Abstract: A speaker for transmitting reflected sound waves off an upper surface down to a listening environment, comprising: a cabinet containing a plurality of audio drivers, direct-firing drivers within the cabinet oriented to transmit sound along a horizontal axis substantially perpendicular to a front surface of the cabinet, and a pair of upward-firing slotted drivers placed proximate to ends of an top surface of the cabinet and oriented at an inclination angle relative to the horizontal axis. The slotted drivers are configured to create an overlapping reflected sound projection for high frequency sound when reflected down to a listening position located at a distance in front of the speaker pair. Such a speaker projects reflected sound that provides wider horizontal or side-to-side dispersion to better cover the listening area.

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

Abstract: Methods, systems, and computer program products for synchronizing audio signals captured by multiple independent devices during an audio event are described. Multiple recording devices, e.g. several smartphones, record the audio event. A computer system receives audio signals from the devices. The system determines a first delay between two audio signals based on cross-correlation of waveforms of the two audio signals. Subsequently, the system detects attacks that are present in each audio signal by computing the derivative of a respective envelope for each audio signal. The system determines a second delay between the two audio signals based on cross-correlation of attacks of the two audio signals. The system synchronizes the audio signals using the second delay upon determining that using the second delay improves sound quality over using the first delay.

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. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

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: Shaped glasses have curved surface lenses with spectrally complementary filters disposed thereon. The filters curved surface lenses are configured to compensate for wavelength shifts occurring due to viewing angles and other sources. Complementary images are projected for viewing through projection filters having passbands that pre-shift to compensate for subsequent wavelength shifts. At least one filter may have more than 3 primary passbands. For example, two filters include a first filter having passbands of low blue, high blue, low green, high green, and red, and a second filter having passbands of blue, green, and red. The additional passbands may be utilized to more closely match a color space and white point of a projector in which the filters are used. The shaped glasses and projection filters together may be utilized as a system for projecting and viewing 3D images.

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

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

Abstract: Spatial information that describes spatial locations of visual objects as in a three-dimensional (3D) image space as represented in one or more multi-view unlayered images is accessed. Based on the spatial information, a cinema image layer and one or more device image layers are generated from the one or more multi-view unlayered images. A multi-layer multi-view video signal comprising the cinema image layer and the device image layers is sent to downstream devices for rendering.

Abstract: An imaging system, comprising a pixel image sensor array disposed on a substrate and comprising a plurality of pixels, a multi-stage timer coupled to said pixel image sensor array for triggering exposures of said pixels, wherein the pixels are grouped into N subsets, and the timer triggers, for each subset, a sequence of at least two exposures of different capture duration of the pixels of said subset, wherein start times of the exposure sequences of the different subsets are temporally offset, at least one ADC coupled to said pixel image sensor array, which converts said exposures of said pixels to pixel digital values, a memory coupled to said at least one ADC to store said pixel digital values, and a logic circuit coupled to said memory to determine for each pixel of the image sensor array which of the corresponding stored pixel digital values to upload to a video frame.

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: Techniques are provided to encode and decode image data comprising a tone mapped (TM) image with HDR reconstruction data in the form of luminance ratios and color residual values. In an example embodiment, luminance ratio values and residual values in color channels of a color space are generated on an individual pixel basis based on a high dynamic range (HDR) image and a derivative tone-mapped (TM) image that comprises one or more color alterations that would not be recoverable from the TM image with a luminance ratio image. The TM image with HDR reconstruction data derived from the luminance ratio values and the color-channel residual values may be outputted in an image file to a downstream device, for example, for decoding, rendering, and/or storing. The image file may be decoded to generate a restored HDR image free of the color alterations.

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: Audio data in a first format may be processed to produce audio data in a second format, which may be a reduced or simplified version of the first format. A loudness correction process may produce loudness-corrected audio data in the second format. A first power of the audio data in the second format and a second power of the loudness-corrected audio data in the second format may be determined. A second-format loudness correction factor for the audio data in the second format may be based, at least in part, on a power ratio between the first power and the second power. A first-format loudness correction factor for the audio data in the first format may be based, at least in part, on the power ratio and a power relationship between the audio data in the first format and the audio data in the second format.

Abstract: A video content controller includes a memory and a microprocessor. The memory is configured to store non-transitory computer-readable instructions and video data representing a temporally-varying scene having a plurality of scene-regions. The microprocessor adapted to execute the instructions to (i) receive a current stream-segment of a video stream corresponding to a first scene-region of the plurality of scene-regions during a current playback-time interval, (ii) download, to a memory, a first video segment of a tagged scene-region within the first scene-region, (iii) receive a first display-region signal indicating at least one of a selection and a feature of the first scene-region, (iv) combine the current stream-segment and the first video segment as a video signal, and (vi) transmit, in response to the first display-region signal, the video signal to a display device.

Abstract: Methods for generating an object based audio program, renderable in a personalizable manner, and including a bed of speaker channels renderable in the absence of selection of other program content (e.g., to provide a default full range audio experience). Other embodiments include steps of delivering, decoding, and/or rendering such a program. Rendering of content of the bed, or of a selected mix of other content of the program, may provide an immersive experience. The program may include multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects), the bed of speaker channels, and other speaker channels. Another aspect is an audio processing unit (e.g., encoder or decoder) configured to perform, or which includes a buffer memory which stores at least one frame (or other segment) of an object based audio program (or bitstream thereof) generated in accordance with, any embodiment of the method.

Abstract: Some disclosed methods involve encoding or decoding directional audio data. Some encoding methods may involve receiving a mono audio signal corresponding to an audio object and a representation of a radiation pattern corresponding to the audio object. The radiation pattern may include sound levels corresponding to plurality of sample times, a plurality of frequency bands and a plurality of directions. The methods may involve encoding the mono audio signal and encoding the source radiation pattern to determine radiation pattern metadata. Encoding the radiation pattern may involve determining a spherical harmonic transform of the representation of the radiation pattern and compressing the spherical harmonic transform to obtain encoded radiation pattern metadata.

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: Improved methods and/or apparatus for decoding an encoded audio signal in soundfield format for L loudspeakers. The method and/or apparatus can render an Ambisonics format audio signal to 2D loudspeaker setup(s) based on a rendering matrix. The rendering matrix has elements based on loudspeaker positions and wherein the rendering matrix is determined based on weighting at least an element of a first matrix with a weighting factor g = 1 L . The first matrix is determined based on positions of the L loudspeakers and at least a virtual position of at least a virtual loudspeaker that is added to the positions of the L loudspeakers.

Abstract: A method for demosaicing an image captured by an image sensor includes (a) computing, for each of a first plurality of interpolated pixel-values and based on neighboring primary pixel-values, a respective first confidence value; (b) generating a first thresholded mapping from a first coordinate mapping by removing each interpolated pixel-value having a first confidence value less than a threshold value; (c) repeating steps of computing and generating for a second and third sub-plurality of sensor pixels to yield a second and a third thresholded mapping; (d) determining high-confidence array-coordinates as array-coordinates included in all thresholded mappings, remaining array-coordinates being low-confidence array-coordinates; (e) forming a refined image including, at each high-confidence array-coordinate, a respective pixel-value-triplet including one primary pixel-value and two interpolated pixel-values; and (f) filling the refined image by assigning, for each low-confidence array-coordinate, a pixel-value

Abstract: A method and an apparatus of encoding/decoding intra prediction mode using a plurality of candidate intra prediction modes are disclosed. The method includes deriving three candidate intra prediction modes about a current block and deriving an intra prediction mode of the current block.

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

Abstract: A 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: An electronic device for encoding a picture is described. The electronic device includes a processor and instructions stored in memory that are in electronic communication with the processor. The instructions are executable to encode a step-wise temporal sub-layer access (STSA) sample grouping. The instructions are further executable to send and/or store the STSA sample grouping.

Abstract: An input media signal that carries input media content is received. The input media content is used to generate output media content in an output media signal. It is determined whether identification-and-timing (IAT) data is to be authored for the output media content. In response to determining that the output IAT data is to be authored for the output media content, output IAT data is authored for the output media content. At least a part of the output IAT data for at least a part of the output media content is encoded, along with the part of the output media content, into the output media signal. In some example scenarios, this output media signal then contains the IAT data and other related data for synchronization of additional media content with the output media content in content rendering/presentation operations.

Abstract: A novel high efficiency image projection system includes a beam-steering modulator, an amplitude modulator, and a controller. In a particular embodiment the controller generates beam-steering drive values from image data and uses the beam-steering drive values to drive the beam-steering modulator. Additionally, the controller utilizes the beam-steering drive values to generate a lightfield simulation of a lightfield projected onto the amplitude modulator by the beam-steering modulator. The controller utilizes the lightfield simulation to generate amplitude drive values for driving the amplitude modulator in order to project a high quality version of the image described by the image data.

Abstract: Embodiments are directed to a method of rendering adaptive audio by receiving input audio comprising channel-based audio, audio objects, and dynamic objects, wherein the dynamic objects are classified as sets of low-priority dynamic objects and high-priority dynamic objects, rendering the channel-based audio, the audio objects, and the low-priority dynamic objects in a first rendering processor of an audio processing system, and rendering the high-priority dynamic objects in a second rendering processor of the audio processing system. The rendered audio is then subject to virtualization and post-processing steps for playback through soundbars and other similar limited height capable speakers.

Abstract: Motion characteristics related to foreground objects and background regions bordering the foreground objects in images are determined. A frame rate conversion (FRC)-related metadata portion is generated based on the motion characteristics. The FRC-related metadata portion is to be used for determining an optimal FRC operational mode with a downstream device for the images. The images are encoded into a video stream. The FRC-related metadata portion is encoded into the video stream as a part of image metadata. The video stream is caused to be transmitted to the downstream device.

Abstract: Video images are rendered in viewports of content viewers. Each content viewer views video images through a respective viewport in the viewports. Spatial locations in the video images to which foveal visions of the content viewers are directed are determined. ROIs in the video images are identified based on the spatial locations in the video images.

Abstract: A coded video sequence is received in a bitstream with a set of content scan adaptive metadata. It is ascertained if the set of content scan adaptive metadata is received. The set of content scan adaptive metadata includes: a maximum content light level parameter; a maximum frame average light level parameter. The maximum content light level parameter and maximum frame average light level parameter are both dependent on a scan type of the frames of the coded video sequence, the scan type being at least one of a progressive frame type, complimentary field pair type, macroblock-adaptive frame-field frame type, and individual field picture type.

Abstract: A spherical image of a spatial environment is received and contains spherically arranged pixel values indexed by a time value. The spherical image is represented in a content creation coordinate system in reference to a spatial position in the spatial environment. The spatial position is indexed by the time value. A spatial relationship is determined between the content creation coordinate system and a spherical image reference coordinate system. Based at least in part on the spatial relationship and the spherically arranged pixel values, spherical distributions of image metadata are determined for the spherical image.

Abstract: Methods and systems for gamut mapping are disclosed. Pixels of an image or points of a look-up-table can be gamut mapped in a multi-step iterative process, by generating a coarse gamut hull and calculating a value of a distance metric for out-of-gamut pixels or LUT points, and subsequently generating a fine gamut hull in the neighborhood of the coarse gamut hull points closest to the out-of-gamut pixel or LUT points under consideration. The out-of-gamut pixel or LUT point is gamut mapped based on the smallest distance metric value calculated.

Abstract: A method for encoding an input audio stream including the steps of obtaining a first playback stream presentation of the input audio stream intended for reproduction on a first audio reproduction system, obtaining a second playback stream presentation of the input audio stream intended for reproduction on a second audio reproduction system, determining a set of transform parameters suitable for transforming an intermediate playback stream presentation to an approximation of the second playback stream presentation, wherein the transform parameters are determined by minimization of a measure of a difference between the approximation of the second playback stream presentation and the second playback stream presentation, and encoding the first playback stream presentation and the set of transform parameters for transmission to a decoder.

Abstract: Methods, systems, and computer program products for automatically positioning a content capturing device are disclosed. A vehicle, e.g., an UAV, carries the content capturing device, e.g., a camcorder. The UAV can position the content capturing device at a best location for viewing a subject based on one or more audio or visual cues. The UAV can follow movement of the subject to achieve best audio or visual effect. In some implementations, a controller device carried by the subject can generate one or more signals for the UAV to follow. The controller device may be coupled to a microphone that records audio. The signals can be used to temporally synchronize video captured at the UAV and audio captured by the microphone.

Abstract: Systems and methods for overlaying a second image/video data onto a first image/video data are described herein. The first image/video data may be intended to be rendered on a display with certain characteristics—e.g., HDR, EDR, VDR or UHD capabilities. The second image/video data may comprise graphics, closed captioning, text, advertisement—or any data that may be desired to be overlaid and/or composited onto the first image/video data. The second image/video data may be appearance mapped according to the image statistics and/or characteristics of the first image/video data. In addition, such appearance mapping may be made according to the characteristics of the display that the composite data is to be rendered. Such appearance mapping is desired to render a composite data that is visually pleasing to a viewer, rendered upon a desired display.

Abstract: The present document describes a method (500) for providing combined audio and video content from a source device (210) to an audio sink device (230) and to a video sink device (220). The method (500) comprises determining (501) latency information regarding the video latency for processing the video content and the audio latency for processing the audio content of the combined audio and video content along the device chain (100) between the source device (210) and the audio and video sink devices (220, 230). Furthermore, the method (5009 comprises delaying (502) the audio content or the video content in dependence of the latency information, and providing (503) the combined audio and video content with the delayed audio content or the delayed video content.

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: 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: In a method to improve the coding efficiency of high-dynamic range (HDR) images, a decoder parses sequence processing set (SPS) data from an input coded bitstream to detect that an HDR extension syntax structure is present in the parsed SPS data. It extracts from the HDR extension syntax structure post-processing information that includes one or more of a color space enabled flag, a color enhancement enabled flag, an adaptive reshaping enabled flag, a dynamic range conversion flag, a color correction enabled flag, or an SDR viewable flag. It decodes the input bitstream to generate a preliminary output decoded signal, and generates a second output signal based on the preliminary output signal and the post-processing information.

Abstract: Dual or multi-modulation display systems are disclosed that comprise projector systems with at least one modulator that may employ non-mechanical beam steering modulation. Many embodiments disclosed herein employ a non-mechanical beam steering and/or polarizer to provide for a highlights modulator.

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: A standard dynamic range (SDR) image and a reference backward reshaping mapping are received. The reference backward reshaping mapping comprises a reference luma backward reshaping mapping. A color preservation mapping function is used with inputs generated from the SDR image and the reference backward reshaping mapping to determine luminance increase for SDR luma histogram bins generated based on luma codewords in the SDR image. A modified backward reshaping mapping is generated and comprises a modified luma backward reshaping mapping generated from the reference backward reshaping function based on the luminance increase for the SDR luma histogram bins. The SDR image and the modified backward reshaping mapping are encoded into an SDR video signal.

Abstract: Audio objects are associated with positional metadata. A received downmix signal comprises downmix channels that are linear combinations of one or more audio objects and are associated with respective positional locators. In a first aspect, the downmix signal, the positional metadata and frequency-dependent object gains are received. An audio object is reconstructed by applying the object gain to an upmix of the downmix signal in accordance with coefficients based on the positional metadata and the positional locators. In a second aspect, audio objects have been encoded together with at least one bed channel positioned at a positional locator of a corresponding downmix channel. The decoding system receives the downmix signal and the positional metadata of the audio objects. A bed channel is reconstructed by suppressing the content representing audio objects from the corresponding downmix channel on the basis of the positional locator of the corresponding downmix channel.

Abstract: Based on viewing tracking data, a viewer's view direction to a three-dimensional (3D) scene depicted by a first video image is determined. The first video image has been streamed in a video stream to the streaming client device before the first time point and rendered with the streaming client device to the viewer at the first time point. Based on the viewer's view direction, a target view portion is identified in a second video image to be streamed in the video stream to the streaming client device to be rendered at a second time point subsequent to the first time point. The target view portion is encoded into the video stream with a higher target spatiotemporal resolution than that used to encode remaining non-target view portions in the second video image.

Abstract: A system is configured to receive player multimedia having live coverage of physical expressions or actions of a player in an online activity and activity multimedia of live participation in the online activity associated with one or more play accounts. In a near continuous fashion, the system is configured to select in real time, for each time point, one or more player items being portions of the player multimedia or one or more activity items being portions of the activity multimedia corresponding to the time point to form a composite item. The system is configured to further transmit in real time the composite item to one or more viewer accounts. The system is configured to then receive viewer data in response and produce future composite items based on the viewer data.

Abstract: A method (910) for rendering an audio signal in a virtual reality rendering environment (180) is described. The method (910) comprises rendering (911) an origin audio signal of an audio source (311, 312, 313) from an origin source position on an origin sphere (114) around an origin listening position (301) of a listener (181). Furthermore, the method (900) comprises determining (912) that the listener (181) moves from the origin listening position (301) to a destination listening position (302). In addition, the method (900) comprises determining (913) a destination source position of the audio source (311, 312, 313) on a destination sphere (114) around the destination listening position (302) based on the origin source position, and determining (914) a destination audio signal of the audio source (311, 312, 313) based on the origin audio signal.

Abstract: A control unit of a multipath data transportation system that optimizes the load of the multiple communication paths of this system when the system transmits a data segment over these paths in parallel with forward error correction. The control unit determines an optimized number of packets to send over each path based on a prediction of quality for each path. The transmitted packets include systematic packets and coded packets.

Abstract: A method for rendering an audio output based on an audio data stream including M audio signals, side information including a series of reconstruction instances of a reconstruction matrix C and first timing data, the side information allowing reconstruction of N audio objects from the M audio signals, and object metadata defining spatial relationships between the N audio objects. The method includes generating a synchronized rendering matrix based on the object metadata, the first timing data, and information relating to a current playback system configuration, the synchronized rendering matrix having a rendering instance for each reconstruction instance, multiplying each reconstruction instance with a corresponding rendering instance to form a corresponding instance of an integrated rendering matrix, and applying the integrated rendering matrix to the audio signals in order to render an audio output.

Abstract: Example embodiments disclosed herein relate to orientation-aware surround sound playback. A method for processing audio on an electronic device that includes a plurality of loudspeakers is disclosed, the loudspeakers arranged in more than one dimension of the electronic device. The method includes, responsive to receipt of a plurality of received audio streams, generating a rendering component associated with the plurality of received audio streams, determining an orientation dependent component of the rendering component, processing the rendering component by updating the orientation dependent component according to an orientation of the loudspeakers and dispatching the received audio streams to the plurality of loudspeakers for playback based on the processed rendering component. Corresponding system and computer program products are also disclosed.