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: 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: Example embodiments disclosed herein relate to a transducer assembly and associated signal processing. A transducer assembly includes two voice coils in a telescopic arrangement and having unequal sizes, and two suspension systems connected to the two voice coils, respectively. The two voice coils extend in opposites directions from their suspension systems. Dimensions of respective wires of the two voice coils are determined based on respective magnetic flux densities in magnetic gaps for receiving the two voice coils. As a result, a residual vibration caused by the unequal-sized voice coils can be further reduced.

Abstract: A method of noise reduction includes using a neural network to control a Wiener filter. The gains estimated by the neural network are combined with the gains produced by the Wiener filter. In this manner, the noise reduction system provides improved results as compared to using only a neural network.

Abstract: In some embodiments, a method for performing enhancement on an audio signal to generate an enhanced audio signal in response to feedback indicative of amount of compression applied to at least one frequency band of the enhanced audio signal. In typical embodiments, the enhancement is or includes bass enhancement. Examples of other types of enhancement performed in other embodiments include dialog enhancement, upmixing, frequency shifting, harmonic injection or transposition, subharmonic injection, virtualization, and equalization. Other aspects are systems (e.g., programmed processors) and devices (e.g., devices having physically-limited bass reproduction capabilities, such as, for example, a notebook, tablet, mobile phone, or other device with small speakers) configured to perform any embodiment of the method.

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: 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 of coding at least one image comprising the steps of splitting the image into a plurality of blocks, of grouping said blocks into a predetermined number of subsets of blocks, of coding each of said subsets of blocks in parallel, the blocks of a subset considered being coded according to a predetermined sequential order of traversal. The coding step comprises, for a current block of a subset considered, the sub-step of predictive coding of said current block with respect to at least one previously coded and decoded block, and the sub-step of entropy coding of said current block on the basis of at least one probability of appearance of a symbol.

Abstract: Methods, systems, and computer program products that provide streaming capabilities to audio input and output devices are disclosed. An audio processing device connects an upstream device to a downstream device. The upstream device is plugged into an input port of the audio processing device. The audio processing device intercepts a signal from the upstream device to the downstream device. The audio processing device converts the signal to digital data and streams the digital data to a server. The digital data can include metadata, e.g., an input gain. The audio processing device can adjust the input gain in response to instructions from the server. The audio processing device feeds a pass-through copy of the audio signal to an output port. A user can connect the downstream device in a usual signal chain into the output port of the audio processing device. The streaming does not affect the user's workflow.

Abstract: Diffuse or spatially large audio objects may be identified for special processing. A decorrelation process may be performed on audio signals corresponding to the large audio objects to produce decorrelated large audio object audio signals. These decorrelated large audio object audio signals may be associated with object locations, which may be stationary or time-varying locations. For example, the decorrelated large audio object audio signals may be rendered to virtual or actual speaker locations. The output of such a rendering process may be input to a scene simplification process. The decorrelation, associating and/or scene simplification processes may be performed prior to a process of encoding the audio data.

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: Creative intent input describing emotion expectations and narrative information relating to media content is received. Expected physiologically observable states relating to the media content are generated based on the creative intent input. An audiovisual content signal with the media content and media metadata comprising the physiologically observable states is provided to a playback apparatus. The audiovisual content signal causes the playback device to use physiological monitoring signals to determine, with respect to a viewer, assessed physiologically observable states relating to the media content and generate, based on the expected physiologically observable states and the assessed physiologically observable states, modified media content to be rendered to the viewer.

Abstract: Methods and systems for designing binaural room impulse responses (BRIRs) for use in headphone virtualizers, and methods and systems for generating a binaural signal in response to a set of channels of a multi-channel audio signal, including by applying a BRIR to each channel of the set, thereby generating filtered signals, and combining the filtered signals to generate the binaural signal, where each BRIR has been designed in accordance with an embodiment of the design method. Other aspects are audio processing units configured to perform any embodiment of the inventive method. In accordance with some embodiments, BRIR design is formulated as a numerical optimization problem based on a simulation model (which generates candidate BRIRs) and at least one objective function (which evaluates each candidate BRIR), and includes identification of a best one of the candidate BRIRs as indicated by performance metrics determined for the candidate BRIRs by each objective function.

Abstract: A novel spatial light modulator (SLM) includes a cover glass, and modulation layer, and a plurality of pixel mirrors, and separates unwanted, reflected light from desired, modulated light. In one embodiment, a geometrical relationship exists between the cover glass and the pixel mirrors, such that light that reflects from the cover glass is separated from light that reflects from the pixel mirrors and is transmitted from the SLM. In one example, one of the cover glass or the pixel mirrors 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: Sample data and metadata related to spatial regions in images may be received from a coded video signal. It is determined whether specific spatial regions in the images correspond to a specific region of luminance levels. In response to determining the specific spatial regions correspond to the specific region of luminance levels, signal processing and video compression operations are performed on sets of samples in the specific spatial regions. The signal processing and video compression operations are at least partially dependent on the specific region of luminance levels.

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.

Abstract: The present invention provides a method and a device for deriving an inter-view motion merging candidate. A method for deriving an inter-view motion merging candidate, according to an embodiment of the present invention, can comprise the steps of: on the basis of encoding information of an inter-view reference block derived by means of a variation vector of a current block, determining whether or not inter-view motion merging of the current block is possible; and, if inter-view motion merging of the current block is not possible, generating an inter-view motion merging candidate of the current block by using encoding information of an adjacent block that is spatially adjacent to the inter-view reference block.

Abstract: Methods for generating an object based audio program which is renderable in a personalizable manner, e.g., to provide an immersive, perception of audio content of the program. Other embodiments include steps of delivering (e.g., broadcasting), decoding, and/or rendering such a program. Rendering of audio objects indicated by the program may provide an immersive experience. The audio content of the program may be indicative of multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects, and typically also a default set of objects which will be rendered in the absence of a selection by a user) and a bed of 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: Given input HDR and SDR images representing the same scene, a prediction model to predict the HDR image from a compressed representation of the input SDR image is generated as follows: a) generate noise data based at least on the characteristics of the HDR image b) generate a noisy SDR image by adding the noise data to the SDR image c) generate an augmented HDR data set and an augmented SDR data set by using the input HDR and SDR images and the noisy SDR image d) generate a prediction model to predict the augmented HDR data set based on the augmented SDR data set and e) solve the prediction model according to a minimization-error criterion to generate a set of prediction parameters to be transmitted to a decoder together with a compressed representation of the input SDR image to reconstruct an approximation of the input HDR image.

Abstract: Some implementations involve analyzing audio packets received during a time interval that corresponds with a conversation analysis segment to determine network jitter dynamics data and conversational interactivity data. The network jitter dynamics data may provide an indication of jitter in a network that relays the audio data packets. The conversational interactivity data may provide an indication of interactivity between participants of a conversation represented by the audio data. A jitter buffer size may be controlled according to the network jitter dynamics data and the conversational interactivity data. The time interval may include a plurality of talkspurts.

Abstract: Images of a user’s head are acquired at a plurality of different orientational angles through image sensors operating in conjunction with a media consumption system. The acquired images of the user’s head are used to select or predict a specific personalized head related transfer function for the user. Spatial audio rendered by audio speakers operating in conjunction with the media consumption system is adjusted or modified based at least in part on the specific personalized HRTF selected for the user.

Abstract: A method for compressing a HOA signal being an input HOA representation with input time frames (C(k)) of HOA coefficient sequences comprises spatial HOA encoding of the input time frames and subsequent perceptual encoding and source encoding. Each input time frame is decomposed (802) into a frame of predominant sound signals (XPS(k?1)) and a frame of an ambient HOA component ({tilde over (C)}AMB(k?1)). The ambient HOA component ({tilde over (C)}AMB(k?1)) comprises, in a layered mode, first HOA coefficient sequences of the input HOA representation (cn(k?1)) in lower positions and second HOA coefficient sequences (cAMB,n(k?1)) in remaining higher positions. The second HOA coefficient sequences are part of an HOA representation of a residual between the input HOA representation and the HOA representation of the predominant sound signals.

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: Audio signals from microphones of a mobile device are received. Each audio signal is generated by a respective microphone of the microphones. First microphones are selected from among the microphones to generate a front audio signal. Second microphones are selected from among the microphones to generate a back audio signal. A first audio signal portion, which is determined based at least in part on the back audio signal, is removed from the front audio signal to generate a modified front audio signal. A second audio signal portion is removed from the modified front audio signal to generate a left-front audio signal. A third audio signal portion is removed from the modified front audio signal to generate aright-front audio signal.

Abstract: An audio bitstream is decoded into audio objects and audio metadata for the audio objects. The audio objects include a specific audio object. The audio metadata specifies frame-level gains that include a first gain and a second gain respectively for a first audio frame and a second audio frame. It is determined, based on the first and second gains, whether sub-frame gains are to be generated for the specific audio object. If so, a ramp length is determined for a ramp used to generate the sub-frame gains for the specific audio object. The ramp of the ramp length is used to generate the sub-frame gains for the specific audio object. A sound field represented by the audio objects with the sub-frame gains is rendered by audio speakers.

Abstract: Described is a method of training a neural-network-based system for determining an indication of an audio quality of an audio input. The method includes obtaining, as input, at least one training set comprising audio samples. The audio samples include audio samples of a first type and audio samples of a second type, wherein each of the first type of audio samples is labelled with information indicative of a respective predetermined audio quality metric, and wherein each of the second type of audio samples is labelled with information indicative of a respective audio quality metric relative to that of a reference audio sample. The method further includes: inputting the training set to the neural-network-based system; and iteratively training the system to predict the respective label information of the audio samples in the training set.

Abstract: In an embodiment, a method comprises: transforming one or more frames of a two-channel time domain audio signal into a time-frequency domain representation including a plurality of time-frequency tiles, wherein the frequency domain of the time-frequency domain representation includes a plurality of frequency bins grouped into subbands. For each time-frequency tile, the method comprises: calculating spatial parameters and a level for the time-frequency tile; modifying the spatial parameters using shift and squeeze parameters; obtaining a softmask value for each frequency bin using the modified spatial parameters, the level and subband information; and applying the softmask values to the time-frequency tile to generate a modified time-frequency tile of an estimated audio source.

Abstract: The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular, a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described. The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of ?f.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding a stereo audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and a discrete representation of the stereo audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.

Abstract: In an embodiment, a spatio-level filter (SLF) is created by obtaining a first set of samples from a plurality of target source level and spatial distributions in frequency subbands in a frequency domain, obtaining a second set of samples from a plurality of background level and spatial distributions in frequency subbands in a frequency domain, adding the first and second sets of samples to create a combined set of samples, detecting level and spatial parameters for each sample in the combined set of samples for each subband, within subbands, weighting the detected level and spatial parameters by their respective level and spatial distributions for the target source and backgrounds; storing the weighted level, spatial parameters and signal-to-noise ratio (SNR) within subbands for each sample in the combined set of samples in a table; and re-indexing the table by the weighted level and spatial parameters and subband.

Abstract: Described herein is a method for improving dialogue intelligibility during playback of audio data on a playback device, wherein the audio data comprise dialogue audio data, and at least one of music and effects audio data, the method including the steps of: determining a volume mixing ratio based on a volume value for playback; mixing the dialogue audio data and the at least one of music and effects audio data based on said volume mixing ratio; and outputting the mixed audio data for playback. Described are further a respective playback device and a respective computer program product.

Abstract: One or more perforation hole pattern methods are applied (402) to generate a spatial distribution of perforation holes forming a semi-random pattern for an image display screen. The image display screen is perforated (404) with the spatial distribution of perforation holes forming the semi-random pattern. Image rendering light is emitted (406) with a light projector toward the image display screen that is installed in an image rendering environment. At least a portion of the image rendering light emitted from the light projector is reflected (408) by the image display screen, toward a viewer.

Abstract: The present document relates an audio encoding and decoding system (referred to as an audio codec system). In particular, the present document relates to a audio codec system which is particularly well suited for voice encoding/decoding. A transform-based speech encoder is configured to encode a speech signal into a bitstream is described. A speech decoder configured to decode audio signals from a bitstream is further described.

Abstract: The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor S.

Abstract: Methods and audio processing units for generating an object based audio program including conditional rendering metadata corresponding to at least one object channel of the program, where the conditional rendering metadata is indicative of at least one rendering constraint, based on playback speaker array configuration, which applies to each corresponding object channel, and methods for rendering audio content determined by such a program, including by rendering content of at least one audio channel of the program in a manner compliant with each applicable rendering constraint in response to at least some of the conditional rendering metadata. Rendering of a selected mix of content of the program may provide an immersive experience.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: Methods and systems are described for processing an image captured with an image sensor, such as a camera. In one embodiment, an estimated ambient light level of the captured image is determined and used to compute an optical-optical transfer function (OOTF) that is used to correct the image to preserve an apparent contrast of the image under the estimated ambient light level in a viewing environment. The estimated ambient light level is determined by scaling pixel values from the image sensor using a function that includes exposure parameters and a camera specific parameter derived from a camera calibration.

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: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: A method for distributing High Dynamic Range (HDR) content to playback devices for displaying images where the HDR content is encoded to an HDR bitstream and the HDR bitstream is subsequently decoded by a playback device. The HDR bitstream contains auxiliary metadata packets that are based upon the processing capability of the playback device.

Abstract: Some methods involve receiving an input audio signal that includes N input audio channels, the input audio signal representing a first soundfield format having a first soundfield format resolution, N being an integer ?2. A first decorrelation process may be applied to two or more of the input audio channels to produce a first set of decorrelated channels, the first decorrelation process maintaining an inter-channel correlation of the set of input audio channels. A first modulation process may be applied to the first set of decorrelated channels to produce a first set of decorrelated and modulated output channels. The first set of decorrelated and modulated output channels may be combined with two or more undecorrelated output channels to produce an output audio signal that includes O output audio channels representing a second and relatively higher-resolution soundfield format than the first soundfield format, O being an integer ?3.

Abstract: Embodiments are disclosed for non-intrusive transducer health detection in an audio system. In an embodiment, a method performed by the audio system comprises outputting one or more encoded inaudible acoustic signals into an acoustic transmission medium using a first transducer. The one or more encoded inaudible acoustic signals are received from the acoustic transmission medium using a second transducer of the audio system. The received one or more encoded inaudible acoustic signals are used to identify failure or degradation of the first or second transducer.

Abstract: Described herein is a method of determining parameters for a generative neural network for processing an audio signal, wherein the generative neural network includes an encoder stage mapping to a coded feature space and a decoder stage, each stage including a plurality of convolutional layers with one or more weight coefficients, the method comprising a plurality of cycles with sequential processes of: pruning the weight coefficients of either or both stages based on pruning control information, the pruning control information determining the number of weight coefficients that are pruned for respective convolutional layers; training the pruned generative neural network based on a set of training data; determining a loss for the trained and pruned generative neural network based on a loss function; and determining updated pruning control information based on the determined loss and a target loss. Further described are corresponding apparatus, programs, and computer-readable storage media.

Abstract: Embodiments are disclosed for projection systems with rotatable anamorphic lenses. In an embodiment, an optical projection system comprises: a light source; an optical integrator configured to receive light from the light source and to distribute a uniform pattern of light; a relay lens system including two or more rotatable anamorphic lenses, the anamorphic lenses oriented about an optical axis to transform the uniform pattern of light into an image having a specified aspect ratio; at least one spatial light modulator configured to receive the image and direct a spatially modulated image along an optical path; and at least one projection lens configured to receive the spatially modulated image from the optical path and to project the spatially modulated image onto an image plane with the specified aspect ratio.

Abstract: In some embodiments, a method for performing at least one of enhancement, decoding, or rendering of a multichannel audio signal in response to compression feedback or feedback from a smart amplifier. For example, the compression feedback may be indicative of amount of compression applied to each of multiple frequency bands, of the audio signal or an enhanced audio signal generated in response thereto. The enhancement (e.g., bass enhancement) may include dynamic routing of audio content of the input audio signal between channels of an enhanced audio signal generated in response thereto. The enhancement and compression may be performed on a per speaker class basis. Other aspects are systems (e.g., programmed processors) and devices (e.g., devices having physically-limited bass reproduction capabilities, such as, for example, a notebook or laptop computer, tablet, soundbar, mobile phone, or other device with small speakers) configured to perform any embodiment of the method.

Abstract: A computer-implemented method of audio processing, the method comprising: receiving audio object data and audio description data, wherein the audio object data includes a first plurality of audio objects; calculating a long-term loudness of the audio object data and a long- term loudness of the audio description data; calculating a plurality of short-term loudnesses of the audio object data and a plurality of short-term loudnesses of the audio description data; reading a first plurality of mixing parameters that correspond to the audio object data; generating a second plurality of mixing parameters based on the first plurality of mixing parameters, the long-term loudness of the audio object data, the long-term loudness of the audio description data, the plurality of short-term loudnesses of the audio object data, and the plurality of short-term loudnesses of the audio description data; generating a gain adjustment visualization corresponding to the second plurality of mixing parameters, the audio object data

Abstract: A content-creation tool includes a processor and a memory. The processor is configured to receive a first video clip and a second video clip, a respective first and second metadata-item thereof being set to a respective first and second metadata-value. The memory stores video-editing software that includes a timeline interface and instructions that, when executed by the processor, control the processor to: add the first video clip to the timeline interface as a first timeline-track that retains the first metadata-value; add the second video clip to the timeline interface as a second timeline-track that retains the second metadata-value; and generate a frame sequence that includes a plurality of video frames. Each video frame is a frame of, or a frame derived from, one of (i) the first timeline-track, (ii) the second timeline-track, and (iii) a composited time-line-track composited from at least one of the first and second timeline-tracks.

Abstract: A method comprises: obtaining softmask values for frequency bins of time-frequency tiles representing an audio signal; reducing, or expanding and limiting, the softmask values; and applying the reduced, or expanded and limited, softmask values to the frequency bins to create a time-frequency representation of an estimated target source. An alternative method comprises, for each time-frequency tile: obtaining softmask values; applying the softmask values to the frequency bins to create a time-frequency domain representation of an estimated target source; obtaining a panning parameter and a source concentration estimates for the target source; determining, using the panning parameter estimate and the softmask values, a magnitude for the time-frequency representation of the estimated target source; determining, using the panning parameter estimate and the source phase concentration estimate, a phase for the time-frequency representation of the estimated target source; and combining the magnitude and the phase.

Abstract: A system and method of editing video content includes receiving input video data; converting the input video data to a predetermined format; generating a plurality of initial metadata values for a frame of the converted video data, the plurality of initial metadata values including a first metadata value corresponding to a first fixed value not calculated from a content including the frame, a second metadata value corresponding to an average luminance value of the frame, and a third metadata value corresponding to a second fixed value not calculated from the content, wherein the first meta-data value, the second metadata value, and the third metadata value include information used by a decoder to render a decoded image on a display.

Abstract: A method for representing a second presentation of audio channels or objects as a data stream, the method comprising the steps of: (a) providing a set of base signals, the base signals representing a first presentation of the audio channels or objects; (b) providing a set of transformation parameters, the transformation parameters intended to transform the first presentation into the second presentation; the transformation parameters further being specified for at least two frequency bands and including a set of multi-tap convolution matrix parameters for at least one of the frequency bands.