Abstract: An apparatus and method of rendering audio. A binaural signal is split on an amplitude weighting basis into a front binaural signal and a rear binaural signal, based on perceived position information of the audio. In this manner, the front-back differentiation of the binaural signal is improved.

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: Dual or multi-modulation display systems comprising a first modulator and a second modulator are disclosed. The first modulator may comprise a plurality of analog mirrors (e.g. MEMS array) and the second modulator may comprise a plurality of mirrors (e.g., DMD array). The display system may further comprise a controller that sends control signals to the first and second modulator. The display system may render highlight features within a projected image by affecting a time multiplexing scheme. In one embodiment, the first modulator may be switched on a sub-frame basis such that a desired proportion of the available light may be focused or directed onto the second modulator to form the highlight feature on a sub-frame rendering basis.

Abstract: A video encoding method according to an embodiment of the present invention includes generating header information that includes information about resolutions of motion vectors of respective blocks, determined based on motion prediction for a unit image. Here, the header information includes flag information indicating whether resolutions of all motion vectors included in the unit image are integer-pixel resolutions. Further, a video decoding method according to another embodiment of the present invention includes extracting information about resolutions of motion vectors of each unit image from header information included in a target bitstream to be decoded; and a decoding unit for decoding the unit image based on the resolution information. Here, the header information includes flag information indicating whether resolutions of all motion vectors included in the unit image are integer-pixel resolutions.

Abstract: Disclosed is a method for determining one or more dialog quality metrics of a mixed audio signal comprising a dialog component and a noise component, the method comprising separating an estimated dialog component from the mixed audio signal by means of a dialog separator using a dialog separating model determined by training the dialog separator based on the one or more quality metrics; providing the estimated dialog component from the dialog separator to a quality metrics estimator; and determining the one or more quality metrics by means of the quality metrics estimator based on the mixed signal and the estimated dialog component. Further disclosed is a method for training a dialog separator, a system comprising circuitry configured to perform the method, and a non-transitory computer-readable storage medium.

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: 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: 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: Encoding and decoding devices for encoding the channels of an audio system having at least four channels are disclosed. The decoding device has a first stereo decoding component which subjects a first pair of input channels to a first stereo decoding, and a second stereo decoding component which subjects a second pair of input channels to a second stereo decoding. The results of the first and second stereo decoding components are crosswise coupled to a third and a fourth stereo decoding component which each performs stereo decoding on one channel resulting from the first stereo decoding component, and one channel resulting from the second stereo decoding component.

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 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 of audio processing includes performing spatial analysis on a binaural signal to estimate level differences and phase differences characteristic of a binaural filter of the binaural signal, performing object extraction on the binaural audio signal using the estimated level and phase differences to generate a left/right main component signal and a left/right residual component signal. The system may process the left/right main and left/right residual components differently using different object processing parameters for e.g. repositioning, equalization, compression, upmixing, channel remapping or storage to generate a processed binaural signal that provides an improved listening experience. Repositioning may be based on head tracking sensor data.

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

Abstract: 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: Some methods may involve receiving a first content stream that includes first audio signals, rendering the first audio signals to produce first audio playback signals, generating first direct sequence spread spectrum (DSSS) signals, generating first modified audio playback signals by inserting the first DSSS signals into the first audio playback signals, and causing a loudspeaker system to play back the first modified audio playback signals, to generate first audio device playback sound. The method(s) may involve receiving microphone signals corresponding to at least the first audio device playback sound and to second through Nth audio device playback sound corresponding to second through Nth modified audio playback signals (including second through Nth DSSS signals) played back by second through Nth audio devices, extracting second through Nth DSSS signals from the microphone signals and estimating at least one acoustic scene metric based, at least partly, on the second through Nth DSSS signals.

Abstract: Described herein is a method for setting up a decoder for generating processed audio data from an audio bitstream, the decoder comprising a Generator of a Generative Adversarial Network, GAN, for processing of the audio data, wherein the method includes the steps of (a) pre-configuring the Generator for processing of audio data with a set of parameters for the Generator, the parameters being determined by training, at training time, the Generator using the full concatenated distribution; and (b) pre-configuring the decoder to determine, at decoding time, a truncation mode for modifying the concatenated distribution and to apply the determined truncation mode to the concatenated distribution. Described are further a method of generating processed audio data from an audio bitstream using a Generator of a Generative Adversarial Network, GAN, for processing of the audio data and a respective apparatus. Moreover, described are also respective systems and computer program products.

Abstract: An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.

Abstract: A method for delivering media content to one or more clients over a distributed system is disclosed. The method may include generating a plurality of network-coded symbols from a plurality of original symbols representing a first media asset. The method may further include generating an original plurality of coded variants of the first media asset. The method may further include distributing a first coded variant of the original plurality of coded variants to a first cache on a first server device for storage in the first cache. The method may further include distributing a second coded variant of the original plurality of coded variants to a second cache on a second server device for storage in the second cache.

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

Abstract: The present disclosure relates to a method of processing audio content including directivity information for at least one sound source, the directivity information comprising a first set of first directivity unit vectors representing directivity directions and associated first directivity gains. The disclosure further relates to corresponding methods of encoding and decoding audio content including directivity information for at least one sound source.

Abstract: An apparatus and method of generating personalized HRTFs. The system is prepared by calculating a model for HRTFs described as the relationship between a finite example set of input data, namely anthropometric measures and demographic information for a set of individuals, and a corresponding set of output data, namely HRTFs numerically simulated using a high-resolution database of 3D scans of the same set of individuals. At the time of use, the system queries the user for their demographic information, and then from a series of images of the user, the system detects and measures various anthropometric characteristics. The system then applies the prepared model to the anthropometric and demographic data as part of generating a personalized HRTF. In this manner, the personalized HRTF can be generated with more convenience than by performing a high-resolution scan or an acoustic measurement of the user, and with less computational complexity than by numerically simulating their HRTF.

Abstract: Some methods may involve receiving a first content stream that includes first audio signals, rendering the first audio signals to produce first audio playback signals, generating first direct sequence spread spectrum (DSSS) signals, generating first modified audio playback signals by inserting the first DSSS signals into the first audio playback signals, and causing a loudspeaker system to play back the first modified audio playback signals, to generate first audio device playback sound. The method(s) may involve receiving microphone signals corresponding to at least the first audio device playback sound and to second through Nth audio device playback sound corresponding to second through Nth modified audio playback signals (including second through Nth DSSS signals) played back by second through Nth audio devices, extracting second through Nth DSSS signals from the microphone signals and estimating at least one acoustic scene metric based, at least partly, on the second through Nth DSSS signals.

Abstract: The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency Q?+r?0 is generated on the basis of existing components at ? and ?+?0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore, compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

Abstract: The present disclosure relates to methods and apparatus for processing media content having video content and associated audio content. A method of processing media content having video content and associated audio content comprises the method includes receiving the video content and the associated audio content, analyzing the associated audio content, determining one or more navigation points for enabling navigation of the media content based on the analysis, wherein the one or more navigation points indicate points of interest in the associated audio content for short-term rewinding and/or fast forwarding, embedding the one or more navigation points into metadata for the media content, and outputting the video content, the associated audio content, and the metadata.

Abstract: A method for steering binauralization of audio is provided. The method comprises steps of: receiving (410) an audio input signal, calculating (430) a confidence value indicating a likelihood that a current audio frame of the audio input signal comprises binauralized audio; determining (450) a state signal based on the confidence value; determining (460) a steering signal, based on the first confidence value, the state signal and an energy value of the audio frame; and generating (470) an audio output signal with steered binauralization by processing the audio input signal according to the steering signal.

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

Abstract: A device includes an electronic processor configured to define a first set of sample pixels from a set of sample pixels determined from received video data according to a first electro-optical transfer function (EOTF) in a first color representation of a first color space; convert the first set of sample pixels to a second EOTF via a mapping function, producing a second set of sample pixels according to the second EOTF; convert the first and second set of sample pixels from the first color representation to a second color representation of the first color space; determine a backward reshaping function by repeatedly applying and adjusting a sample backward reshaping function so as to minimize a difference between predicted pixel values obtained by applying the sample backward reshaping function to the pixels of the converted first set and the pixels of the converted second set.

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

Abstract: 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: An apparatus for processing an input audio signal relies on a cascade of filterbanks, the cascade having a synthesis filterbank for synthesizing an audio intermediate signal from the input audio signal, the input audio signal being represented by a plurality of first subband signals generated by an analysis filterbank, wherein a number of filterbank channels of the synthesis filterbank is smaller than a number of channels of the analysis filterbank. The apparatus furthermore has a further analysis filterbank for generating a plurality of second subband signals from the audio intermediate signal, wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank, so that a sampling rate of a subband signal of the plurality of second subband signals is different from a sampling rate of a first subband signal of the plurality of first subband signals.

Abstract: The processing of audio signals during playback is provided, so that audio signals that fall below a specified threshold loudness level are processed to avoid making unwanted background noise audible. N-channel audio is received from a playback volume controller/leveler (101). The level of the audio is compared with a threshold level. If the level is greater than the threshold level, the audio is processed with a first amount of gain in accordance with a first dynamic range control (DRC) compression curve that is tuned for professionally produced audio. If the level is less than or equal to the threshold level, the audio is processed with a second amount of gain in accordance with a second DRC compression curve that is designed to avoid boosting unwanted background noise. After applying the gain to the audio, the audio is sent to a downstream device.

Abstract: An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighbourhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream. At small computational expense, the endpoint may to adapt to different sample rates in received bitstreams.

Abstract: Volume leveler controller and controlling method are disclosed. In one embodiment, A volume leveler controller includes an audio content classifier for identifying the content type of an audio signal in real time; and an adjusting unit for adjusting a volume leveler in a continuous manner based on the content type as identified. The adjusting unit may configured to positively correlate the dynamic gain of the volume leveler with informative content types of the audio signal, and negatively correlate the dynamic gain of the volume leveler with interfering content types of the audio signal.

Abstract: Encoding/decoding an audio signal having one or more audio components, wherein each audio component is associated with a spatial location. A first audio signal presentation (z) of the audio components, a first set of transform parameters (w(f)), and signal level data (?2) are encoded and transmitted to the decoder. The decoder uses the first set of transform parameters (w(f)) to form a reconstructed simulation input signal intended for an acoustic environment simulation, and applies a signal level modification (?) to the reconstructed simulation input signal. The signal level modification is based on the signal level data (?2) and data (p2) related to the acoustic environment simulation. The attenuated reconstructed simulation input signal is then processed in an acoustic environment simulator. With this process, the decoder does not need to determine the signal level of the simulation input signal, thereby reducing processing load.

Abstract: A method of audio content identification includes using a two-stage classifier. The first stage includes previously-existing classifiers and the second stage includes a new classifier. The outputs of the first and second stages calculated over different time periods are combined to generate a steering signal. The final classification results from a combination of the steering signal and the outputs of the first and second stages. In this manner, a new classifier may be added without disrupting existing classifiers.

Abstract: The invention discloses rendering sound field signals, such as Higher-Order Ambisonics (HOA), for arbitrary loudspeaker setups, where the rendering results in highly improved localization properties and is energy preserving. This is obtained by rendering an audio sound field representation for arbitrary spatial loudspeaker setups and/or by a a decoder that decodes based on a decode matrix (D). The decode matrix (D) is based on smoothing and scaling of a first decode matrix {circumflex over (D)} with smoothing coefficients. The first decode matrix {circumflex over (D)} is based on a mix matrix G and a mode matrix {tilde over (?)}, where the mix matrix G was determined based on L speakers and positions of a spherical modelling grid related to a HOA order N, and the mode matrix {tilde over (?)} was determined based on the spherical modelling grid and the HOA order N.

Abstract: Disclosed is a method for managing acoustic feedback in real-time audio communications in a communications system, the method comprising determining, by means of a detection module, whether a first communication device is in loudspeaker mode, whether the first communication device is in real-time audio communications with a second communication, and whether the first communication device and the second communication device are in a same acoustic space. Upon determining that this is the case a request signal for requesting one or more measures against acoustic feedback is provided to a mitigation module. Further disclosed are a device and a system configured to perform the method, a non-transitory computer-readable medium, an encoder and a decoder.

Abstract: A non-transitory computer-readable-medium storing instructions that, when executed by a processor of an image projector, cause the image projector to perform operations including receiving a first image data, determining a thermal state of the image projector based at least in part on a content of the first image data, generating a second image data based on the first image data and the thermal state; emitting light in response to the second image data, and projecting an image onto a screen based on the emitted light, wherein the first image data corresponds to a frame of a video, and the second image data corresponds to the frame of the video.

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

Abstract: Theater designs including social media, and theater entrances configured to draw patrons into a main area of the theater including an entrance having a curved wall and curved walkway having a horizon drop-off for display of content related to a motion picture or other feature to be shown in the theater. The content may be interactive or otherwise linked to patrons passing a hallway including the curved wall. Spatialized sound corresponding to the video “moves” with objects in the video and with patrons as they traverse walkway.

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: 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 is provided for coding at least one image split up into partitions, a current partition to be coded containing data, at least one data item of which is allotted a sign. The coding method includes, for the current partition, the following steps: calculating the value of a function representative of the data of the current partition with the exclusion of the sign; comparing the calculated value with a predetermined value of the sign; as a function of the result of the comparison, modifying or not modifying at least one of the data items of the current partition, in the case of modification, coding the at least one modified data item.

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: The positions of a plurality of speakers at a media consumption site are determined. Audio information in an object-based format is received. Gain adjustment value for a sound content portion in the object-based format may be determined based on the position of the sound content portion and the positions of the plurality of speakers. Audio information in a ring-based channel format is received. Gain adjustment value for each ring-based channel in a set of ring-based channels may be determined based on the ring to which the ring-based channel belongs and the positions of the speakers at a media consumption site.

Abstract: A method (900) for rendering audio in a virtual reality rendering environment (180) is described. The method (900) comprises rendering (901) an origin audio signal of an origin audio source (113) of an origin audio scene (111) from an origin source position on a sphere (114) around a listening position (201) of a listener (181). Furthermore, the method (900) comprises determining (902) that the listener (181) moves from the listening position (201) within the origin audio scene (111) to a listening position (202) within a different destination audio scene (112). In addition, the method (900) comprises applying (903) a fade-out gain to the origin audio signal to determine a modified origin audio signal, and rendering (903) the modified origin audio signal of the origin audio source (113) from the origin source position on the sphere (114) around the listening position (201, 202).

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 (600) for decoding an encoded audio signal (102) is described. The encoded audio signal (102) comprises a sequence of frames. Furthermore, the encoded audio signal (102) is indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. Different subsets of DRC profiles from the plurality of DRC profiles are comprised within different frames of the sequence of frames, such that two or more frames of the sequence of frames jointly comprise the plurality of DRC profiles.