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: 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: 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: 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: 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: Methods are described to communicate source color volume information in a coded bitstream using SEI messaging. Such data include at least the minimum, maximum, and average luminance values in the source data plus optional data that may include the color volume x and y chromaticity coordinates for the input color primaries (e.g., red, green, and blue) of the source data, and the color x and y chromaticity coordinates for the color primaries corresponding to the minimum, average, and maximum luminance values in the source data. Messaging data signaling an active region in each picture may also be included.

Abstract: A backward reshaping mapping table is initially generated as an inverse of a forward reshaping mapping table. The backward reshaping mapping table is updated by replacing the content-mapped luminance codewords with forward reshaped luminance codewords generated by applying a luminance forward mapping to the sampled luminance codewords. The luminance forward mapping is constructed from the forward reshaping mapping table. The backward reshaping mapping table and the luminance forward mapping are used to generate backward reshaping mappings for creating a reconstructed image from a forward reshaped image. The forward reshaped image is encoded, in a video signal, along with image metadata specifying the backward reshaping mappings. A recipient device of the video signal applies the backward reshaping mappings to the forward reshaped image to create the reconstructed image of the second dynamic range.

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: Methods and systems to adjust differently brightness and contrast for dark and bright pictures on a display are provided. Given a tone-mapping curve mapping an input dynamic range to a display comprising a minimum and maximum display luminance value, the maximum display luminance value is lowered to an adjusted luminance value according to user defined parameters. The input dynamic range is tone-mapped to the display dynamic range using the adjusted luminance value. For brightness control, the tone mapped image is stretched linearly back to the maximum display luminance value. For contrast control, a gamma or power EOTF of the display is adjusted according to the adjusted luminance. For displays with global backlight control, the global backlight is adjusted only when contrast is adjusted.

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

Abstract: A method for generating loudspeaker signals associated with a target screen size is disclosed. The method includes receiving a bit stream containing encoded higher order ambisonics signals, the encoded higher order ambisonics signals describing a sound field associated with a production screen size. The method further includes decoding the encoded higher order ambisonics signals to obtain a first set of decoded higher order ambisonics signals representing dominant components of the sound field and a second set of decoded higher order ambisonics signals representing ambient components of the sound field. The method also includes combining the first set of decoded higher order ambisonics signals and the second set of decoded higher order ambisonics signals to produce a combined set of decoded higher order ambisonics signals.

Abstract: Dual or multi-modulation display system 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: 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: 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: A projection system and calibration method therefor relate to a light source configured to emit a light in response to an image data, an illumination optical system configured to steer the light, the illumination optical system including a first minor and a second minor, a digital micromirror device (DMD) including a plurality of micromirrors respectively configured to reflect the steered light to a filter as on-state light or to reflect the steered light as off-state light to a light dump; determining a deviation between an actual angle of orientation and an expected angle of orientation of the DMD; calculating a first amount of angle adjustment corresponding to the first minor and a second amount of angle adjustment corresponding to the second mirror; and actuating the first mirror according to the first amount and the second mirror according to the second amount.

Abstract: The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described.

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

Abstract: In an encoder, a high-dynamic range (HDR) image is encoded using a family of local forward reshaping functions selected according to an array of forward mapping indices (FMI) indicating which local forward reshaping function needs to be used for each pixel in the HDR image to generate a reshaped standard dynamic range (SDR) image. A decoder, given the reshaped SDR image, iteratively generates a reconstructed HDR image and estimated reshaped SDR images by adjusting a local FMI array and a local array of backward mapping indices (BMI) until an error metric related to the difference between the local BMI and FMI arrays and the difference between the estimate SDR images and the reshaped SDR image satisfy a convergence criterion. Techniques for generating families of local forward reshaping functions and local backward reshaping functions based on a global forward reshaping function are also presented.

Abstract: A method and apparatus therefor comprises: receiving an image data and a power metadata, wherein the power metadata includes information relating to a power consumption or an expected power consumption; determining, based on the power metadata, an amount and a duration of a drive modification that may be performed by a target display in response to the power consumption or the expected power consumption; and performing a power management of the target display based on the power metadata to modify a driving of at least one light-emitting element associated with the target display relative to a manufacturer-determined threshold, based on a result of the determining, wherein the power metadata includes at least one of a temporal luminance energy metadata, a spatial luminance energy metadata, a spatial temporal fluctuation metadata, or combinations thereof.

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

Abstract: 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: 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: 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, methods, and computer program products for echo cancellation with prediction filter adaptation and detection of wideband offset between a reference signal (available to the echo canceller) and an output signal (unavailable to the echo canceller), where the output signal has been generated by applying at least one level shift to the reference signal, e.g. such that the level shift is unknown to the echo canceller.

Abstract: A method of audio processing includes calculating an estimated echo residual power based on a long-term amount of echo cancellation as indicated by an estimated power gain profile, where the estimated power gain profile is based on a reference signal and an echo residual signal. Echo suppression of the echo residual may then be performed based on the estimated echo residual power.

Abstract: Metadata and methods for variable-frame rate (VFR) video playback are presented. Proposed metadata include syntax parameters related to the presentation time duration, picture source type (e.g., original, duplicate, or interpolated), picture position in a scene (e.g., first, last, or in the middle), and motion-related information with respect to a previous picture. A decoder may use these metadata to apply appropriate frame-rate conversion techniques to reduce artifacts during VFR playback.

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: 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: 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: The application relates to HFR (High Frequency Reconstruction/Regeneration) of audio signals. In particular, the application relates to a method and system for performing HFR of audio signals having large variations in energy level across the low frequency range which is used to reconstruct the high frequencies of the audio signal. A system configured to generate a plurality of high frequency subband signals covering a high frequency interval from a plurality of low frequency subband signals is described.

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: Methods and apparatus for decoding a compressed Higher Order Ambisonics (HOA) representation of a sound or soundfield. The method may include receiving a bit stream containing the compressed HOA representation and decoding, based on a determination that there are multiple layers, the compressed HOA representation from the bitstream to obtain a sequence of decoded HOA representations. A first subset of the sequence of decoded HOA representations is determined based only on corresponding ambient HOA components. A second subset of the sequence of decoded HOA representations is determined based on corresponding ambient HOA components and corresponding predominant sound components.

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 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: In a method to improve backwards compatibility when decoding high-dynamic range images coded in a wide color gamut (WCG) space which may not be compatible with legacy color spaces, hue and/or saturation values of images in an image database are computed for both a legacy color space (say, YCbCr-gamma) and a preferred WCG color space (say, IPT-PQ). Based on a cost function, a reshaped color space is computed so that the distance between the hue values in the legacy color space and rotated hue values in the preferred color space is minimized HDR images are coded in the reshaped color space. Legacy devices can still decode standard dynamic range images assuming they are coded in the legacy color space, while updated devices can use color reshaping information to decode HDR images in the preferred color space at full dynamic range.

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: 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 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: 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 method of double talk detection includes using up-sampling. Audio signals received from the far end are up-sampled prior to output by the loudspeaker at the near end. The microphone at the near end captures audio at the up-sampled rate, and the audio output by the loudspeaker is detectable due to having no energy in the up-sampled frequency bands. The double talk detector uses this information to generate a signal for suppressing the echo of the far end audio from the captured audio signal that is transmitted to the far end.