Abstract: This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal.

Abstract: An input image represented in an input domain is received from an input video signal. Forward reshaping is performed on the input image to generate a forward reshaped image represented in a reshaped image domain. Non-reshaping encoding operations are performed to encode the reshaped image into an encoded video signal. At least one of the non-reshaping encoding operations is implemented with an ML model that has been previously trained with training images in one or more training datasets in a preceding training stage. A recipient device of the encoded video signal is caused to generate a reconstructed image from the forward reshaped image.

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

Abstract: A multi-stream rendering system and method may render and play simultaneously a plurality of audio program streams over a plurality of arbitrarily placed loudspeakers. At least one of the program streams may be a spatial mix. The rendering of said spatial mix may be dynamically modified as a function of the simultaneous rendering of one or more additional program streams. The rendering of one or more additional program streams may be dynamically modified as a function of the simultaneous rendering of the spatial mix.

Abstract: The described embodiments include systems and methods for producing and adapting images, such as video images, for presentation on display devices that have various different aspects ratios, such as 4:3, 16:9, 9:16, etc. In one embodiment, a method for producing content, such as video images, can begin by selecting an original aspect ratio and determining, within at least a first scene in the content, a position of a subject in the first scene. In one embodiment, the original aspect ratio can be substantially square (e.g., 1:1). Metadata can then be created, based on the position of the subject in the first scene, to guide playback devices to asymmetrically crop the content, relative to the position, for display on display devices that have aspect ratios that are different than the original aspect ratio. Other methods and systems are also described.

Abstract: A method for providing media content within a media distribution network. The method comprises transforming source media content into an interim format, thereby providing transformed content. Furthermore, the method comprises storing the transformed content on at least one core storage unit. In addition, the method comprises receiving a request for the source media content from a client. The method further comprises encoding the transformed content or intermediate coded content derived therefrom into encoded content suitable for transmission over a core network and/or an edge network, as well as sending the encoded content via the core network and/or the edge network to the client.

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: A method for channel identification of a multi-channel audio signal comprising X>1 channels is provided. The method comprises the steps of: identifying, among the X channels, any empty channels, thus resulting in a subset of Y?X non-empty channels; determining whether a low frequency effect (LFE) channel is present among the Y channels, and upon determining that an LFE channel is present, identifying the determined channel among the Y channels as the LFE channel; dividing the remaining channels among the Y channels not being identified as the LFE channel into any number of pairs of channels by matching symmetrical channels; and identifying any remaining unpaired channel among the Y channels not being identified as the LFE channel or divided into pairs as a center channel.

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: 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: 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 video content controller includes a memory and a microprocessor. The memory is configured to store non-transitory computer-readable instructions and video data representing a temporally-varying scene having a plurality of scene-regions. The microprocessor adapted to execute the instructions to (i) receive a current stream-segment of a video stream corresponding to a first scene-region of the plurality of scene-regions during a current playback-time interval, (ii) download, to a memory, a first video segment of a tagged scene-region within the first scene-region, (iii) receive a first display-region signal indicating at least one of a selection and a feature of the first scene-region, (iv) combine the current stream-segment and the first video segment as a video signal, and (vi) transmit, in response to the first display-region signal, the video signal to a display device.

Abstract: The disclosure herein generally relates to capturing, acoustic pre-processing, encoding, decoding, and rendering of directional audio of an audio scene. In particular, it relates to a device adapted to modify a directional property of a captured directional audio in response to spatial data of a microphone system capturing the directional audio. The disclosure further relates to a rendering device configured to modify a directional property of a received directional audio in response to received spatial data.

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: 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: Input images are received as input to a multi-node system. The input images are divided into segments assigned to respective nodes of the multi-node system. Primary and secondary scenes are identified in the segments to ensure compliance with minimum and average distance constraints. Scene-level forward reshaping mappings are generated for the scenes by a respective node for an assigned segment. Forward reshaped images in the segment are generated by the node using the forward reshaping mappings and encoded into an output video signal, which enables a recipient device to generate reconstructed images and to render display images derived from the reconstructed images on an image display.

Abstract: The present disclosure relates to a method and variable quality playback system for selecting a quality of media content. The method comprising receiving (S4001) media content of a data segment (1010) over at least one network path (1031a, 1031b, 1031c), the media content being encoded with network or application-layer code and storing (S4002) the media content in a network or application-layer decoder (1050). The network or application-layer decoder (1050) is configured to decode the media content and provide decoded media content to a buffer (5061) associated with a media renderer (1060). The method further comprises obtaining a decoding metric of the network or application-layer decoder (1050), the decoding metric indicating a property of the decoding process and selecting the quality of the media content of subsequent data segments based on the decoding metric.

Abstract: Novel methods and systems for coordinating sound to both internal and external speakers for a device is disclosed. The audio signal is distributed among the internal and external speakers and aligned so that signals going to the internal speakers are aligned with signals going to the external speakers.

Abstract: Systems and methods for performing color mapping operations. One system includes a processor to perform post-production editing of image data. The processor is configured to identify a first region of an image and identify a second region of the image. The first region includes a first white point having a first tone, and the second region includes a second white point having a second tone. The processor is further configured to determine a color mapping function based on the first tone, apply the color mapping function to the second region of the image, and generate an output image.

Abstract: A method for decoding video includes receiving quantized coefficients representative of a block of video representative of a plurality of pixels. The quantized coefficients are dequantized based upon a function of a remainder. The dequantized coefficients are inverse transformed to determine a decoded residue.

Abstract: Some implementations involve receiving a content stream that includes audio data, receiving at least one type of level adjustment indication relating to playback of the audio data and controlling a level of the input audio data, based on the at least one type of level adjustment indication, to produce level-adjusted audio data. Some examples involve determining, based at least in part on the type(s) of level adjustment indication, a multiband limiter configuration, applying the multiband limiter to the level-adjusted audio data, to produce multiband limited audio data and providing the multiband limited audio data to one or more audio reproduction transducers of an audio environment.

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: The present document relates to audio coding systems. In particular, the present document relates to efficient methods and systems for parametric multi-channel audio coding. An audio encoding system configured to generate a bitstream indicative of a downmix signal and spatial metadata for generating a multi-channel upmix signal from the downmix signal is described. The system comprises a downmix processing unit configured to generate the downmix signal from a multi-channel input signal; wherein the downmix signal comprises m channels and wherein the multi-channel input signal comprises n channels; n, m being integers with m<n. Furthermore, the system comprises a parameter processing unit configured to determine the spatial metadata from the multi-channel input signal.

Abstract: A method for decoding an encoded audio signal is described. The encoded audio signal may comprise a sequence of frames and may be indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. Different subsets of DRC profiles may be comprised within different frames. The method may comprise determining a first rendering mode from the plurality of different rendering modes; determining one or more DRC profiles from a subset of DRC profiles comprised within a current frame; determining whether at least one of the DRC profiles is applicable to the first rendering mode; selecting a default DRC profile as a current DRC profile, if none of the DRC profiles is applicable to the first rendering mode; wherein definition data of the default DRC profile is known at a decoder; and decoding the current frame using the current DRC profile.

Abstract: The disclosure relates to a neural network based system for speech enhancement, comprising a generative network for generating an enhanced audio signal and a conditioning network for generating conditioning information for the generative network. The conditioning network comprises a plurality of layers and is configured to receive an audio signal as input: propagate the audio signal through the plurality of layers; and provide one or more first internal representations of the audio signal or processed versions thereof as the conditioning information, wherein the one or more first internal representations of the audio signal are extracted at respective layers of the conditioning network. The generative network is configured to receive a noise vector and the conditioning information as input; and generate the enhanced audio signal based on the noise vector and the conditioning information. The disclosure further relates to a method of training the system.

Abstract: A method and a device for encoding and decoding intra prediction are disclosed. An image decoding method for performing intra prediction comprises the steps of: receiving a bitstream including data on prediction modes of a current block and a block adjacent to the current block; extracting the data from the received bitstream so as to confirm the prediction mode of the adjacent block; determining whether a boundary pixel within the adjacent block can be used as a reference pixel for the current block in consideration of the prediction mode of the adjacent block; obtaining the reference pixel of the current block according to the determined result; generating a prediction block predicted in the frame on the basis of the obtained reference pixel; and decoding the current block by using the generated prediction block.

Abstract: Methods and systems for generating and using dynamic spatial metadata in image and video processing are described. In an encoder, in addition to global metadata, local, spatial metadata for two or more image regions or image objects are generated, smoothed, and embedded as spatial metadata values. In a decoder, the decoder can reconstruct the spatial metadata and use interpolation techniques to generate metadata for specific regions of interest. Examples of generating spatial metadata related to min, mid, and max luminance values in an image are provided.

Abstract: Camera tracking data with respect to a camera operating in a 3D physical space is received. An image portion depicting one or more visual objects not physically present in the 3D physical space is generated using a camera perspective derived from the camera tracking data. The one or more visual objects is caused to be visually combined with the camera perspective into a personal image taken by the camera.

Abstract: The present disclosure relates to the field of audio enhancement, and in particular to methods, devices and software for supervised training of a machine learning model, MLM, the MLM trained to enhance a degraded audio signal by calculating gains to be applied to frequency bands of the degraded audio signal. The present disclosure further relates to methods, devices and software for use of such a trained MLM.

Abstract: There is provided encoding and decoding methods for representing spatial audio that is a combination of directional sound and diffuse sound. An exemplary encoding method includes inter alia creating a single- or multi-channel downmix audio signal by downmixing input audio signals from a plurality of microphones in an audio capture unit capturing the spatial audio; determining first metadata parameters associated with the downmix audio signal, wherein the first metadata parameters are indicative of one or more of: a relative time delay value, a gain value, and a phase value associated with each input audio signal; and combining the created downmix audio signal and the first metadata parameters into a representation of the spatial audio.

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

Abstract: Some implementations involve receiving a content stream that includes audio data, receiving at least one type of level adjustment indication relating to playback of the audio data and controlling a level of the input audio data, based on the at least one type of level adjustment indication, to produce level-adjusted audio data. Some examples involve determining, based at least in part on the type(s) of level adjustment indication, a multiband limiter configuration, applying the multiband limiter to the level-adjusted audio data, to produce multiband limited audio data and providing the multiband limited audio data to one or more audio reproduction transducers of an audio environment.

Abstract: A display for displaying image data includes defining virtual color gamuts based on a plurality of primary display colors associated with a light source. At least one of the virtual color gamuts is defined to approximate an established color gamut. Intensity values associated with the virtual color gamuts are generated based on received video data, and the intensity values associated with the virtual color gamuts are used to generate drive values for the primary colors of the light source. A display using one or more virtual color gamuts is also disclosed.

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

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

Abstract: A method, for generating (a) a forward reshaping function for compressing an input high-dynamic range (HDR) image into a reshaped standard-dynamic-range (SDR) image and (b) a backward reshaping function for decompressing the reshaped SDR image into a reconstructed HDR image, includes (i) optimizing the forward reshaping function to minimize a deviation between the reshaped SDR image and an input SDR image corresponding to the input HDR image, (ii) optimizing the backward reshaping function to minimize a deviation between the reconstructed HDR image and the input HDR image, and (iii) until a termination condition is met, applying a correction to the input SDR image and reiterating, based on the input SDR image as corrected, the steps of optimizing the forward and backward reshaping functions.

Abstract: A method of controlling headphones having external microphone signal pass-through functionality may involve controlling a display to present a geometric shape on the display and receiving an indication of digit motion from a sensor system associated with the display. The sensor system may include a touch sensor system or a gesture sensor system. The indication may be an indication of a direction of digit motion relative to the display. The method may involve controlling the display to present a sequence of images indicating that the geometric shape either enlarges or contracts, depending on the direction of digit motion and changing a headphone transparency setting according to a current size of the geometric shape. The headphone transparency setting may correspond to an external microphone signal gain setting and/or a media signal gain setting of the headphones.

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

Abstract: In a method to improve backwards compatibility when decoding high-dynamic range images coded in a wide color gamut (WCG) space which may not be compatible with legacy color spaces, hue and/or saturation values of images in an image database are computed for both a legacy color space (say, YCbCr-gamma) and a preferred WCG color space (say, IPT-PQ). Based on a cost function, a reshaped color space is computed so that the distance between the hue values in the legacy color space and rotated hue values in the preferred color space is minimized. HDR images are coded in the reshaped color space. Legacy devices can still decode standard dynamic range images assuming they are coded in the legacy color space, while updated devices can use color reshaping information to decode HDR images in the preferred color space at full dynamic range.

Abstract: An input image to a pipeline of chained reshaping functions is received. Reference images are generated from the input image. The input image and the reference images are used to determine operational parameters for chained reshaping functions in the pipeline of chained reshaping functions. A reshaped image generated from one or more of the chained reshaping functions is encoded in a video signal along with image metadata. The image metadata includes some or all of the operational parameters specifying the chained reshaping functions. A recipient device of the video signal is caused to use the image metadata and the reshaped image to generate a reconstructed image.

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: 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: Input images are received as input to a multi-node system. The input images are divided into segments assigned to respective nodes of the multi-node system. Primary and secondary scenes are identified in the segments to ensure compliance with minimum and average distance constraints. Scene-level forward reshaping mappings are generated for the scenes by a respective node for an assigned segment. Forward reshaped images in the segment are generated by the node using the forward reshaping mappings and encoded into an output video signal, which enables a recipient device to generate reconstructed images and to render display images derived from the reconstructed images on an image display.

Abstract: Decoding of Ambisonics representations for a stereo loudspeaker setup is known for first-order Ambisonics audio signals. But such first-order Ambisonics approaches have either high negative side lobes or poor localisation in the frontal region. The invention deals with the processing for stereo decoders for higher-order Ambisonics HOA.

Abstract: A digital watermarking method including receiving, by an electronic processor, an original image signal containing a series of original visual images, where the original image signal encoded uses a perceptual quantizer (PQ) luminance level encoding transfer function resulting in PQ luminance steps within the original image signal, and where the PQ luminance steps have varying sizes across a luminance range. The method further includes receiving, by the electronic processor, a watermark image signal including a watermark, and adjusting the strength of the watermark by at least a first weighting factor that is a predetermined first number of PQ luminance steps.

Abstract: A method (200) of processing audio in an immersive audio format comprising at least one height audio channel, comprising: obtaining (250) two height audio signals from at least a portion of the at least one height audio channel: modifying (270) a relative phase between the two height audio signals in frequency bands in which phase differences are predominantly out of phase to obtain two phase modified height audio signals: and playing back (290) the processed audio comprising the two phase modified height audio signals with at least two audio loudspeakers. The phase differences occur as a result of having monaural signals emanated from two audio loudspeakers at one or more listening positions symmetrically off-center with respect to the at least two loudspeakers. laterally spaced with respect to each of said one or more listening positions. The method allows perception of sound height/elevation without using overhead speakers.

Abstract: Methods and systems for improving coding efficiency of video.

Abstract: Apparatus and methods for providing software and hardware based solutions to the problem of synthesizing noise for a digital image. According to one aspect, a probability image is generated and noise blocks are randomly placed at locations in the probability image where the locations have probability values that are compared to a threshold criterion, creating a synthesized noise image. Embodiments include generating synthesized film grain images and synthesized digital camera noise images.