Abstract: Methods, systems, and bitstream syntax are described for metadata signaling and conversion for film grain encoding and synthesis. Given a bitstream with MPEG film-grain SEI messaging, for each picture, a processor: detects if the film grain model is suitable for film-grain synthesis using the AV1 autoregressive with additive blending noise model, and then: transcodes the MPEG film grain SEI parameters to corresponding AV1 film grain parameters, synthesizes the film grain, and adds it to the decoded video pictures according to the AV1 specification. An example process for translating AV1 parameters to MPEG film-grain SEI messaging is also provided.

Abstract: Described are techniques in video coding and/or decoding that allow for selectively breaking prediction and/or in loop filtering across segment boundaries between different segments of a video picture. A high layer syntax element, such as a parameter set or a slice header, may contain one or more indications signalling to an encoder and/or decoder whether an associated prediction or loop filtering tool may be applied across the segment boundary. In response to such one or more indications, the encoder and/or decoder may then control the prediction or loop filtering tool accordingly.

Abstract: An audio object including audio content and object metadata is received. The object metadata indicates an object spatial position of the audio object to be rendered by audio speakers in a playback environment. Based on the object spatial position and source spatial positions of the audio speakers, initial gain values for the audio speakers are determined. The initial gain values can be used to select a set of audio speakers from among the audio speakers. Based on the object spatial position and a set of source spatial positions at which the set of audio speakers are respectively located in the playback environment, a set of non-negative optimized gain values for the set of audio speakers is determined. The audio object at the object spatial position is rendered with the set of optimized gain values for the set of audio speakers.

Abstract: The present document relates to a method of layered encoding of a frame of a compressed higher-order Ambisonics, HOA, representation of a sound or sound field. The compressed HOA representation comprises a plurality of transport signals. The method comprises assigning the plurality of transport signals to a plurality of hierarchical layers, the plurality of layers including a base layer and one or more hierarchical enhancement layers, generating, for each layer, a respective HOA extension payload including side information for parametrically enhancing a reconstructed HOA representation obtainable from the transport signals assigned to the respective layer and any layers lower than the respective layer, assigning the generated HOA extension payloads to their respective layers, and signaling the generated HOA extension payloads in an output bitstream.

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: 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: A method for enhancing audio signals is provided. In some implementations, the method involves (a) obtaining a training set comprising a plurality of training samples, each training sample comprising a distorted audio signal and a clean audio signal. In some implementations, the method involves (b), for a training sample of the plurality of training samples: obtaining a frequency-domain representation of the distorted audio signal; providing the frequency-domain representation to a convolutional neural network (CNN) comprising a plurality of convolutional layers and to a recurrent element, wherein an output of the recurrent element is provided to a subset of the plurality of convolutional layers; generating a predicted enhancement mask, wherein the CNN generates the predicted enhancement mask; generating a predicted enhanced audio signal based on the predicted enhancement mask; and updating weights associated with the CNN and the recurrent element based on the predicted enhanced audio signal.

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

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: A method for reverberation suppression may involve receiving an input audio signal. The method may involve classifying a media type of the input audio signal as one of a group comprising at least: 1) speech; 2) music; or 3) speech over music. The method may involve determining whether to perform dereverberation on the input audio signal based at least on a determination that the media type of the input audio signal has been classified as speech. The method may involve generating an output audio signal by performing dereverberation on the input audio signal in response to determining that dereverberation is to be performed on the input audio signal.

Abstract: A projection system and method therefor comprises a first light source configured to emit a first-eye light, wherein the first-eye light includes a first set of wavelengths; a second light source configured to emit a second-eye light, wherein the second-eye light includes a second set of wavelengths; a first projector including first projection optics configured to receive a first input light; and an optical switch configured to be switched between an a first mode and a second mode, wherein the optical switch is configured to, in the first mode, combine the first-eye light and the second-eye light into a combined light and direct the combined light to the first projection optics as the first input light.

Abstract: A method for reverberation suppression may involve receiving an input audio signal. The method may involve calculating an initial reverberation suppression gain for the input audio signal for at least one frame of the input audio signal. The method may involve calculating at least one adjusted reverberation suppression gain, where the at least one adjusted reverberation suppression gain adjusts at least one of: 1) a reverberation suppression decay based on a reverberation intensity detected in the input audio signal; 2) gains applied to different frequency bands of the input audio signal based on an amount of room resonance detected in the input audio signal; or 3) a loudness of the input audio signal based on a direct part of the input audio signal. The method may involve generating an output audio signal by applying the at least one adjusted reverberation suppression gain to the input audio signal.

Abstract: A video delivery system for luminance adjustment based upon a viewer adaptation state comprises a processor configured to: receive a source image including a current image frame including metadata corresponding to a mean luminance value of the current image frame, and the source image including an upcoming image frame including metadata corresponding to a mean luminance value of the upcoming image frame. The processor is configured to determine an ambient luminance value based on an ambient luminance, determine an incident luminance value based on the ambient luminance value and the mean luminance value, determine a difference between a current pupil size and a target pupil size, and generate an output image by modifying the source image based on a luminance adjustment factor, the luminance adjustment factor being a function of the difference between the current pupil size and the target pupil size.

Abstract: A projection system and method therefor includes a light source configured to emit a light; a spatial light modulator configured to receive the light and generate a modulated light; a lens configured to spatially Fourier transform the modulated light; a filter including an aperture, the filter configured to transmit at least one diffraction order of the modulated light as Fourier-transformed by the lens and to block a remaining portion of the modulated light; and a controller configured to: for each of a plurality of subperiods, cause the projection system to project an image through the filter, and between each of the plurality of subperiods, shift the image by a partial pixel distance.

Abstract: A projection system and method includes a light source configured to emit a light in response to an image data; a phase light modulator configured to receive the light from the light source and to apply a spatially-varying phase modulation on the light, thereby to steer the light and generate a projection light; and a controller configured to dynamically determine, based on at least one of a user input or a sensor signal, a target geometry of a projection surface on which the projection light is projected, determine, based on the target geometry, a phase configuration for a frame of the image data, and provide a phase control signal to the phase light modulator, the phase control signal configured to cause the phase light modulator to generate the projection light in accordance with the phase configuration for the frame.

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: Embodiments are disclosed for context aware audio processing. In an embodiment, an audio processing method comprises: receiving, with one or more sensors of a device, environment information about an audio recording captured by the device; detecting, with at least one processor of the device, a context of the audio recording based on the audio recording and the environment information; determining, with the at least one processor, a model based on the context; processing, with the at least one processor, the audio recording based on the model to produce a processed audio recording with suppressed noise; determining, with the at least one processor, an audio processing profile based on the context; and combining, with the at least one processor, the audio recording and the processed audio recording based on the audio processing profile.

Abstract: A reference picture information decoding unit (13) omits decoding of a reference list sorting presence or absence flag and/or a reference list sorting order based on the number of current picture referable pictures.

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: An input 3D point cloud including a spatial distribution of points is received. Patches including pre-reshaped patch data are generated from the input 3D point cloud. Encoder-side reshaping is performed on the pre-reshaped patch data to generate reshaped patch data for the patches. The reshaped patch data is encoded into a 3D video signal, which a recipient device of the 3D video signal can decode to generate a reconstructed 3D point cloud that approximates the input 3D point cloud.

Abstract: A method for encoding envelope information is provided. In some implementations, the method involves determining a first downmixed signal associated with a downmixed channel associated with an audio signal to be encoded. In some implementations, the method involves determining energy levels of the first downmixed signal for a plurality of frequency bands. In some implementations, the method involves determining whether to encode information indicative of the energy levels in a bitstream. In some implementations, the method involves encoding the determined energy levels. In some implementations, the method involves generating an energy control value indicating that energy levels are encoded. In some implementations, the method involves generating the bitstream, wherein the energy control value and the information indicative of the energy levels are usable by the decoder to adjust energy levels associated with the first downmixed signal.

Abstract: Described is a method of processing an audio signal. The method includes a first step for applying enhancement to a first component of the audio signal and/or applying suppression to a second component of the audio signal relative to the first component, and a second step of modifying an output of the first step by applying a deep learning based model to the output of the first step, for perceptually improving the first component of the audio signal. Also described is an apparatus for carrying out the method, as well as corresponding programs and computer-readable storage media.

Abstract: The present disclosure relates to a method and audio processing system for performing dynamic range adjustment of spatial audio objects. The method comprises obtaining (step S1) a plurality of spatial audio objects (10), obtaining (step S2) at least one rendered audio presentation of the spatial audio objects (10) and determining (step S3) signal level data associated with each presentation audio channel in said set of presentation audio channels. The method further comprises obtaining (step S31) a threshold value and, for each time segment, selecting (step S4) a selected presentation audio channel which is associated with a highest or a lowest signal level, determining (step S5) a gain based on the threshold value and the representation of the signal level of the selected audio channel, and applying (step S6) the gain of each time segment to corresponding time segments of the spatial audio objects.

Abstract: A projection system includes a light source configured to emit a light in response to an image data, a phase light modulator configured to receive the light from the light source and to apply a spatially-varying phase modulation on the light, thereby generating a projection light and steering the light on a reconstruction field, wherein the reconstruction field is a complex plane on which a reconstruction image is formed, and a controller configured to control the light source, control the phase light modulator, initialize (401) the reconstruction field to an initial value, and iteratively for each of a plurality of subframes within a frame of the image data: set (402) the reconstruction field to the initial value for the first iteration or set (402) the reconstruction field to a subsequent-iteration reconstruction field value for any subsequent-iteration, map (403) the reconstruction field to a modulation field, wherein the modulation field is a complex plane of the phase light modulator which modulates a ph

Abstract: Methods, systems, and bitstream syntax are described for the entropy modeling of latent features in image and video coding using a combination of probability density functions. Using high-level syntax elements, an encoder may signal to compliant decoders the multi-distribution entropy model using: the number of one or more PDFs being used, an identifier of each PDF being used among a list of available PDFs, the number of PDF parameters in each PDF, and syntax elements indicating which PDF parameters across two or more PDFs being used are being shared.

Abstract: Described is a method of audio processing in a HbbTV terminal device. The method includes receiving a decoded broadcast feed including a first audio track, receiving HbbTV content relating to the broadcast feed, the HbbTV content including a second audio track, extracting level-related information from the decoded broadcast feed, wherein the level-related information is embedded in the decoded broadcast feed and enables to obtain an indication of an original audio level of the first audio track, analyzing the first audio track for determining an actual audio level of the first audio track, determining a gain factor based on the actual audio level and the original audio level, and generating a third audio track for output by the HbbTV terminal device based on the first audio track, the second audio track, and the gain factor. Also described is an apparatus for carrying out the method, as well as corresponding programs and computer-readable storage media.

Abstract: Embodiments are disclosed for speaker diarization supporting episodical content. In an embodiment, a method comprises: receiving media data including one or more utterances; dividing the media data into a plurality of blocks; identifying segments of each block of the plurality of blocks associated with a single speaker; extracting embeddings for the identified segments in accordance with a machine learning model, wherein extracting embeddings for identified segments further comprises statistically combining extracted embeddings for identified segments that correspond to a respective continuous utterance associated with a single speaker; clustering the embeddings for the identified segments into clusters; and assigning a speaker label to each of the embeddings for the identified segments in accordance with a result of the clustering. In some embodiments, a voiceprint is used to identify a speaker and the speaker identity for a speaker label.

Abstract: A computing device system including a computing device having a housing and electronic components disposed within the housing, where the electronic components include a controller, a memory, and a power source. A display screen is supported on the housing, and a socket extends into the housing. A removable speaker is selectively received within the socket, where the removable speaker includes a power source that is automatically charged when received within the socket.

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: 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 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 are disclosed for adaptive display management using one or more viewing environment parameters. Given the one or more viewing environment parameters, an effective luminance range for a target display, and an input image, a tone-mapped image is generated based on a tone-mapping curve, an original PQ luminance mapping function, and the effective luminance range of the display. Corrected PQ (PQ?) luminance mapping functions are generated according to the viewing environment parameters and, optionally, the transmissivity properties and reflectivity properties of the target display.

Abstract: A method for performing gain control on audio signals is provided. In some implementations, the method involves determining downmixed signals associated with one or more downmix channels associated with a current frame of an audio signal to be encoded. In some implementations, the method involves determining whether an overload condition exists for an encoder. In some implementation, the method involves determining a gain parameter. In some implementations, the method involves determining at least one gain transition function based on the gain parameter and a gain parameter associated with a preceding frame of the audio signal. In some implementations, the method involves applying the at least one gain transition function to one or more of the downmixed signals. In some implementations, the method involves encoding the downmixed signals in connection with information indicative of gain control applied to the current frame.

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: Embodiments are disclosed for context aware soundscape control. In an embodiment, an audio processing method comprises: capturing, using a first set of microphones on a mobile device, a first audio signal from an audio scene; capturing, using a second set of microphones on a pair of earbuds, a second audio signal from the audio scene; capturing, using a camera on the mobile device, a video signal from a video scene; generating, with at least one processor, a processed audio signal from the first audio signal and the second audio signal, the processed audio signal generated with adaptive soundscape control based on context information; and combining, with the at least one processor, the processed audio signal and the captured video signal as multimedia output.

Abstract: Several embodiments of scalable image processing systems and methods are disclosed herein whereby color management processing of source image data to be displayed on a target display is changed according to varying levels of metadata.

Abstract: Disclosed is a portable computing device (1) comprising a keyboard (13) and an acoustic transducer (16, 17), the keyboard (13) comprising a key (14, 15), wherein the acoustic transducer (16, 17) is placed in the key (14, 15), and wherein the key (14, 15) comprises a sound port (150) allowing sound generated by the transducer (16, 17) to propagate.

Abstract: An audio processing unit (APU) is disclosed. The APU includes a buffer memory configured to store at least one frame of an encoded audio bitstream, where the encoded audio bitstream includes audio data and a metadata container. The metadata container includes a header and one or more metadata payloads after the header. The one or more metadata payloads include dynamic range compression (DRC) metadata, and the DRC metadata is or includes profile metadata indicative of whether the DRC metadata includes dynamic range compression (DRC) control values for use in performing dynamic range compression in accordance with at least one compression profile on audio content indicated by at least one block of the audio data.

Abstract: A method (700) for rendering an audio signal of an audio source (211, 212, 213) in a virtual reality rendering environment (180) is described. The method (700) comprises determining (701) whether or not a directivity pattern (232) of the audio source (211, 212, 213) is to be taken into account for a listening situation of a listener (181) within the virtual reality rendering environment (180). Furthermore, the method (700) comprises rendering (702) an audio signal of the audio source (211, 212, 213) without taking into account the directivity pattern (232) of the audio source (211, 212, 213), if it is determined that the directivity pattern (232) of the audio source (211, 212, 213) is not to be taken into account for the listening situation of the listener (181).

Abstract: A method for delivering media to a playback device including outputting first test media to be viewed by a first user. The method further includes receiving a first user input related to a first perception of the first test media by the first user and indicating a first personalized quality of experience of the first user with respect to the first test media. The method further includes generating a first personalized sensitivity profile including one or more viewing characteristics of the first user based on the first user input, and determining, based at least in part on the first personalized sensitivity profile, a first media parameter. The first media parameter is determined in order to increase an efficiency of media delivery to the first playback device over a network while preserving the first personalized quality of experience of the first user.

Abstract: A method for decoding an encoded audio bitstream in an audio processing system is disclosed. The method includes extracting from the encoded audio bitstream a first waveform-coded signal comprising spectral coefficients corresponding to frequencies up to a first cross-over frequency for a time frame and performing parametric decoding at a second cross-over frequency for the time frame to generate a reconstructed signal. The second cross-over frequency is above the first cross-over frequency and the parametric decoding uses reconstruction parameters derived from the encoded audio bitstream to generate the reconstructed signal. The method also includes extracting from the encoded audio bitstream a second waveform-coded signal comprising spectral coefficients corresponding to a subset of frequencies above the first cross-over frequency for the time frame and interleaving the second waveform-coded signal with the reconstructed signal to produce an interleaved signal for the time frame.

Abstract: The present disclosure provides methods, devices and computer program products for non-uniform quantization of parameters. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system taking the non-uniformly quantized parameters into account. According to the disclosure, such an approach renders it possible to reduce bit consumption without substantially reducing the quality of the reconstructed audio object.

Abstract: A volumetric image of a scene can be created, in one embodiment, by recording, through a camera in a device, a series of images of the scene as the camera is moved along a path relative to the scene; during the recording, the device stores motion path metadata about the path, and the series of images is associated with the motion path metadata and a metadata label is associated with the series of images, the metadata label indicating that the recorded series of images represent a volumetric image of the scene. The series of images, the motion path metadata and the metadata label can be assembled into a package for distribution to devices that can view the volumetric image, which may be referred to as a limited volumetric image. The devices that receive the volumetric image can display the individual images in the series of images or as a video.

Abstract: Several embodiments of scalable image processing systems and methods are disclosed herein whereby color management processing of source image data to be displayed on a target display is changed according to varying levels of metadata.

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: A method for performing calibration of magnetometers is provided. In some embodiments, the method involves obtaining a sequence of gyroscope measurements from one or more gyroscopes and a sequence of magnetometer measurements from one or more magnetometers. In some embodiments, the method involves determining a sequence of angular velocity estimates based on the sequence of gyroscope measurements. In some embodiments, the method involves determining a first estimate of a derivative of an external magnetic field based on the sequence of magnetometer measurements. In some embodiments, the method involves determining a second estimate of the derivative of the external magnetic field based on the sequence of angular velocity estimates. In some embodiments, the method involves identifying magnetometer calibration constants based on a difference between the first estimate of the derivative and the second estimate of the derivative.

Abstract: A handheld imaging device has a data receiver that is configured to receive reference encoded image data. The data includes reference code values, which are encoded by an external coding system. The reference code values represent reference gray levels, which are being selected using a reference grayscale display function that is based on perceptual non-linearity of human vision adapted at different light levels to spatial frequencies. The imaging device also has a data converter that is configured to access a code mapping between the reference code values and device-specific code values of the imaging device. The device-specific code values are configured to produce gray levels that are specific to the imaging device. Based on the code mapping, the data converter is configured to transcode the reference encoded image data into device-specific image data, which is encoded with the device-specific code values.

Abstract: A method and apparatus for decompressing a Higher Order Ambisonics (HOA) signal representation is disclosed. The apparatus includes an input interface that receives an encoded directional signal and an encoded ambient signal and an audio decoder that perceptually decodes the encoded directional signal and encoded ambient signal to produce a decoded directional signal and a decoded ambient signal, respectively. The apparatus further includes an extractor for obtaining side information related to the directional signal and an inverse transformer for converting the decoded ambient signal from a spatial domain to an HOA domain representation of the ambient signal. The apparatus also includes a synthesizer for recomposing a Higher Order Ambisonics (HOA) signal from the HOA domain representation of the ambient signal and the decoded directional signal. The side information includes a direction of the directional signal selected from a set of uniformly spaced directions.

Abstract: A projection system for etendue utilization includes a first light source configured to emit a light, the light including a first etendue component and a second etendue component, wherein the first etendue component has a lower etendue than the second etendue component, a first projection optics configured to project a first image on a screen, a second projection optics configured to project a second image on the screen, and an etendue splitter component. The etendue splitter component is configured to receive the light from the light source, extract, from the light, the first etendue component and the second etendue component, provide the first etendue component to the first projection optics, and provide the second etendue component to the second projection optics.

Abstract: The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel on the basis of the first and second frequency-domain representations of the first input channel, the first frequency-domain representation of the second input channel and a complex prediction coefficient. The upmixing can be suspended responsive to control data.