Abstract: When compressing an HOA data frame representation, a gain control (15, 151) is applied for each channel signal before it is perceptually encoded (16). The gain values are transferred in a differential manner as side information. However, for starting decoding of such streamed compressed HOA data frame representation absolute gain values are required, which should be coded with a minimum number of bits. For determining such lowest integer number (?e) of bits the HOA data frame representation (C(k)) is rendered in spatial domain to virtual loudspeaker signals lying on a unit sphere, followed by normalisation of the HOA data frame representation (C(k)). Then the lowest integer number of bits is set to ? e = ? log 2 ( ? log 2 ( K M ? A ? X · O ) ? + 1 ) ? .

Abstract: Methods are described to communicate source color volume information in a coded bitstream using SEI messaging. Such data include at least the minimum, maximum, and average luminance values in the source data plus optional data that may include the color volume x and y chromaticity coordinates for the input color primaries (e.g., red, green, and blue) of the source data, and the color x and y chromaticity coordinates for the color primaries corresponding to the minimum, average, and maximum luminance values in the source data. Messaging data signaling an active region in each picture may also be included.

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

Abstract: Methods are described to communicate source color volume information in a coded bitstream using SEI messaging. Such data include at least the minimum, maximum, and average luminance values in the source data plus optional data that may include the color volume x and y chromaticity coordinates for the input color primaries (e.g., red, green, and blue) of the source data, and the color x and y chromaticity coordinates for the color primaries corresponding to the minimum, average, and maximum luminance values in the source data. Messaging data signaling an active region in each picture may also be included.

Abstract: A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

Abstract: An audio processing system and method which calculates, based on spatial metadata of the audio object, a panning coefficient for each of the audio objects in relation to each of a plurality of predefined channel coverage zones. Converts the audio signal into submixes in relation to the predefined channel coverage zones based on the calculated panning coefficients and the audio objects. Each of the submixes indicating a sum of components of the plurality of the audio objects in relation to one of the predefined channel coverage zones. Generating a submix gain by applying an audio processing to each of the submix and controls an object gain applied to each of the audio objects. The object gain being as a function of the panning coefficients for each of the audio objects and the submix gains in relation to each of the predefined channel coverage zones.

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: The method for decoding an intra-picture prediction mode includes the steps of: determining whether the intra-picture prediction mode of a current prediction unit is identical to a first intra-picture prediction mode candidate or a second intra-picture prediction mode candidate based on bit information; and when the intra-picture prediction mode of the current prediction unit is identical to the first intra-picture prediction mode candidate and/or to the second intra-picture prediction mode candidate, determining whether the first intra-picture prediction mode candidate or the second intra-picture prediction mode candidate is identical to the intra-picture prediction mode of the current prediction unit on the basis of additional bit information, and decoding the intra-picture prediction mode of the current prediction unit.

Abstract: There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.

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: Systems and methods for providing a binaural virtualization by upmixing the left and right input signals to produce left, right, and center channels, mixing the left and right input signals with the upmixed left and right channels respectively at a proportion given by a center-only reverb amount value, then reverberating the output of the mixing prior to virtualization. This can be further simplified by mode switching between two different filtering modes: a standard mode and a simplified mode.

Abstract: A method for encoding an image having been cut up into partitions. The method includes: predicting data of a current partition based on an already encoded and then decoded reference partition, generating a predicted partition; determining residual data by comparing data relating to the current partition with the predicted partition, the residual data associated with various digital data items. Prior producing a signal containing the encoded information, performing the following steps; determining, from the predetermined residual data, a subset containing residual data capable of being modified; calculating the value of a function representative of the residual data; comparing the calculated value with a value of at least one of the digital data items; based on the comparison, modification or non-modification of at least one of the residual data items of the subset; and, in the event of a modification, entropy encoding the at least one modified residual data item.

Abstract: Described is a method of processing position information indicative of an object position of an audio object, wherein the object position is usable for rendering of the audio object, that comprises: obtaining listener orientation information indicative of an orientation of a listener's head; obtaining listener displacement information indicative of a displacement of the listener's head; determining the object position from the position information; modifying the object position based on the listener displacement information by applying a translation to the object position; and further modifying the modified object position based on the listener orientation information. Further described is a corresponding apparatus for processing position information indicative of an object position of an audio object, wherein the object position is usable for rendering of the audio object.

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: 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: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. For coding, portions of the original HOA representation are predicted from the directional signal components. This prediction provides side information which is required for a corresponding decoding. By using some additional specific purpose bits, a known side information coding processing is improved in that the required number of bits for coding that side information is reduced on average.

Abstract: The disclosure relates to methods, apparatus and systems for side load processing of packetized media streams. In an embodiment, the apparatus comprises: a receiver for receiving a bitstream, and a splitter for identifying a packet type in the bitstream and splitting, based on the identification of a value of the packet type in the bit stream into a main stream and an auxiliary stream.

Abstract: A method for clustering audio objects may involve identifying a plurality of audio objects, wherein each audio object of the plurality of audio objects is associated with respective metadata that indicates respective spatial position information and respective rendering metadata. The method may involve assigning audio objects of the plurality of audio objects to categories of rendering metadata of a plurality of categories of rendering metadata, wherein at least one category of rendering metadata comprises a plurality of types of rendering metadata to be preserved. The method may involve determining an allocation of a plurality of audio object clusters to each category of rendering metadata. The method may involve rendering audio objects of the plurality of audio objects to an allocated plurality of audio object clusters based on the metadata that indicates spatial position information and based on the assignments of the audio objects to the categories of rendering metadata.

Abstract: Some methods involve causing a plurality of audio devices in an audio environment to reproduce audio data, each audio device of the plurality of audio devices including at least one loudspeaker and at least one microphone, determining audio device location data including an audio device location for each audio device of the plurality of audio devices and obtaining microphone data from each audio device of the plurality of audio devices. Some methods involve determining a mutual audibility for each audio device of the plurality of audio devices relative to each other audio device of the plurality of audio devices, determining a user location of a person in the audio environment, determining a user location audibility of each audio device of the plurality of audio devices at the user location and controlling one or more aspects of audio device playback based, at least in part, on the user location audibility.

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

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

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

Abstract: Methods and systems for frame rate scalability are described. Support is provided for input and output video sequences with variable frame rate and variable shutter angle across scenes, or for input video sequences with fixed input frame rate and input shutter angle, but allowing a decoder to generate a video output at a different output frame rate and shutter angle than the corresponding input values. Techniques allowing a decoder to decode more computationally-efficiently a specific backward compatible target frame rate and shutter angle among those allowed are also presented.

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

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

Abstract: A control unit of a multipath data transportation system that optimizes the load of the multiple communication paths of this system when the system transmits a data segment over these paths in parallel with forward error correction. The control unit determines an optimized number of packets to send over each path based on a prediction of quality for each path. The transmitted packets include systematic packets and coded packets.

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

Abstract: A digital PSF for use in a dual modulation display. The invention allows the use of less than optimal point spread (PSF) functions in the optics between the pre-modulator and primary modulator of a dual modulation projection system. This technique uses multiple halftones per frame in the pre-modulator synchronized with a modified bit sequence in the primary modulator to produce a compensation image that reduces the errors produced by the sub-optimal PSF. The invention includes the application to dual modulation and dual modulated 3D viewing systems.

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

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

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

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

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

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

Abstract: Methods and systems for frame rate scalability are described. Support is provided for input and output video sequences with variable frame rate and variable shutter angle across scenes, or for input video sequences with fixed input frame rate and input shutter angle, but allowing a decoder to generate a video output at a different output frame rate and shutter angle than the corresponding input values. Techniques allowing a decoder to decode more computationally-efficiently a specific backward compatible target frame rate and shutter angle among those allowed are also presented.

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

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

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

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

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

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

Abstract: Methods and systems for performing at least one audio activity (e.g., conducting a phone call or playing music or other audio content) in an environment including by determining an estimated location of a user in the environment in response to sound uttered by the user (e.g., a voice command), and controlling the audio activity in response to determining the estimated user location. The environment may have zones which are indicated by a zone map and estimation of the user location may include estimating in which of the zones the user is located. The audio activity may be performed using microphones and loudspeakers which are implemented in or coupled to smart audio devices.

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

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

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

Abstract: An audio session management method may involve: determining, by an audio session manager, one or more first media engine capabilities of a first media engine of a first smart audio device, the first media engine being configured for managing one or more audio media streams received by the first smart audio device and for performing first smart audio device signal processing for the one or more audio media streams according to a first media engine sample clock; receiving, by the audio session manager and via a first application communication link, first application control signals from the first application; and controlling the first smart audio device according to the first media engine capabilities, by the audio session manager, via first audio session management control signals transmitted to the first smart audio device via a first smart audio device communication link and without reference to the first media engine sample clock.

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

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

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

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