Abstract: Methods and systems are described for video coding and decoding using cross-component prediction tools. The proposed methods include: inter-luma based modeling for chroma, cross component residue prediction, and convolutional cross-component intra prediction.

Abstract: The present disclosure relates to a method and system for performing speech classification for an audio signal. The method comprises obtaining the audio signal comprising a sequence of audio frames and determining, for each audio frame, a first speech confidence metric using a first speech classifier. For each given audio frame of at least a subset of the sequence of audio frames the method comprises classifying each respective audio frame of a first context window associated with the given audio frame as a speech-frame or non-speech frame by comparing the first speech confidence metric of the respective audio frame to a first predetermined threshold, determining an adaptive threshold based on the number of speech frames of the first context window and determining a first binary speech classification indicator for the given audio frame based on the first speech confidence metric and the adaptive threshold.

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

Abstract: Systems, devices, and methods are described for determining a “coupled” pair of Left/Right ear Head Related Transfer Functions (HRTFs) that are adapted from an original pair of Left/Right ear HRTFs, wherein the inter-aural delay of the coupled HRTFs is formed using all-pass filters that provide the correct inter-aural delay at low frequencies. The all-pass filters are adapted to limit the inter-aural phase difference at high frequencies. Furthermore, a low-complexity process is described for rapid generation of suitable all-pass filters.

Abstract: In one example, a method of streaming a multiple perspective video of a scene includes generating two or more streams of decoding units (DUs) to encode the videos of different respective views of the scene by intra-coding a selected region of a respective frame, inter-coding other regions of the respective frame, and changing the selected region in accordance with a gradual decoding refresh period. The method also includes generating a stream of packets to carry the generated streams of DUs. The stream of packets includes source packets and parity packets configured to provide unequal error protection to different sets of the DUs based on the respective importance ranks assigned thereto. In some examples, the unequal error protection is implemented using rateless forward error correction coding and a greedy algorithm for optimally distributing the budget of parity packets among the various DUs.

Abstract: An aspect of the present disclosure relates to processing audio comprising decoding a first bitstream (b1) to obtain decoded immersive audio content (A), decoding a second bitstream (bp) to obtain pose information (P, V, V?) associated with a user of a lightweight processing device, determining a first head-pose (P?) based on the pose information, providing a downmix representation (Dmx) of the immersive audio content (A) corresponding to the first head pose (P?), rendering a set of binaural representations (BINn) of the immersive audio content (A), wherein the binaural representations correspond to a second set of head poses (Pn), computing reconstruction metadata (M) to enable reconstruction of the set of binaural representations from the downmix representation (Dmx), the metadata (M) including the first head pose (P?), and encoding the downmix representation (Dmx) and the reconstruction metadata (M) in a third bitstream (b2).

Abstract: A method for generating personalized head-related transfer functions (pHRTFs) for a user of a media playback device comprising acquiring a feature set, x, including anthropometric features acquired with an image capturing system, estimating an initial parameter set, y?, including pHRTF model parameters for the user based on a statistical relationship between the pHRTF model parameters, y, and the feature set, x, estimating a set of final model parameters, y?, and generating a set of pHRTFs from the set of final model parameters. The set of final model parameters are based on the initial parameter set, y?, a demographic prior distribution describing expected variation of pHRTF model parameters, and an accuracy prior distribution describing expected errors in the initial parameter set, y. The accuracy prior distribution is derived from accuracy statistics associated with the image capture system.

Abstract: Methods and systems are described for intra-prediction using template matching (TM) in video coding. The proposed methods include adaptive fusion when using template-based intra mode derivation using the most probable modes (TIMD), and fusion in intra mode prediction with template matching (Intra TMP).

Abstract: A coding efficiency increase is achieved by using a common signalization within the bitstream with regard to activation of merging and activation of the skip mode. One possible state of one or more syntax elements within the bitstream may signalize for a current sample set of a picture that the sample set is to be merged and has no prediction residual encoded and inserted into the bitstream. A common flag may signalize whether the coding parameters associated with a current sample set are to be set according to a merge candidate or to be retrieved from the bitstream, and whether the current sample set of the picture is to be reconstructed based on a prediction signal depending on the coding parameters associated with the current sample set, without any residual data, or to be reconstructed by refining the prediction signal depending on the coding parameters associated with the current sample set by means of residual data within the bitstream.

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: Method for encoding scene-based audio is provided. In some implementations, the method involves determining, by an encoder, a spatial direction of a dominant sound component in a frame of an input audio signal. In some implementations, the method involves determining rotation parameters based on the determined spatial direction and a direction preference of a coding scheme to be used to encode the input audio signal. In some implementations, the method involves rotating sound components of the frame based on the rotation parameters such that, after being rotated, the dominant sound component has a spatial direction that aligns with the direction preference of the coding scheme. In some implementations, the method involves encoding the rotated sound components of the frame of the input audio signal using the coding scheme in connection with an indication of the rotation parameters or an indication of the spatial direction of the dominant sound component.

Abstract: Disclosed are methods and systems which convert a multi-microphone input signal to a multichannel output signal making use of a time-and frequency-varying matrix. For each time and frequency tile, the matrix is derived as a function of a dominant direction of arrival and a steering strength parameter. Likewise, the dominant direction and steering strength parameter are derived from characteristics of the multi-microphone signals, where those characteristics include values representative of the inter-channel amplitude and group-delay differences.

Abstract: A novel spatial light modulator (SLM) includes a cover glass, and modulation layer, and a plurality of pixel minors, and separates unwanted, reflected light from desired, modulated light. In one embodiment, a geometrical relationship exists between the cover glass and the pixel minors, such that light that reflects from the cover glass is separated from light that reflects from the pixel minors and is transmitted from the SLM. In one example, one of the cover glass or the pixel minors is angled with respect to the modulation layer. In another example embodiment, the cover glass has a particular thickness, which introduces destructive interference between light that reflects from the top and bottom surfaces of the cover glass. In another embodiment antireflective coatings are disposed between optical interfaces of the SLM. In another embodiment, light from the SLM is directed through an optical filter to remove unwanted light.

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

Abstract: The present disclosure relates to a method of decoding audio scene content from a bitstream by a decoder that includes an audio renderer with one or more rendering tools.

Abstract: The present invention relates to a method for predicting transform coefficients representing frequency content of an adaptive block length media signal, by receiving a frame and receiving block length information indicating a number of quantized transform coefficients for each block in the frame, the number of quantized transform coefficients being one of a first or second number, wherein the first number is greater than the second number, determining a first block has the second number of quantized transform coefficients, converting the first block into a converted block having the first number of quantized transform coefficients, conditioning a main neural network trained to predict at least one output variable given at least one conditioning variable, the at least one conditioning variable being based on information regarding the converted block and block length information for the first block, providing at least one predicted transform coefficients from an output stage of the main neural network.

Abstract: The present disclosure relates to a method and system for predicting a future orientation of an orientation tracker (100). The method comprising obtaining a sequence of angular velocity samples, each angular velocity sample indicating an angular velocity at a point in time and obtaining a sequence of angular acceleration samples, each angular acceleration sample indicating an acceleration or deceleration of the angular velocity at each point in time. Wherein said method further comprises determining (S5a), for each point in time where the angular velocity is accelerating, a predicted orientation of the orientation tracker (100) based on a first order prediction of an accumulated rotation of the orientation tracker (100) and determining (S5c), for each point in time where the angular velocity is decelerating, a predicted orientation of the orientation tracker (100) based on a second order prediction of the accumulated rotation of the orientation tracker (100).

Abstract: Methods and systems for generating trim-pass metadata for high dynamic range (HDR) video are described. The trim-pass prediction pipeline includes a feature extraction network followed by a fully connected network which maps extracted features to trim-pass values. In a first architecture, the feature extraction network is based on four cascaded convolutional networks. In a second architecture, the feature extraction network is based on a modified MobileNetV3 neural network. In both architectures, the fully connected network is formed by a set of three linear networks, each set customized to best match its corresponding feature extraction network.

Abstract: The present disclosure generally relates to user interfaces and techniques for capturing image data. In some embodiments, method comprises: displaying, on the display, a user interface including a preview portion; displaying images in the preview portion corresponding to image data captured by the camera; and while displaying the images in the preview portion, performing a capture process including: capturing a series of images corresponding to preview images in the preview portion using the camera; and determining pose data associated with the series of images; in accordance with the pose data meeting or exceeding a first threshold, causing output of a first set of instructional prompts; and in accordance with the pose data meeting or exceeding a second threshold, causing output of a second set of instructional prompts; and ceasing the capture process in response to a determination that a set of sufficiency criteria are met.

Abstract: Techniques for adaptive processing of media data based on separate data specifying a state of the media data are provided. A device in a media processing chain may determine whether a type of media processing has already been performed on an input version of media data. If so, the device may adapt its processing of the media data to disable performing the type of media processing. If not, the device performs the type of media processing. The device may create a state of the media data specifying the type of media processing. The device may communicate the state of the media data and an output version of the media data to a recipient device in the media processing chain, for the purpose of supporting the recipient device’s adaptive processing of the media data.

Abstract: A first image for rendering on a first image display in a combination of a stationary image display and a non-stationary image display is received. A visual object depicted in the first image is identified. A corresponding image portion in a second image is generated for rendering on a second image display in the combination of the stationary image display and the non-stationary image display. The corresponding image portion in the second image as rendered on the second image display overlaps in a vision field of a viewer with the visual object depicted in the second image as rendered on the first image display to modify one or more visual characteristics of the visual object. The second image is caused to be rendered on the second image concurrently while the first image is being rendered on the second image display.

Abstract: Methods, systems, and bitstream syntax are described for the fusion of latent features in multi-level, end-to-end, neural networks used in image and video compression. The fused architecture may be static or dynamic based on image characteristics (e.g., natural images versus screen content images) or other coding parameters, such as bitrate constrains or rate-distortion optimization. A variety of multi-level fusion architectures are discussed.

Abstract: Described herein is a method of metadata-based dynamic processing of audio data for playback, the method including: receiving, by a decoder, a bitstream including audio data and metadata for dynamic loudness adjustment; decoding, by the decoder, the audio data and the metadata to obtain decoded audio data and the metadata; determining, by the decoder, from the metadata, one or more processing parameters for dynamic loudness adjustment based on a playback condition; applying the determined one or more processing parameters to the decoded audio data to obtain processed audio data; and outputting the processed audio data for playback. Described is further a method of encoding audio data and metadata for dynamic loudness adjustment into a bitstream. Moreover, described are a respective decoder and encoder, a respective system and computer program products.

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: 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 method for encoding a video picture into a bitstream of encoded video picture data, includes: obtaining a grid of coding-tree units to split at least one component of the video picture into coding-tree units, each coding-tree unit being a picture area subdivided according to a coding tree; determining at least one shifting offset by aligning at least one boundary of the grid of coding-tree units with at least one boundary separating picture areas with low spatial activity from picture areas with high spatial activity of the video picture; shifting the grid of coding-tree units according to the at least one shifting offset; obtaining encoded video picture data by encoding at least one coding unit (CU) of a coding tree associated with each coding-tree unit (CTU) of the shifted grid of coding-tree units; and writing the encoded video picture data into the bitstream.

Abstract: In one example, a method of generating a volumetric video based on a monocular video includes: obtaining a respective foreground image and a respective background image based on image segmentation of a frame of the monocular video; completing the respective background image by inpainting one or more occluded areas therein based on one or more neighboring frames of the monocular video; computing a background depth map corresponding to the completed background image and a foreground depth map corresponding to the respective foreground image; generating a first multiplane image (MPI) based on the respective foreground image and the foreground depth map; generating a second MPI based on the completed background image and the background depth map; and composing the first and second MPIs into a third MPI representing a frame of the volumetric video corresponding to the frame of the monocular video.

Abstract: Embodiments are disclosed for spatial noise filling in multi-channel codecs. In an embodiment, a method of regenerating background noise ambience in a multi-channel codec by generating spatial hole filling noise comprises: computing noise estimates based on a primary downmix channel generated from an input audio signal representing a spatial audio scene with background noise ambience; computing spectral shaping filter coefficients based on the noise estimates; spectrally shaping the multi-channel noise signal using the spectral shaping filter coefficients and a noise distribution, the spectral shaping resulting in a diffused, multi-channel noise signal with uncorrelated channels; spatially shaping the diffused, uncorrelated multi-channel noise signal with uncorrelated channels based on a noise ambience of the spatial audio scene; and adding the spatially and spectrally shaped multi-channel noise to a multi-channel codec output to synthesize the background noise ambience of the spatial audio scene.

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: Methods and apparatus for neural-field-based multiple description coding in the source domain and/or coefficient domain. According to an example embodiment, a method for multiple description coding implemented at an electronic decoder comprises receiving, from an electronic encoder, a plurality of descriptions represented by a neural field network. Each of the descriptions is characterized by a respective first set of neural-field-network parameter values obtained via training the neural field network with a multimedia object. Each of the respective first sets of the neural-field-network parameter values corresponds to a different respective sampling of the multimedia object or of a larger second set of neural-field-network parameter values corresponding to the multimedia object.

Abstract: A system for managing user-generated content (UGC) and professionally generated content (PGC) is disclosed. The system is programmed to receive digital audio data having two channels from a social media platform. The system is programmed to extract spatial features that capture differences in the two channels from the digital audio data. The system is programmed to also extract temporal features, spectral features, and background features from the digital audio data. The system is programmed to then use the extracted features to determine whether to process the digital audio data as UGC or PGC before playback.

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

Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: An audio processing method may involve receiving output signals from each microphone of a plurality of microphones in an audio environment, the output signals corresponding to a current utterance of a person. The method may involve determining, responsive to the output signals and based at least in part on audio device location information and echo management system information, one or more audio processing changes to apply to audio data being rendered to loudspeaker feed signals for two or more audio devices in the audio environment. The audio processing changes may involve a reduction in a loudspeaker reproduction level for one or more loudspeakers in the audio environment. The method may involve causing one or more types of audio processing changes to be applied. The audio processing changes may have the effect of increasing a speech to echo ratio at one or more microphones.

Abstract: A system includes one or more processors and memory. The memory stores instructions for execution by the one or more processors, including instructions for: obtaining a configuration request for a communications network; configuring a network of models (e.g., oscillators or oscillators' settings) into an initial configuration representing the configuration request for the communications network; reading out a final configuration of the network of models, the final configuration representing a solution to the configuration request for the communications network; and providing information over the communications network according to the configuration request.

Abstract: Described is a method of processing a media stream.

Abstract: Decoding of Ambisonics representations for a stereo loudspeaker setup is known for first-order Ambisonics audio signals. But such first-order Ambisonics approaches have either high negative side lobes or poor localisation in the frontal region. The invention deals with the processing for stereo decoders for higher-order Ambisonics HOA. The desired panning functions can be derived from a panning law for placement of virtual sources between the loudspeakers. For each loudspeaker a desired panning function for all possible input directions at sampling points is defined. The panning functions are approximated by circular harmonic functions, and with increasing Ambisonics order the desired panning functions are matched with decreasing error. For the frontal region between the loudspeakers, a panning law like the tangent law or vector base amplitude panning (VBAP) are used. For the rear directions panning functions with a slight attenuation of sounds from these directions are defined.

Abstract: Some disclosed methods involve causing one or more loudspeakers in an audio environment to emit sound and receiving microphone signals from one or more microphones in the audio environment corresponding to an acoustic response of the audio environment to the emitted sound. Some disclosed methods involve detecting, by the control system, a change of the acoustic response of the audio environment. Some disclosed methods involve changing one or more aspects of media processing for media played back by one or more devices in the audio environment based, at least in part, on the change of the acoustic response of the audio environment. Alternatively, or additionally, some disclosed methods may involve changing the lighting in the environment, changing a gain of one or more microphone signals, locking or unlocking one or more devices, or combinations thereof, based at least in part on the change of the acoustic response.

Abstract: Described is a method of performing automatic audio enhancement on an input audio signal including at least one speech-articulation noise event. The method comprises: segmenting the input audio signal into a number of audio frames; obtaining at least one feature parameter from the audio frames; and determining, based at least in part on the obtained feature parameter, a respective type of the speech-articulation noise event and a respective time-frequency range associated with the speech-articulation noise event within the input audio signal.

Abstract: An audio processing method may involve receiving audio signals and associated spatial data, listener position data, loudspeaker position data and loudspeaker orientation data, and rendering the audio data for reproduction, based, at least in part, on the spatial data, the listener position data, the loudspeaker position data and the loudspeaker orientation data, to produce rendered audio signals. The rendering may involve applying a loudspeaker orientation factor that tends to reduce a relative activation of a loudspeaker based, at least in part, on an increased loudspeaker orientation angle. In some examples, the rendering may involve modifying an effect of the loudspeaker orientation factor based, at least in part, on a loudspeaker importance metric. The loudspeaker importance metric may correspond to a loudspeaker's importance for rendering an audio signal at the audio signal's intended perceived spatial position.

Abstract: In some embodiments, a pitch filter for filtering a preliminary audio signal generated from an audio bitstream is disclosed. The pitch filter has an operating mode selected from one of either: (i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal, and (ii) an inactive mode where the pitch filter is disabled. The preliminary audio signal is generated in an audio encoder or audio decoder having a coding mode selected from at least two distinct coding modes, and the pitch filter is capable of being selectively operated in either the active mode or the inactive mode while operating in the coding mode based on control information.

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: Methods and apparatus for estimating metadata for images having absent metadata or unusable form of metadata. According to an example embodiment, a method of estimating metadata includes accessing first and second images of a scene, the first and second images having a first dynamic range (DR) and a different second DR, respectively. The method also includes: generating a third image of the scene having the second DR by applying a mapping function to the first image, the mapping function being configured using an applicable metadata set; generating a sequence of updated metadata sets by iteratively updating the applicable metadata set based on a cost function quantifying a difference between the second image and the third image; and computing values of the cost function to select an output metadata set from the sequence, the output metadata set having estimated metadata for the second image.

Abstract: Described herein is a method for training a machine learning algorithm. The method may comprise receiving a first input multichannel audio signal. The method may comprise generating, using the machine learning algorithm, an intermediate audio signal based on the first input multichannel audio signal. The method may comprise rendering the intermediate audio signal into a first output multichannel audio signal. Further, the method may comprise improving the machine learning algorithm based on a difference between the first input multichannel audio signal and the first output multichannel audio signal. Described herein are further an apparatus for generating an intermediate audio format from an input multichannel audio signal as well as a respective computer program product comprising a computer-readable storage medium with instructions adapted to carry out said method when executed by a device having processing capability.

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: Some examples involve rendering received audio data by determining a first relative activation of a set of loudspeakers in an environment according to a first rendering configuration corresponding to a first set of speaker activations, receiving a first rendering transition indication indicating a transition from the first rendering configuration to a second rendering configuration and determining a second set of speaker activations corresponding to a simplified version of the second rendering configuration. Some examples involve performing a first transition from the first set of speaker activations to the second set of speaker activations, determining a third set of speaker activations corresponding to a complete version of the second rendering configuration and performing a second transition to the third set of speaker activations without requiring completion of the first transition.

Abstract: A projection display system comprises a light source configured to emit a light in response to a content data; an optical modulator configured to modulate the light; and a controller configured to adjust a light level of the projection display system based on the content data and a metadata relating to a future frame, thereby to reduce a perceptibility of a visual artifact.

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

Abstract: Enclosed are embodiments for multisource methods and systems for coded media. In some embodiments, a method comprises: at a first device: receiving media data representing a media asset; obtaining a first plurality of data elements including at least one of bitstream identification data, content-specific encode data and media segment data; encoding at least a portion of the media data in accordance with a first coding process into coded data corresponding to the media asset; generating a second plurality of data elements different from the first plurality of data elements based on information associated with the first coding process; combining the first plurality of data elements and the second plurality of data elements into one or more coded bitstreams representing the media asset; and transmitting the one or more coded bitstreams to one or more second devices using one or more network paths.