Abstract: An apparatus and method relating to use of a physical writing surface (132) during a videoconference or presentation. Snapshots of a whiteboard (132) are identified by applying a difference measure to the video data (e.g., as a way of comparing frames at different times). Audio captured by a microphone may be processed to generate textual data, wherein a portion of the textual data is associated with each snapshot. The writing surface may be identified (enrolled) using gestures. Image processing techniques may be used to transform views of a writing surface.

Abstract: A system for decoding a video bitstream includes receiving a frame of the video that includes at least one slice and at least one tile and where each of the at least one slice and the at least one tile are not all aligned with one another.

Abstract: An intra prediction mode encoding and decoding method, an image decoding device, and an image encoding device operate by deriving most probable modes (MPMs) from surrounding prediction units adjacent to a current prediction unit and deriving an intra prediction mode of the current prediction unit on the basis of an MPM flag indicating whether an MPM having the same prediction mode as the intra prediction mode of the current prediction unit exists among the derived MPMs.

Abstract: Methods, systems, and computer program products for rending an audio object having an apparent size are disclosed. An audio processing system receives audio panning data including a first grid mapping first virtual sound sources in a space and speaker positions to speaker gains. The first grid specifies first speaker gains of the first virtual sound sources in the space. The audio processing system determines a second grid of second virtual sound sources in the space, including mapping the first virtual sound sources into the second virtual sound sources of the second virtual sources. The audio processing system selects at least one of the first grid or second grid for rendering an audio object based on an apparent size of the audio object. The audio processing system renders the audio object based on the selected grid or grids.

Abstract: An integrated cinema amplifier comprises a power supply stage that distributes power over a plurality of channels for rendering immersive audio content in a surround sound listening environment. The amplifier automatically detects maximum and net power availability and requirements based on audio content by decoding audio metadata and dynamically adjusts gains to each channel or sets of channels based on content and operational/environmental conditions. A power supply stage provides power to drive a plurality of channels corresponding to speaker feeds to a plurality of speakers. The amplifier has a front panel having an LED array with each LED associated with a respective channel or group of channels of the multi-channel amplifier, and a control unit configured to light the LEDs according to display patterns based on operating status or error conditions of the amplifier.

Abstract: Embodiments are described for a method of rendering audio for playback through headphones comprising receiving digital audio content, receiving binaural rendering metadata generated by an authoring tool processing the received digital audio content, receiving playback metadata generated by a playback device, and combining the binaural rendering metadata and playback metadata to optimize playback of the digital audio content through the headphones.

Abstract: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder.

Abstract: Light from an array of laser light sources are spread to cover the modulating face of a DMD or other modulator. The spread may be performed, for example, by a varying curvature array of lenslets, each laser light directed at one of the lenslets. Light from neighboring and/or nearby light sources overlap at a modulator. The lasers are energized at different energy/brightness levels causing the light illuminating the modulator to itself be modulated (locally dimmed). The modulator then further modulates the locally dimmed lights to produce a desired image. A projector according to the invention may utilize, for example, a single modulator sequentially illuminated or separate primary color modulators simultaneously illuminated.

Abstract: The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

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

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

Abstract: Systems and methods of rendering DCI-compliant image data on Enhanced Dynamic Range (EDR) display systems are disclosed. One embodiment of an EDR projector system comprises a first modulator and a second modulator. One method for rendering DCI-compliant image data on an EDR projector system comprises: receiving input image data, said image data comprising a plurality of image formats; determining whether the input image data comprises DCI image data; if the input image data comprises DCI image data, then performing dynamic range (DR) processing on the DCI image data; and rendering the dynamic range processed DCI image data on the EDR projector system. One DR processing method is to set the first modulator to a desired luminance level—e.g., fully ON or a ratio of DCI max luminance to the EDR max luminance. In addition, a desired minimum level of luminance may be set for the EDR projector.

Abstract: An audio processing system (100) accepts an audio bitstream having one of a plurality of predefined audio frame rates. The system comprises a front-end component (110), which receives a variable number of quantized spectral components, corresponding to one audio frame in any of the predefined audio frame rates, and performs an inverse quantization according to predetermined, frequency-dependent quantization levels. The front-end component may be agnostic of the audio frame rate. The audio processing system further comprises a frequency-domain processing stage (120) and a sample rate converter (130), which provide a reconstructed audio signal sampled at a target sampling frequency independent of the audio frame rate. By its frame-rate adaptability, the system can be configured to operate frame-synchronously in parallel with a video processing system that accepts plural video frame rates.

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: Some disclosed teleconferencing methods may involve detecting a howl state during a teleconference. The teleconference may involve two or more teleconference client locations and a teleconference server. The teleconference server may be configured for providing full-duplex audio connectivity between the teleconference client locations. The howl state may be a state of acoustic feedback involving two or more teleconference devices in a teleconference client location. Detecting the howl state may involve an analysis of both spectral and temporal characteristics of teleconference audio data. Some disclosed teleconferencing methods may involve determining which client location is causing the howl state. Some such methods may involve mitigating the howl state and/or sending a howl state detection message.

Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: At a first time point, a first light capturing device at a first spatial location in a three-dimensional (3D) space captures first light rays from light sources located at designated spatial locations on a viewer device in the 3D space. At the first time point, a second light capturing device at a second spatial location in the 3D space captures second light rays from the light sources located at the designated spatial locations on the viewer device in the 3D space. Based on the first light rays captured by the first light capturing device and the second light rays captured by the second light capturing device, at least one of a spatial position and a spatial direction, at the first time point, of the viewer device is determined.

Abstract: The present disclosure relates to reverberation generation for headphone virtualization. A method of generating one or more components of a binaural room impulse response (BRIR) for headphone virtualization is described. In the method, directionally-controlled reflections are generated, wherein directionally-controlled reflections impart a desired perceptual cue to an audio input signal corresponding to a sound source location. Then at least the generated reflections are combined to obtain the one or more components of the BRIR. Corresponding system and computer program products are described as well.

Abstract: Bordering pixels delineating a texture hole region are identified in a target image. Depth values of the bordering pixels are automatically clustered into two depth value clusters. A specific estimation direction is selected from multiple candidate estimation directions for a texture hole pixel in a texture hole region. A depth value of the texture hole pixel is estimated by interpolating depth values of two bordering background pixels in the specific estimation direction. The estimated depth value is used to warp the texture hole pixel into a reference view represented by a temporal reference image. A pixel value of the texture hole pixel is predicted based on a reference pixel value of a reference pixel from the reference image to which the texture hole pixel is warped using the estimated depth value.

Abstract: Laser or narrow band light sources (e.g., red, green, and blue) are utilized to form left (e.g., R1, G1, B1) and right (e.g., R2, G2, B2) images of a 3D projection. Off-axis viewing of the projections which has the potential to cause crosstalk and/or loss of energy/brightness in any channel or color, is eliminated (or reduced to only highly oblique viewing angles) via the combined use of any of guard bands between light bands of adjacent channels, curvature of viewing filters, and selection of passband wavelengths that maximize usability of the passband as it “shifts” due to varying or increasing angles of off-axis viewing. Implemented with any number of light sources, the light sources selected may also be converted to showing 2D images where the additional light sources are utilized to affect a desirable increase in color gamut.

Abstract: Apparatuses and methods for data traffic management in multi-source content delivery are described. The apparatus includes a downloader and a controller. The downloader is coupled to servers via communication links. The controller is configured to determine initial download requests for the servers based on predetermined information about a quality of the links. The controller is also configured to send the initial download requests to the servers with the downloader. The controller is further configured to update the information about the quality of the communication links after the downloader receives data associated with a data file from the servers via the communication links. The controller is also configured to determine subsequent download requests for the servers based on the updated information about the quality of the communication links. The controller of further configured to send the subsequent download requests to the servers via the downloader.

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

Abstract: Methods and systems for reducing banding artifacts when displaying images are described. Identified image bands are filtered using an adaptive sparse finite response filter, where the tap-distance in the sparse filter is adapted according to an estimated width of each image band. Image debanding may be performed across multiple pixel orientations, such as rows, columns, a 45-degree angle, or a ?45-degree angle. Given a threshold to decide whether sparse filtering needs to be performed or not, an iterative debanding process is also proposed.

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

Abstract: A novel projection system includes a light source, a phase modulator, an amplitude modulator, and a controller having temporal lightfield simulation capabilities. The phase modulator spatially modulates a lightfield from the light source to generate an intermediate image on the amplitude modulator. The amplitude modulator spatially modulates the intermediate image to form a final image. The controller models the phase state of the phase modulator during transitions between phase modulator frames and generates lightfield simulations of the intermediate image during the transition. The controller utilizes the lightfield simulations to generate and provide sets of amplitude drive values to the amplitude modulator at a faster rate than that at which the phase modulator is capable of switching.

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: Methods and systems for performing at least one audio activity (e.g., conducting a phone call or playing music or other audio content) in an environment including by determining an estimated location of a user in the environment in response to sound uttered by the user (e.g., a voice command), and controlling the audio activity in response to determining the estimated user location. The environment may have zones which are indicated by a zone map and estimation of the user location may include estimating in which of the zones the user is located. The audio activity may be performed using microphones and loudspeakers which are implemented in or coupled to smart audio devices.

Abstract: Embodiments are described for a high-frequency waveguide that improves the performance of large-scale surround sound and immersive audio environments. A horn waveguide is configured to be asymmetric about one of a vertical axis and horizontal axis of the waveguide to form an asymmetric horn waveguide. A spherical enclosure surrounds the asymmetric horn waveguide to form a horn speaker, and a three-axis mounting system is configured to fix the horn speaker to one of a wall or ceiling surface of the venue, wherein the mounting system facilitates rotating the horn speaker to a location that provides maximum coverage of the venue within the passband of the asymmetric horn waveguide.

Abstract: Embodiments are directed to a speaker system that contains an array of multiple dispersion drivers that creates an expansive acoustic pattern to playback multi-channel audio content through a standalone speaker. The speaker system comprises an interface receiving stereo audio; an upmixer generating surround sound formatted audio from the stereo audio including one or more height channels; a virtualizer/downmixer component coupled to the upmixer and generating speaker feeds for two or more loudspeaker output sections, configured to play back the stereo audio, wherein each output section is further configured to play its own dedicated stereo audio signals; and a set of drivers each coupled to a respective output section and configured to project sound in at least two different dispersion patterns.

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

Abstract: Scenes in video images are identified based on image content of the video images. Regional cross sections of the video images are determined based on the scenes in the video images. Image portions of the video images in the regional cross sections are encoded into multiple video sub-streams at multiple different spatiotemporal resolutions. An overall video stream that includes the multiple video sub-streams is transmitted to a streaming client device.

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

Abstract: A method and apparatus for reconstructing N audio channels from M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal representing the M audio channels and decoding the encoded audio signal to obtain a frequency domain representation of the M audio channels. The method further includes extracting a parameter from the bitstream and reconstructing at least one of the N audio channels using the parameter. The parameter represents an angle between two signals, at least one of which is included in the M audio channels.

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

Abstract: The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor S.

Abstract: Improved tools for authoring and rendering audio reproduction data are provided. Some such authoring tools allow audio reproduction data to be generalized for a wide variety of reproduction environments. Audio reproduction data may be authored by creating metadata for audio objects. The metadata may be created with reference to speaker zones. During the rendering process, the audio reproduction data may be reproduced according to the reproduction speaker layout of a particular reproduction environment.

Abstract: Microphone signals are received from microphones of a computer device. Each microphone signal of the microphone signals is acquired by a respective microphone of the microphones. A previously unselected microphone is selected from the microphones as a reference microphone, which generates a reference microphone signal. An adaptive filter is used to create, based on microphone signals of the microphones other than the reference microphone, predicted microphone signals for the reference microphone. Based on the predicted microphone signals for the reference microphone, an enhanced microphone signal is outputted for the reference microphone. The enhanced microphone signal may be used as microphone signal for the reference microphone in subsequent audio processing operations.

Abstract: Methods for dialogue enhancing audio content, comprising providing a first audio signal presentation of the audio components, providing a second audio signal presentation, receiving a set of dialogue estimation parameters configured to enable estimation of dialogue components from the first audio signal presentation, applying said set of dialogue estimation parameters to said first audio signal presentation, to form a dialogue presentation of the dialogue components; and combining the dialogue presentation with said second audio signal presentation to form a dialogue enhanced audio signal presentation for reproduction on the second audio reproduction system, wherein at least one of said first and second audio signal presentation is a binaural audio signal presentation.

Abstract: Projection displays include a highlight projector and a main projector. Highlights projected by the highlight projector boost luminance in highlight areas of a base image projected by the main projector. Various highlight projectors are based on steerable beams, holographic projectors, and spatial light modulators. A Fourier transform component and a mask positioned on the Fourier plane thereof are used to attenuate or eliminate selected spatial frequencies, e.g., to increase peak luminance without raising the black level of the projected image.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding a stereo audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and a discrete representation of the stereo audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.

Abstract: A projection display system comprises a light source configured to emit a light in response to a content data; an optical modulator configured to modulate the light; and a controller configured to adjust a light level of the projection display system based on the content data and a metadata relating to a future frame, thereby to reduce a perceptibility of a visual artifact.

Abstract: Some implementations involve analyzing audio packets received during a time interval that corresponds with a conversation analysis segment to determine network jitter dynamics data and conversational interactivity data. The network jitter dynamics data may provide an indication of jitter in a network that relays the audio data packets. The conversational interactivity data may provide an indication of interactivity between participants of a conversation represented by the audio data. A jitter buffer size may be controlled according to the network jitter dynamics data and the conversational interactivity data. The time interval may include a plurality of talkspurts.

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

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

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

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

Abstract: Described herein is a method of decoding an audio or speech signal, the method including the steps of: (a) receiving, by a decoder, a coded bitstream including the audio or speech signal and conditioning information; (b) providing, by a bitstream decoder, decoded conditioning information in a format associated with a first bitrate; (c) converting, by a converter, the decoded conditioning information from the format associated with the first bitrate to a format associated with a second bitrate; and (d) providing, by a generative neural network, a reconstruction of the audio or speech signal according to a probabilistic model conditioned by the conditioning information in the format associated with the second bitrate. Described are further an apparatus for decoding an audio or speech signal, a respective encoder, a system of the encoder and the apparatus for decoding an audio or speech signal as well as a respective computer program product.

Abstract: Sampled data is packaged in checkerboard format for encoding and decoding. The sampled data may be quincunx sampled multi-image video data (e.g., 3D video or a multi-program stream), and the data may also be divided into sub-images of each image which are then multiplexed, or interleaved, in frames of a video stream to be encoded and then decoded using a standardized video encoder. A system for viewing may utilize a standard video decoder and a formatting device that de-interleaves the decoded sub-images of each frame reformats the images for a display device. A 3D video may be encoded using a most advantageous interleaving format such that a preferred quality and compression ratio is reached. In one embodiment, the invention includes a display device that accepts data in multiple formats.