT involves recursively breaking down the DFT into small FFTs based on the Cooley-Tucker algorithm and using radix-4 if a number of points of the FFT is a power of 4 and using mixed radix if the number is not a power of 4. Performing the small FFTs involves applying twiddle factors. Applying the twiddle factors involves referring to pre-computed values for the twiddle factors.

Abstract: Embodiments for a speaker system that produces a near-field sound pattern for rendering immersive audio content in a portable device. An array of drivers projects sound upwards from a top surface of the portable device to form upward-firing speakers; a set of speakers projects sound downwards from a bottom surface of the portable device to form downward-firing speakers. A decoder/renderer component receives immersive audio content, decodes height audio signals from the content and sends direct audio signals to the downward-firing speakers. A crossover performs a high-pass filter function to pass high frequency components of the decoded height audio signals to the upward-firing speakers and low frequency components of the decoded height audio signals to the downward-firing speakers.

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

Abstract: Methods and systems for frame rate scalability are described. Support is provided for input and output video sequences with variable frame rate and variable shutter angle across scenes, or for input video sequences with fixed input frame rate and input shutter angle, but allowing a decoder to generate a video output at a different output frame rate and shutter angle than the corresponding input values. Techniques allowing a decoder to decode more computationally-efficiently a specific backward compatible target frame rate and shutter angle among those allowed are also presented.

Abstract: Some implementations may involve receiving, via an interface system, personnel location data indicating a location of at least one person and receiving, from an orientation system, headset orientation data corresponding with the orientation of a headset. First environmental element location data, indicating a location of at least a first environmental element, may be determined. Based at least in part on the headset orientation data, the personnel location data and the first environmental element location data, headset coordinate locations of at least one person and at least the first environmental element in a headset coordinate system corresponding with the orientation of the headset may be determined. An apparatus may be caused to provide spatialization indications of the headset coordinate locations. Providing the spatialization indications may involve controlling a speaker system to provide environmental element sonification corresponding with at least the first environmental element location data.

Abstract: The method for decoding an intra-picture prediction mode includes the steps of: determining whether the intra-picture prediction mode of a current prediction unit is identical to a first intra-picture prediction mode candidate or a second intra-picture prediction mode candidate based on bit information; and when the intra-picture prediction mode of the current prediction unit is identical to the first intra-picture prediction mode candidate and/or to the second intra-picture prediction mode candidate, determining whether the first intra-picture prediction mode candidate or the second intra-picture prediction mode candidate is identical to the intra-picture prediction mode of the current prediction unit on the basis of additional bit information, and decoding the intra-picture prediction mode of the current prediction unit.

Abstract: Systems and methods to transmit data over multiple communication channels in parallel with forward error correction. Original packets are evenly distributed to the channels as the initial systematically channel-encoded packets. Subsequent channel-encoded packets are configured to be linearly independent of their base sets of channel-encoded packets, where a base set for a subsequent channel-encoded packet includes those scheduled to be transmitted before the subsequent packet in the same channel as the subsequent packet, and optionally one or more initial packets from other channels. The compositions of the sequences of the encoded packets can be predetermined without the content of the packets; and the channel-encoded packets can be generated from the original packets on-the-fly by the transmitters of the channels during transmission. When a sufficient number of packets have been received via the channels, a recipient may terminate their transmissions.

Abstract: An audio processing method may involve receiving media input audio data corresponding to a media stream and headphone microphone input audio data, determining a media audio gain for at least one of a plurality of frequency bands of the media input audio data and determining a headphone microphone audio gain for at least one of a plurality of frequency bands of the headphone microphone input audio data. Determining the headphone microphone audio gain may involve determining a feedback risk control value, for at least one of the plurality of frequency bands, corresponding to a risk of headphone feedback between at least one external microphone of a headphone microphone system and at least one headphone speaker and determining a headphone microphone audio gain that will mitigate actual or potential headphone feedback in at least one of the plurality of frequency bands, based at least partly upon the feedback risk control value.

Abstract: Described herein is a method for processing of media data for playback, wherein the method includes the steps of: (a) fetching, by a web proxy, from two or more media servers media data and a media manifest file including metadata information relating to the fetched media data, and merging, by the web proxy, the media data; (b) modifying, by said web proxy, the content of the media manifest file and/or the content of the media data; and (c) providing, by said web proxy, the media manifest file and the media data as modified in step (b) to a media retrieval element for receiving and processing the media manifest file and the media data for decoding or playback, wherein a localhost address is assigned to the web proxy and the web proxy acts as a server for said media retrieval element. Described are further a respective device and computer program product.

Abstract: An apparatus may include a housing adapted for at least partial insertion into a concha bowl of a human ear, at least one speaker residing in or on the housing, a control system residing in or on the housing and a positioning element attached to the housing. The control system may be configured for controlling the speaker and configured for radio frequency (RF) communication. The positioning element may be configured to fit at least partially inside a concha of the human ear and may be configured to retain the housing at least partially within the concha bowl. The positioning element may include one or more wires Control System configured for communication with the control system. The one or more wires may be configured for at receiving and/or transmitting RF radiation. In some examples, the positioning element may be, or may include, a concha lock. The positioning element may include a loop antenna.

Abstract: An apparatus and method of transmission control for an audio device. The audio device uses sources other than the microphone to determine nuisance, and uses this to calculate a gain as well as to make the transmit decision. Using the gain results in a more nuanced nuisance mitigation than using the transmit decision on its own.

Abstract: Apparatus and methods are provided that employ one or more of a variety of techniques for reducing the time required to display high resolution images on a high dynamic range display having a light source layer and a display layer. In one technique, the image resolution is reduced, an effective luminance pattern is determined for the reduced resolution image, and the resolution of the effective luminance pattern is then increased to the resolution of the display layer. In another technique, the light source layer's point spread function is decomposed into a plurality of components, and an effective luminance pattern is determined for each component. The effective luminance patterns are then combined to produce a total effective luminance pattern. Additional image display time reduction techniques are provided.

Abstract: One or more context aware processing parameters and an ambient audio stream are received. One or more sound characteristics associated with the ambient audio stream are identified using a machine learning model. One or more actions to perform are determined using the machine learning model and based on the one or more context aware processing parameters and the identified one or more sound characteristics. The one or more actions are performed.

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: An apparatus for processing an input audio signal relies on a cascade of filterbanks, the cascade having a synthesis filterbank for synthesizing an audio intermediate signal from the input audio signal, the input audio signal being represented by a plurality of first subband signals generated by an analysis filterbank, wherein a number of filterbank channels of the synthesis filterbank is smaller than a number of channels of the analysis filterbank. The apparatus furthermore has a further analysis filterbank for generating a plurality of second subband signals from the audio intermediate signal, wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank, so that a sampling rate of a subband signal of the plurality of second subband signals is different from a sampling rate of a first subband signal of the plurality of first subband signals.

Abstract: Given a sequence of images in a first codeword representation, methods, processes, and systems are presented for integrating reshaping into a next generation video codec for encoding and decoding the images, wherein reshaping allows part of the images to be coded in a second codeword representation which allows more efficient compression than using the first codeword representation. A variety of architectures are discussed, including: an out-of-loop reshaping architecture, an in-loop-for intra pictures only reshaping architecture, an in-loop architecture for prediction residuals, and a hybrid in-loop reshaping architecture. Syntax methods for signaling reshaping parameters, and image-encoding methods optimized with respect to reshaping are also presented.

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: A method of processing of a sequence of video frames from a camera capturing a writing surface for subsequent transmission to at least one of a remote videoconferencing client and a remote videoconferencing server. The method comprises receiving the sequence of video frames from the camera; and selecting an image area of interest in the video frames, comprising selecting one of a sub-area of the video frames and an entire area of the video frames. The method also comprises, for each current video frame of the sequence of video frames, generating a pen stroke mask by applying adaptive thresholding to the image area of interest. The method also comprises generating an output video frame using the pen stroke mask. Corresponding systems and computer readable media are disclosed.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. For coding, portions of the original HOA representation are predicted from the directional signal components. This prediction provides side information which is required for a corresponding decoding. By using some additional specific purpose bits, a known side information coding processing is improved in that the required number of bits for coding that side information is reduced on average.

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 an apparatus for decoding an encoded Unified Audio and Speech stream. The apparatus comprises a core decoder for decoding the encoded Unified Audio and Speech stream. The core decoder includes an upmixing unit adapted to perform mono to stereo upmixing. The upmixing unit includes a decorrelator unit D adapted to apply a decorrelation filter to an input signal. The decorrelator unit is adapted to determine filter coefficients for the decorrelation filter by referring to pre-computed values. The present disclosure further relates to a an apparatus for encoding a Unified Audio and Speech stream, as well as to corresponding methods and storage media.

Abstract: Creative intent input describing emotion expectations and narrative information relating to media content is received. Expected physiologically observable states relating to the media content are generated based on the creative intent input. An audiovisual content signal with the media content and media metadata comprising the physiologically observable states is provided to a playback apparatus. The audiovisual content signal causes the playback device to use physiological monitoring signals to determine, with respect to a viewer, assessed physiologically observable states relating to the media content and generate, based on the expected physiologically observable states and the assessed physiologically observable states, modified media content to be rendered to the viewer.

Abstract: Methods, processes, and systems are presented for combining signal reshaping (also referred to as luma mapping chroma residuals scaling) with local illumination compensation (LIC) in video coding. Examples and trade-offs when the LIC model parameters are computed in the original domain, the reshaped domain, or a mixed domain, are presented.

Abstract: Some disclosed methods involve multi-band bass management. Some such examples may involve applying multiple high-pass and low-pass filter frequencies for the purpose of bass input management. Some disclosed methods treat at least some low-frequency signals as audio objects that can be panned. Some disclosed methods involve panning low and high frequencies separately. Following high-pass rendering, a power audit may determine a low-frequency deficit factor that is to be reproduced by subwoofers or other low-frequency-capable loudspeakers.

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: An audio processing system and method which calculates, based on spatial metadata of the audio object, a panning coefficient for each of the audio objects in relation to each of a plurality of predefined channel coverage zones. Converts the audio signal into submixes in relation to the predefined channel coverage zones based on the calculated panning coefficients and the audio objects. Each of the submixes indicating a sum of components of the plurality of the audio objects in relation to one of the predefined channel coverage zones. Generating a submix gain by applying an audio processing to each of the submix and controls an object gain applied to each of the audio objects. The object gain being as a function of the panning coefficients for each of the audio objects and the submix gains in relation to each of the predefined channel coverage zones.

Abstract: Systems and methods for adjusting video processing curves are disclosed herein. In one embodiment a method for applying an adjustment to an original curve derived from a set of input image data is disclosed, comprising: receiving a set of input image data to be adjusted; calculating an original curve from the set of input image data; receiving an adjustment curve, the adjustment curve based upon a desired image parameter; and applying the adjustment curve to the original curve to produce a resulting curve. In another embodiment, a display management unit (DMU) comprising a processor is disclosed that, upon receiving a set of input image data, processes the original curve according to: calculating an original curve from the set of input image data; receiving an adjustment curve, the adjustment curve based upon a desired image parameter; and applying the adjustment curve to the original curve to produce a resulting curve.

Abstract: Peripheral-vision expanded images are streamed to a video streaming client. The peripheral-vision expanded images are generated from source images in reference to view directions of the viewer at respective time points. View direction data is collected and received in real time while the viewer is viewing display images derived from the peripheral-vision expanded images. A second peripheral-vision expanded image is generated from a second source image in reference to a second view direction of the viewer at a second time point. The second peripheral-vision expanded image has a focal-vision image portion covering the second view direction of the viewer and a peripheral-vision image portion outside the focal-vision image portion. The second peripheral-vision expanded image is transmitted to the video streaming client.

Abstract: A picture coding method of the present invention codes a picture signal and a ratio of a number of luminance pixels and a number of chrominance pixels for the picture signal, and then one coding method out of at least two coding methods is selected depending on the ratio. Next, data related to a picture size is coded in accordance with the selected coding method. The data related to the picture size indicates a size of the picture corresponding to the picture signal or an output area, which is a pixel area to be outputted in decoding in a whole pixel area coded in the picture signal coding.

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.