Abstract: The present disclosure relates to a method and system for performing packet loss concealment using a neural network system. The method comprises obtaining a representation of an incomplete audio signal, inputting the representation of the incomplete audio signal to an encoder neural network and outputting a latent representation of a predicted complete audio signal. The latent representation is input to a decoder neural network which outputs a representation of a predicted complete audio signal comprising a reconstruction of the original portion of the complete audio signal, wherein said encoder neural network and said decoder neural network have been trained with an adversarial neural network.

Abstract: Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.

Abstract: A system for suppressing noise and enhancing speech and a related method are disclosed. The system trains a neural network model that takes banded energies corresponding to an original noisy waveform and produces a speech value indicating the amount of speech present in each band at each frame. The neural model comprises a feature extraction block that implements some lookahead. The feature extraction block is followed by an encoder with steady down-sampling along the frequency domain forming a contracting path. The encoder is followed by a corresponding decoder with steady up-sampling along the frequency domain forming an expanding path. The decoder receives scaled output feature maps from the encoder at a corresponding level. The decoder is followed by a classification block that generates a speech value indicating an amount of speech present for each frequency band of the plurality of frequency bands at each frame of the plurality of frames.

Abstract: Using a standard-based RGB to YCbCr color transform a new RGB to YCC 3×3 transformation matrix and a 3×1 offset vector are derived under a set of coding-efficiency constraints. The new RGB to YCC 3×3 transform comprises a luminance scaling factor and a 2×2 chroma sub-matrix that preserves the energy of the standard-based RGB to YCbCr transform while maintaining or improving coding efficiency. It also adds support for an authorization or watermarking mechanism in streaming video applications. Examples of using the new color transform using image reshaping are also provided.

Abstract: A global index value is generated for selecting a global reshaping function for an input image of a relatively low dynamic range using luma codewords in the input image. Image filtering is applied to the input image to generate a filtered image. The filtered values of the filtered image provide a measure of local brightness levels in the input image. Local index values are generated for selecting specific local reshaping functions for the input image using the global index value and the filtered values of the filtered image. A reshaped image of a relatively high dynamic range is generated by reshaping the input image with the specific local reshaping functions selected using the local index values.

Abstract: A method (910) for rendering an audio signal in a virtual reality rendering environment (180) is described. The method (910) comprises rendering (911) an origin audio signal of an audio source (311, 312, 313) from an origin source position on an origin sphere (114) around an origin listening position (301) of a listener (181). Furthermore, the method (900) comprises determining (912) that the listener (181) moves from the origin listening position (301) to a destination listening position (302). In addition, the method (900) comprises determining (913) a destination source position of the audio source (311, 312, 313) on a destination sphere (114) around the destination listening position (302) based on the origin source position, and determining (914) a destination audio signal of the audio source (311, 312, 313) based on the origin audio signal.

Abstract: The present invention relates to a method and device for processing a first and a second audio signal representing an input binaural audio signal acquired by a binaural recording device. The present invention further relates to a method for rendering a binaural audio signal on a speaker system. The method for processing a binaural signal comprising extracting audio information from the first audio signal, computing a band gain for reducing noise in the first audio signal and applying the band gains to respective frequency bands of the first audio signal in accordance with a dynamic scaling factor, to provide a first output audio signal. Wherein the dynamic scaling factor has a value between zero and one and is selected so as to reduce quality degradation for the first audio signal.

Abstract: Media input audio data corresponding to a media stream and microphone input audio data from at least one microphone may be received. A first level of at least one of a plurality of frequency bands of the media input audio data, as well as a second level of at least one of a plurality of frequency bands of the microphone input audio data, may be determined. Media output audio data and microphone output audio data may be produced by adjusting levels of one or more of the first and second plurality of frequency bands based on the perceived loudness of the microphone input audio data, of the microphone output audio data, of the media output audio data and the media input audio data. One or more processes may be modified upon receipt of a mode-switching indication.

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.

Abstract: A deep-learning-based system for performing automated multitrack mixing based on a plurality of input audio tracks is described herein. The system comprises one or more instances of a deep-learning-based first network and one or more instances of a deep-learning-based second network. Particularly, the first network is configured to, based on the 5 input audio tracks, generate parameters for use in the automated multitrack mixing. The second network is configured to, based on the parameters, apply signal processing and at least one mixing gain to the input audio tracks, for generating an output mix of the audio tracks.

Abstract: A method for coding includes; segmenting an image into blocks; grouping blocks into a number of subsets; coding, using an entropy coding module, each subset, by associating digital information with symbols of each block of a subset, including, for the first block of the image, initializing state variables of the coding module; and generating a data sub-stream representative of at least one of the coded subsets of blocks. Where a current block is the first block to be coded of a subset, symbol occurrence probabilities for the first current block are determined based on those for a coded and decoded predetermined block of at least one other subset. Where the current block is the last coded block of the subset: writing, in the sub-stream representative of the subset, the entire the digital information associated with the symbols during coding of the blocks of the subset, and implementing the initializing sub-step.

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: 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: 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: A projection system and calibration method therefore relate to a light source configured to emit a light in response to an image data, an optical system configured to project the light emitted by the light source; receiving an input associated with a plurality of light values corresponding to a plurality of primary lightfields; converting the input associated with the plurality of light values to a plurality of projector primary color values; determining a gain map based on the plurality Values of projector primary color values; applying the gain map to an image to perform a chromaticity uniformity correction by adjusting levels of the plurality of primary lightfields so that a primary mixture is the same over an image frame, and projecting the image with the optical system in the image frame, wherein the second image is corrected by the gain map.

Abstract: A method of generating a substitution frame for a lost audio frame of an audio signal is presented. The method may comprise determining an audio filter based on samples of a valid audio frame preceding the lost audio frame. The method may comprise generating the substitution frame based on the audio filter and the samples of the valid audio frame preceding the lost audio frame. The method may be advantageously applied to a low frequency effects (LFE) channel of a multi-channel audio signal.

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: 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 and apparatus for reconstructing N audio channels from M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal representing the M audio channels and decoding the encoded audio signal to obtain a frequency domain representation of the M audio channels. The method further includes extracting a parameter from the bitstream and reconstructing at least one of the N audio channels using the parameter. The parameter represents an angle between two signals, at least one of which is included in the M audio channels.

Abstract: In a cloud-based system for encoding high dynamic range (HDR) video, each node receives a video segment and bumper frames. Each segment is subdivided into primary scenes and secondary scenes to derive scene-based forward reshaping functions that minimize the amount of reshaping-related metadata when coding the video segment, while maintaining temporal continuity among scenes processed by multiple nodes. Methods to generate scene-based forward and backward reshaping functions to optimize video coding and improve the coding efficiency of reshaping-related metadata are also examined.

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: Methods for dialogue enhancing audio content, comprising providing a first audio signal presentation of the audio components, providing a second audio signal presentation, receiving a set of dialogue estimation parameters configured to enable estimation of dialogue components from the first audio signal presentation, applying said set of dialogue estimation parameters to said first audio signal presentation, to form a dialogue presentation of the dialogue components; and combining the dialogue presentation with said second audio signal presentation to form a dialogue enhanced audio signal presentation for reproduction on the second audio reproduction system, wherein at least one of said first and second audio signal presentation is a binaural audio signal presentation.

Abstract: Described is a method of frame-wise encoding metadata for an input signal, the metadata comprising a plurality of at least partially interrelated parameters calculable from the input signal. The method comprises, for each frame: iteratively performing, by using a looping process, steps of: determining a processing strategy from a plurality of processing strategies for calculating and quantizing the parameters; calculating and quantizing the parameters based on the determined processing strategy to obtain quantized parameters; and encoding the quantized parameters. In particular, each of the plurality of processing strategies comprises a respective first indication indicative of an ordering related to the calculation and quantization of individual parameters; and the processing strategy is determined based on at least one bitrate threshold.

Abstract: A method comprising receiving a first input bit stream for a first parametrically coded input audio signal, the first input bit stream including data representing a first input core audio signal and a first set including at least one spatial parameter relating to the first parametrically coded input audio signal. A first covariance matrix of the first parametrically coded audio signal is determined based on the spatial parameter(s) of the first set. A modified set including at least one spatial parameter is determined based on the determined first covariance matrix, wherein the modified set is different from the first set. An output core audio signal is determined, which is based on, or constituted by, the first input core audio signal. An output bit stream for a parametrically coded output audio signal is generated, the output bit stream including data representing the output core audio signal and the modified set.

Abstract: Embodiments are disclosed for a matrix coded stereo signal with periphonic elements. A mixing matrix, suitable for processing a multi-channel audio input signal, is constructed so that the resulting multi-channel output signal contains the same audio elements from the input signal, wherein the spatial relationships between audio elements, as defined by panning rules associated with the input signal format, are preserved in the output signal, as defined by matrix encoding rules associated with the output signal format. The choice of the coefficients of the mixing matrix is governed by a phase-preference rule that is used to determine the appropriate phase to apply to each input signal channel.

Abstract: Embodiments of a configurable loudspeaker using a user orientable routing card that allows for multiple electrical drive modes in a non-powered loudspeaker system. The speaker has one or more drivers mounted in enclosure, an audio input interface configured to be coupled to an audio source through one or more amplifiers, and a connector interface configured to receive a routing card. The routing card is configured to be inserted in a first orientation to connect the audio input interface in a first operating mode with respect to driver selection and connection to the one or more amplifiers, and a second orientation to connect the audio input interface in a second operating mode with respect to driver selection and connection to the one or more amplifiers.

Abstract: A novel projection system includes a base signal source, a highlight signal source, a base/highlight destination, and a shared optical element. A base signal provided by the base source and a highlight signal provided by the highlight source are combined by the shared optical element. In a particular embodiment, the base signal source and the highlight signal source each include a light source, a spatial light modulator, and optics, and the base/highlight destination includes optics and a spatial light modulator. In a more particular embodiment, the base signal source and the highlight source provide spatially modulated lightfields to the shared optical element. In another particular embodiment, the base signal and the highlight signal are modulated by the spatial light modulator of the base/highlight destination after being combined.

Abstract: Apparatuses and methods for data traffic management in multi-source content delivery are described. The apparatus includes a downloader and a controller. The downloader is coupled to servers via communication links. The controller is configured to determine initial download requests for the servers based on predetermined information about a quality of the links. The controller is also configured to send the initial download requests to the servers with the downloader. The controller is further configured to update the information about the quality of the communication links after the downloader receives data associated with a data file from the servers via the communication links. The controller is also configured to determine subsequent download requests for the servers based on the updated information about the quality of the communication links. The controller of further configured to send the subsequent download requests to the servers via the downloader.

Abstract: The present disclosure relates to reverberation generation for headphone virtualization. A method of generating one or more components of a binaural room impulse response (BRIR) for headphone virtualization is described. In the method, directionally-controlled reflections are generated, wherein directionally-controlled reflections impart a desired perceptual cue to an audio input signal corresponding to a sound source location. Then at least the generated reflections are combined to obtain the one or more components of the BRIR. Corresponding system and computer program products are described as well.

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 and determining, based on the output signals, one or more aspects of context information relating to the person, including an estimated current proximity of the person to one or more microphone locations. The method may involve selecting two or more loudspeaker-equipped audio devices based, at least in part, on the one or more aspects of the context information, determining one or more types of audio processing changes to apply to audio data being rendered to loudspeaker feed signals for the audio devices and causing one or more types of audio processing changes to be applied. In some examples, the audio processing changes have the effect of increasing a speech to echo ratio at one or more microphones.

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 one embodiment, there is provided an asymmetrical acoustic horn. The asymmetrical acoustic horn includes a single acoustic waveguide. The single acoustic waveguide includes a first asymmetrical horn section configured to support one or more first acoustic transducers, and a second asymmetrical horn section configured to support one or more second acoustic transducers, the one or more second acoustic transducers having a different frequency range than the one or more first acoustic transducers. The first asymmetrical horn section and the second asymmetrical horn section are contiguous with each other and are configured to separate respective ones of the one or more first acoustic transducers from corresponding ones of the one or more second acoustic transducers by a corresponding one or more predetermined and fixed distances.

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: A computing device comprises a device image display outputting device display light; an optical configuration for a viewer of the computing device to view external display images rendered with external display light from an external image display and device display images rendered with the device display light; a display light combiner to combine the external display light and the device display light to reach the viewer's vision field. The external display light and the device display light are of different light properties. The display light combiner selectively reflects the device display light toward the viewer's vision field and selectively transmits the external display light toward the viewer's vision field.

Abstract: A system utilizing a high throughput coding mode for CABAC in HEVC is described. The system may include an electronic device configured to obtain a block of data to be encoded using an arithmetic based encoder; to generate a sequence of syntax elements using the obtained block; to compare an Absolute-3 value of the sequence or a parameter associated with the Absolute-3 value to a preset value; and to convert the Absolute-3 value to a codeword using a first code or a second code that is different than the first code, according to a result of the comparison.

Abstract: A forward reshaping mapping is generated to map a source image to a corresponding forward reshaped image of a lower dynamic range. The source image is spatially downsampled to generate a resized image into which noise is injected to generate a noise injected image. The forward reshaping mapping is applied to map the noise injected image to generate a noise embedded image of the lower dynamic range. A video signal is encoded with the noise embedded image and delivered to a recipient device for the recipient device to render a display image generated from the noise embedded image.

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: 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: Light from an array of laser light sources are spread to cover the modulating face of a DMD or other modulator. The spread may be performed, for example, by a varying curvature array of lenslets, each laser light directed at one of the lenslets. Light from neighboring and/or nearby light sources overlap at a modulator. The lasers are energized at different energy/brightness levels causing the light illuminating the modulator to itself be modulated (locally dimmed). The modulator then further modulates the locally dimmed lights to produce a desired image. A projector according to the invention may utilize, for example, a single modulator sequentially illuminated or separate primary color modulators simultaneously illuminated.

Abstract: A multi-view image stream encoded with primary and secondary image is accessed. Each primary image stream comprises groups of pictures (GOPs). Each secondary image stream comprises I-frames generated from a corresponding primary image stream. Viewpoint data collected in real time is received from a recipient decoding device to indicate that the viewer's viewpoint has changed from a specific time point. A camera is selected based on the viewer's changed viewpoint. It is determined whether the specific time point corresponds to a non-I-frame in a GOP of a primary image stream of the selected camera. If so, an I-frame from a secondary image stream corresponding to the primary image stream is transmitted to the recipient decoding device.

Abstract: A non-random-access video stream is received. A first image block is encoded after second image blocks according to a non-random-access processing order. View direction data is received to indicate a viewer's view direction coinciding with a location covered by the first image block. The first image block is encoded into the random-access video stream before the second image blocks in a random-access processing order. The random-access video stream is delivered to a recipient decoding device operated by the viewer to cause the first image block to be processed and rendered before the second image blocks according to the random-access processing order.

Abstract: Methods and systems for generating a set of forward and backward reshaping functions for the efficient coding of high-dynamic range (HDR) images are provided. Given an initial set of forward reshaping functions, output forward reshaping functions are constructed by a) using the forward reshaping functions to generate a first set of corresponding backward reshaping functions b) generating a second set of backward reshaping functions using a multi-segment polynomial representation with a common set of pivot points c) generating an output set of backward reshaping functions by optimizing the polynomial representation of the second set of backward reshaping functions to minimize gap values between consecutive segments and d) using the output set of backward reshaping functions to generate the output set of forward reshaping functions by minimizing the distance between original input HDR codewords and reconstructed HDR codewords.

Abstract: Described are methods of processing audio data for hum noise detection and/or removal. The audio data comprises a plurality of frames. One method incudes: classifying frames of the audio data as either content frames or noise frames, using one or more content activity detectors; determining a noise spectrum from one or more frames of the audio data that are classified as noise frames; determining one or more hum noise frequencies based on the determined noise spectrum; generating an estimated hum noise signal based on the one or more hum noise frequencies; and removing hum noise from at least one frame of the audio data based on the estimated hum noise signal. Also described are apparatus for carrying out the methods, as well as corresponding programs and computer-readable storage media.

Abstract: In a cloud-based system for encoding high dynamic range (HDR) video, a computing node is assigned to be a dispatcher node, segmenting the input video into scenes and generating a scene to segment allocation to be used by other computing nodes. The scene to segment allocation process includes one or more iterations with an initial random assignment of scenes to computing nodes, followed by a refined assignment based on optimizing the allocation cost across all the computing nodes. Methods to generate scene-based forward and backward reshaping functions to optimize video coding and improve the coding efficiency of reshaping-related metadata are also examined.

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: Systems and methods of rendering DCI-compliant image data on Enhanced Dynamic Range (EDR) display systems are disclosed. One embodiment of an EDR projector system comprises a first modulator and a second modulator. One method for rendering DCI-compliant image data on an EDR projector system comprises: receiving input image data, said image data comprising a plurality of image formats; determining whether the input image data comprises DCI image data; if the input image data comprises DCI image data, then performing dynamic range (DR) processing on the DCI image data; and rendering the dynamic range processed DCI image data on the EDR projector system. One DR processing method is to set the first modulator to a desired luminance level—e.g., fully ON or a ratio of DCI max luminance to the EDR max luminance. In addition, a desired minimum level of luminance may be set for the EDR projector.

Abstract: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder.

Abstract: 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: Novel methods and systems for compensating for ambient light around displays are disclosed. A shift in the PQ curve applied to an image can compensate for sub-optimal ambient light conditions for a display, with the PQ shift being either an addition to a compensation value in PQ space followed by a subtraction of the compensation value in linear space, or an addition to the compensation value in linear space followed by a subtraction of the compensation value in PQ space. Further adjustments to the PQ curve can also be made to provide an improved image quality with respect to image luminance.

Abstract: One or more media contents are received. A viewer's light adaptive states are predicted as a function of time as if the viewer is watching display mapped images derived from the one or more media contents. The viewer's light adaptive states are used to detect an excessive change in luminance in a specific media content portion of the one or more media contents. The excessive change in luminance in the specific media content portion of the one or more media contents is caused to be reduced while the viewer is watching one or more corresponding display mapped images derived from the specific media content portion of the one or more media contents.