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: The present document describes a method (700) for encoding a multi-channel input signal (201). The method (700) comprises determining (701) a plurality of downmix channel signals (203) from the multi-channel input signal (201) and performing (702) energy compaction of the plurality of downmix channel signals (203) to provide a plurality of compacted channel signals (404). Furthermore, the method (700) comprises determining (703) joint coding metadata (205) based on the plurality of compacted channel signals (404) and based on the multi-channel input signal (201), wherein the joint coding metadata (205) is such that it allows upmixing of the plurality of compacted channel signals (404) to an approximation of the multi-channel input signal (201). In addition, the method (700) comprises encoding (704) the plurality of compacted channel signals (404) and the joint coding metadata (205).

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 including steerable beams, holographic projectors and spatial light modulators are described.

Abstract: The described embodiments include systems and methods for producing and adapting images, such as video images, for presentation on display devices that have various different aspects ratios, such as 4:3, 16:9, 9:16, etc. In one embodiment, a method for producing content, such as video images, can begin by selecting an original aspect ratio and determining, within at least a first scene in the content, a position of a subject in the first scene. In one embodiment, the original aspect ratio can be substantially square (e.g., 1:1). Metadata can then be created, based on the position of the subject in the first scene, to guide playback devices to asymmetrically crop the content, relative to the position, for display on display devices that have aspect ratios that are different than the original aspect ratio. Other methods and systems are also described.

Abstract: Saliency regions are identified in a global scene depicted by volumetric video. Saliency video streams that track the saliency regions are generated. Each saliency video stream tracks a respective saliency region. A saliency stream based representation of the volumetric video is generated to include the saliency video streams. The saliency stream based representation of the volumetric video is transmitted to a video streaming client.

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: A method of audio processing includes generating harmonics in a hybrid complex quadrature mirror filter domain. Generating the harmonics may include multiplication, using a feedback delay loop, and dynamic compression. The harmonics may be generated based on one or more hybrid sub-bands of the complex transform domain signal.

Abstract: Systems, methods, and computer program products are disclosed for adaptive downmixing of audio signals with improved continuity. An audio encoding system receives an input multi-channel audio signal including a primary input audio channel and L non-primary input audio channels. The system determines a set of L input gains. For each of the channels and gains, the system forms a respective scaled non-primary input audio channel. The system forms a primary output audio channel from the sum of the primary input audio channel and the scaled non-primary input audio channels. The system determines a set of L prediction gains. The system forms a prediction channel from the primary output audio channel. The system forms L non-primary output audio channels. The system forms an output multi-channel audio signal from the primary output audio channel and the L non-primary output audio channels.

Abstract: The present document describes a method (600) for encoding a multi-channel input signal (101) which comprises N different channels. The method (600) comprises, for a current frame of a sequence of frames, determining (601) whether the current frame is an active frame or an inactive frame using a signal and/or a voice activity detector, and determining (602) a downmix signal (103) based on the multi-channel input signal (101), wherein the downmix signal (103) comprises N channels or less. In addition, the method (600) comprises determining (603) upmixing metadata (105) comprising a set of parameters for generating, based on the downmix signal (103), a reconstructed multi-channel signal (111) comprising N channels, wherein the upmixing metadata (105) is determined in dependence of whether the current frame is an active frame or an inactive frame. The method (600) further comprises encoding (604) the upmixing metadata (105) into a bitstream.

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.

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: Computer-implemented methods and devices for combined audio separation and classification are provided. An estimated separated signal is time gated based on a determination of an audio classifier of, at least in part, the original mix of signals before separation. Combined separation, classification, and time gating of both the estimated signal and a residual signal are also provided.

Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

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

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

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

Abstract: An audio session management method may involve: determining, by an audio session manager, one or more first media engine capabilities of a first media engine of a first smart audio device, the first media engine being configured for managing one or more audio media streams received by the first smart audio device and for performing first smart audio device signal processing for the one or more audio media streams according to a first media engine sample clock; receiving, by the audio session manager and via a first application communication link, first application control signals from the first application; and controlling the first smart audio device according to the first media engine capabilities, by the audio session manager, via first audio session management control signals transmitted to the first smart audio device via a first smart audio device communication link and without reference to the first media engine sample clock.

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

Abstract: A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

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

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

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

Abstract: A transient multi-tone test signal is generated to detect piano noise distortion played back by a speaker. The test signal has a number, such as three, formant tones resembling a piano chord and selectable such that the frequencies and harmonics remain distributed in frequency. The test signal provides a basis for defining normal speaker operating performance and allows a user to define speaker responses that fall outside of normal boundaries by highlighting piano noise distortion detected during speaker playback of the test signal.

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

Abstract: Methods, systems, and bitstream syntax are described for constrained processing in video coding. Using one or more syntax elements and either explicit or implicit signaling, an encoder may signal to compliant decoders that certain features of the main profile, such as subpictures and multi-layered scalable coding, are disabled.

Abstract: Methods and systems for chromatic ambient-light support are provided. Given an input surround correlated color temperature (CCT), its normalized value is mapped to a preferred gray CCT value using a sigmoid-like function model generated based on viewer experimental data. The preferred gray CCT value is then mapped to an adjusted CCT value so that viewing conditions on a display with the input surround CCT value match a D65 surround CCT value.

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

Abstract: In some embodiments, virtualization methods for generating a binaural signal in response to channels of a multi-channel audio signal, which apply a binaural room impulse response (BRIR) to each channel including by using at least one feedback delay network (FDN) to apply a common late reverberation to a downmix of the channels. In some embodiments, input signal channels are processed in a first processing path to apply to each channel a direct response and early reflection portion of a single-channel BRIR for the channel, and the downmix of the channels is processed in a second processing path including at least one FDN which applies the common late reverberation. Typically, the common late reverberation emulates collective macro attributes of late reverberation portions of at least some of the single-channel BRIRs. Other aspects are headphone virtualizers configured to perform any embodiment of the method.

Abstract: The present document describes a dynamic range control unit (210) configured to apply dynamic range control, referred to as DRC, to an audio signal (211). The DRC unit (210) is configured to downsample a subband signal (212) derived from the audio signal (211), to provide a downsampled subband signal (321), to determine a DRC gain (329) based on the downsampled subband signal (321), and to apply the DRC gain (329) to the subband signal (212), to provide a compressed subband signal (213) of a compressed audio signal (214).

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: 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: 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: Described herein is a method of processing audio content for rendering in a three-dimensional audio scene, wherein the audio content comprises a sound source at a source position, the method comprising: obtaining a voxelized representation of the three-dimensional audio scene, wherein the voxelized representation indicates volume elements in which sound can propagate and volume elements by which sound is occluded; generating a two-dimensional projection map for the audio scene based on the voxelized representation by applying a projection operation to the voxelized representation that projects onto a horizontal plane; and determining parameters indicating a virtual source position of a virtual sound source based on the source position, a listener position, and the projection map, to simulate, by rendering a virtual source signal from the virtual source position, an impact of acoustic diffraction by the three-dimensional audio scene on a source signal of the sound source at the source position.

Abstract: Described herein are methods, apparatus and computer products for decoding an encoded MPEG-D USAC bitstream. Described herein are such methods, apparatus and computer products that reduce a computational complexity.

Abstract: Dual and multi-modulator projector display systems and techniques are disclosed. In one embodiment, a projector display system comprises a light source; a controller, a first modulator, receiving light from the light source and rendering a halftone image of said the input image; a blurring optical system that blurs said halftone image with a Point Spread Function (PSF); and a second modulator receiving the blurred halftone image and rendering a pulse width modulated image which may be projected to form the desired screen image. Systems and techniques for forming a binary halftone image from input image, correcting for misalignment between the first and second modulators and calibrating the projector system—e.g. over time—for continuous image improvement are also disclosed.

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: 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: 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: 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: 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 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: 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.