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: 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: A novel high efficiency image projection system includes a beam-steering modulator, an amplitude modulator, and a controller. In a particular embodiment the controller generates beam-steering drive values from image data and uses the beam-steering drive values to drive the beam-steering modulator. Additionally, the controller utilizes the beam-steering drive values to generate a lightfield simulation of a lightfield projected onto the amplitude modulator by the beam-steering modulator. The controller utilizes the lightfield simulation to generate amplitude drive values for driving the amplitude modulator in order to project a high quality version of the image described by the image data.

Abstract: Embodiments are directed to a method of rendering adaptive audio by receiving input audio comprising channel-based audio, audio objects, and dynamic objects, wherein the dynamic objects are classified as sets of low-priority dynamic objects and high-priority dynamic objects, rendering the channel-based audio, the audio objects, and the low-priority dynamic objects in a first rendering processor of an audio processing system, and rendering the high-priority dynamic objects in a second rendering processor of the audio processing system. The rendered audio is then subject to virtualization and post-processing steps for playback through soundbars and other similar limited height capable speakers.

Abstract: Methods for echo estimation or echo management (echo suppression or cancellation) on an input audio signal, with at least one of adaptation of a sparse prediction filter set, modification (for example, truncation) of adapted prediction filter impulse responses, generation of a composite impulse response from adapted prediction filter impulse responses, or use of echo estimation and/or echo management resources in a manner determined at least in part by classification of the input audio signal as being (or not being) echo free. Other aspects are systems configured to perform any embodiment of any of the methods.

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: The present document describes a method (500) for providing combined audio and video content from a source device (210) to an audio sink device (230) and to a video sink device (220). The method (500) comprises determining (501) latency information regarding the video latency for processing the video content and the audio latency for processing the audio content of the combined audio and video content along the device chain (100) between the source device (210) and the audio and video sink devices (220, 230). Furthermore, the method (5009 comprises delaying (502) the audio content or the video content in dependence of the latency information, and providing (503) the combined audio and video content with the delayed audio content or the delayed video content.

Abstract: A system is configured to receive player multimedia having live coverage of physical expressions or actions of a player in an online activity and activity multimedia of live participation in the online activity associated with one or more play accounts. In a near continuous fashion, the system is configured to select in real time, for each time point, one or more player items being portions of the player multimedia or one or more activity items being portions of the activity multimedia corresponding to the time point to form a composite item. The system is configured to further transmit in real time the composite item to one or more viewer accounts. The system is configured to then receive viewer data in response and produce future composite items based on the viewer data.

Abstract: Based on viewing tracking data, a viewer's view direction to a three-dimensional (3D) scene depicted by a first video image is determined. The first video image has been streamed in a video stream to the streaming client device before the first time point and rendered with the streaming client device to the viewer at the first time point. Based on the viewer's view direction, a target view portion is identified in a second video image to be streamed in the video stream to the streaming client device to be rendered at a second time point subsequent to the first time point. The target view portion is encoded into the video stream with a higher target spatiotemporal resolution than that used to encode remaining non-target view portions in the second video image.

Abstract: In some embodiments, an encoder device is disclosed to receive an input video stream containing images in a first dynamic range including a first image. The device receives a second image representing the first image. The device obtains statistical data for the first and the second images. The device determines, at a first time delay, a scene cut data from the input video stream and storing the scene cut data in a first sliding window. The device determines, at a second time delay, a first smoothing mapping function based on a second sliding window and the determined scene cut data. The device determines, at a third time delay, a second smoothing mapping function based on a third sliding window and the determined scene cut data. The device generates, at the third time delay, a composer metadata for the first image based on the first and second smoothing mapping functions.

Abstract: Example embodiments disclosed herein relate to orientation-aware surround sound playback. A method for processing audio on an electronic device that includes a plurality of loudspeakers is disclosed, the loudspeakers arranged in more than one dimension of the electronic device. The method includes, responsive to receipt of a plurality of received audio streams, generating a rendering component associated with the plurality of received audio streams, determining an orientation dependent component of the rendering component, processing the rendering component by updating the orientation dependent component according to an orientation of the loudspeakers and dispatching the received audio streams to the plurality of loudspeakers for playback based on the processed rendering component. Corresponding system and computer program products are also disclosed.

Abstract: In one embodiment, a dual modulator display systems and methods for rendering target image data upon the dual modulator display system are disclosed where the display system receives target image data, possible HDR image data and first calculates display control signals and then calculates backlight control signals from the display control signals. This order of calculating display signals and then backlight control signals later as a function of the display systems may tend to reduce clipping artifacts. In other embodiments, it is possible to split the input target HDR image data into a base layer and a detail layer, wherein the base layer is the low spatial resolution image data that may be utilized as for backlight illumination data. The detail layer is higher spatial resolution image data that may be utilized for display control data.

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 performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.

Abstract: A method for rendering an audio output based on an audio data stream including M audio signals, side information including a series of reconstruction instances of a reconstruction matrix C and first timing data, the side information allowing reconstruction of N audio objects from the M audio signals, and object metadata defining spatial relationships between the N audio objects. The method includes generating a synchronized rendering matrix based on the object metadata, the first timing data, and information relating to a current playback system configuration, the synchronized rendering matrix having a rendering instance for each reconstruction instance, multiplying each reconstruction instance with a corresponding rendering instance to form a corresponding instance of an integrated rendering matrix, and applying the integrated rendering matrix to the audio signals in order to render an audio output.

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: Computer-implemented methods for training a neural network, as well as for implementing audio encoders and decoders via trained neural networks, are provided. The neural network may receive an input audio signal, generate an encoded audio signal and decode the encoded audio signal. A loss function generating module may receive the decoded audio signal and a ground truth audio signal, and may generate a loss function value corresponding to the decoded audio signal. Generating the loss function value may involve applying a psychoacoustic model. The neural network may be trained based on the loss function value. The training may involve updating at least one weight of the neural network.

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. The high frequency regeneration is performed as a post-processing operation with a delay of 3010 samples per audio channel.

Abstract: The present document relates to a method of layered encoding of a compressed sound representation of a sound or sound field. The compressed sound representation comprises a basic compressed sound representation comprising a plurality of components, basic side information for decoding the basic compressed sound representation to a basic reconstructed sound representation of the sound or sound field, and enhancement side information including parameters for improving the basic reconstructed sound representation.

Abstract: To support lossless mode at the block level when in-loop reshaping (LMCS) is enabled, the following changes are proposed to the existing LMCS pipeline. In intra mode, encode lossless blocks in the original domain, thus bypassing inverse mapping after reconstruction in the decoder. In inter mode, encode lossless blocks in the original domain, thus bypassing both forward mapping after motion compensation and inverse mapping after reconstruction in the decoder. In both modes, disable any LMCS-related color scaling.

Abstract: A projector system comprising a laser light source, a collimating lens, a fly-eye lens, an integrating rod and a first modulator is disclosed. The light from a laser light source/fiber illuminates a collimator to substantially collimate the light and then is transmitted through a fly's-eye lens. The fly's-eye lens provides a desired angular/spatial light distribution for further processing to a first modulator of the projector system.

Abstract: A method for generating metadata for use by a video decoder for displaying video content encoded by a video encoder includes: (1) accessing a target tone mapping curve; (2) accessing a decoder tone curve corresponding to a tone curve used by the video decoder for tone mapping the video content; (3) generating a plurality of parameters of a trim-pass function used by the video decoder to apply after applying the decoder tone curve to the video content, wherein the parameters of the trim-pass function are generated to approximate the target tone curve with the combination of the trim-pass function and the decoder tone curve, and (4) generating the metadata for use by the video decoder, including said plurality of parameters of the trim-pass function.

Abstract: Methods, processes, and systems are presented for inter-layer scaling and mapping to perform dynamic-range scalability in video coding using single-loop video decoders. Two approaches are proposed: a) A mapping function is applied to each of the color components, b) a mapping function is applied to the luma component and luma-based scaling is applied to the chroma components. Both schemes may be combined with existing schemes for in-loop reshaping, such as luma mapping and chroma residual scaling employed in the versatile video codec (VVC).

Abstract: A method (600) for decoding an encoded audio signal (102) is described. The encoded audio signal (102) comprises a sequence of frames. Furthermore, the encoded audio signal (102) is indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. Different subsets of DRC profiles from the plurality of DRC profiles are comprised within different frames of the sequence of frames, such that two or more frames of the sequence of frames jointly comprise the plurality of DRC profiles.

Abstract: Systems, methods, and computer program products of audio processing based on Adaptive Intermediate Spatial Format (AISF) are described. The AISF is an extension to ISF that allows spatial resolution around an ISF ring to be adjusted dynamically with respect to content of incoming audio objects. An AISF encoder device adaptively warps each ISF ring during ISF encoding to adjust angular distance between objects, resulting in increase in uniformity of energy distribution around the ISF ring. At an AISF decoder device, matrices that decode sound positions to the output speaker take into account the warping that was performed at the AISF encoder device to reproduce the true positions of sound sources.

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

Abstract: An apparatus and method of blind detection of binauralized audio. If the input content is detected as binaural, a second binauralization may be avoided. In this manner, the user experience avoids audio artifacts introduced by multiple binauralizations.

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

Abstract: A variable-resolution graphic equalizer providing an improved interface for controlling gain values across the entire audio spectrum using many narrow-band filters (e.g., 120). It allows user selection of a frequency range for graphic equalization and automatically maps a reduced and fixed number of sliders to the selected range based on the number of filter bands falling within the selected range. In an audio processing system, specific user interface regions are highlighted to display selected frequency ranges and corresponding selected sliders to allow for rapid and precise equalization of the full audio spectrum using the many narrow-band filters.

Abstract: Dual and multi-modulator projector display systems and techniques are disclosed. In one embodiment, a projector display system comprises a light source; a controller, a first modulator, receiving light from the light source and rendering a halftone image of said the input image; a blurring optical system that blurs said halftone image with a Point Spread Function (PSF); and a second modulator receiving the blurred halftone image and rendering a pulse width modulated image which may be projected to form the desired screen image. Systems and techniques for forming a binary halftone image from input image, correcting for misalignment between the first and second modulators and calibrating the projector system—e.g. over time—for continuous image improvement are also disclosed.

Abstract: A coded video sequence is received in a bitstream with a set of content scan adaptive metadata. It is ascertained if the set of content scan adaptive metadata is received. The set of content scan adaptive metadata includes: a maximum content light level parameter; a maximum frame average light level parameter. The maximum content light level parameter and maximum frame average light level parameter are both dependent on a scan type of the frames of the coded video sequence, the scan type being at least one of a progressive frame type, complimentary field pair type, macroblock-adaptive frame-field frame type, and individual field picture type.

Abstract: In some embodiments, virtualization methods for generating a binaural signal in response to channels of a multi-channel audio signal, which apply a binaural room impulse response (BRIR) to each channel including by using at least one feedback delay network (FDN) to apply a common late reverberation to a downmix of the channels. In some embodiments, input signal channels are processed in a first processing path to apply to each channel a direct response and early reflection portion of a single-channel BRIR for the channel, and the downmix of the channels is processed in a second processing path including at least one FDN which applies the common late reverberation. Typically, the common late reverberation emulates collective macro attributes of late reverberation portions of at least some of the single-channel BRIRs. Other aspects are headphone virtualizers configured to perform any embodiment of the method.

Abstract: A method for generating loudspeaker signals associated with a target screen size is disclosed. The method includes receiving a bit stream containing encoded higher order ambisonics signals, the encoded higher order ambisonics signals describing a sound field associated with a production screen size. The method further includes decoding the encoded higher order ambisonics signals to obtain a first set of decoded higher order ambisonics signals representing dominant components of the sound field and a second set of decoded higher order ambisonics signals representing ambient components of the sound field. The method also includes combining the first set of decoded higher order ambisonics signals and the second set of decoded higher order ambisonics signals to produce a combined set of decoded higher order ambisonics signals.

Abstract: The present document describes a method (400) for encoding a soundfield representation (SR) input signal (101, 301) describing a soundfield at a reference position, wherein the SR input signal (101, 301) comprises a plurality of channels for a plurality of different directivity patterns of the soundfield at the reference position. The method (400) comprises extracting (401) one or more audio objects (103, 303) from the SR input signal (101, 301). Furthermore, the method (400) comprises determining (402) a residual signal (102, 302) based on the SR input signal (101, 301) and based on the one or more audio objects (103, 303). The method (400) also comprises performing joint coding of the one or more audio objects (103, 303) and/or the residual signal (102, 302). In addition, the method (400) comprises generating (403) a bitstream (701) based on data generated in the context of joint coding of the one or more audio objects (103, 303) and/or the residual signal (102, 302).

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media, for wireless audio source switching. One of the methods includes receiving user input selecting a user interface control corresponding to one of a plurality of paired audio source devices with a wireless audio sink device that is a slave device of a first master/slave connection with a first paired audio source device of the plurality of paired audio source devices. The wireless audio sink device initiates a request to become a master device of a second master/slave connection with a second paired audio source device corresponding to the selected user interface control. The wireless audio sink device sends to the second paired audio source device a request to become a slave device of the second master/slave connection. The wireless audio sink device receives audio input from the second paired audio source device and outputs the received audio input.

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: 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: 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: Embodiments are described for rendering spatial audio content through a system that is configured to reflect audio off of one or more surfaces of a listening environment. The system includes an array of audio drivers distributed around a room, wherein at least one driver of the array of drivers is configured to project sound waves toward one or more surfaces of the listening environment for reflection to a listening area within the listening environment and a renderer configured to receive and process audio streams and one or more metadata sets that are associated with each of the audio streams and that specify a playback location in the listening environment.

Abstract: A method for demosaicing an image captured by an image sensor includes (a) computing, for each of a first plurality of interpolated pixel-values and based on neighboring primary pixel-values, a respective first confidence value; (b) generating a first thresholded mapping from a first coordinate mapping by removing each interpolated pixel-value having a first confidence value less than a threshold value; (c) repeating steps of computing and generating for a second and third sub-plurality of sensor pixels to yield a second and a third thresholded mapping; (d) determining high-confidence array-coordinates as array-coordinates included in all thresholded mappings, remaining array-coordinates being low-confidence array-coordinates; (e) forming a refined image including, at each high-confidence array-coordinate, a respective pixel-value-triplet including one primary pixel-value and two interpolated pixel-values; and (f) filling the refined image by assigning, for each low-confidence array-coordinate, a pixel-value

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

Abstract: 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: Given a representation of a forward reshaping function in an M-piecewise linear representation (M-PWL), methods, processes, and systems are presented for generating the forward reshaping function in an N-PWL representation (where N<M), by properly adjusting the first and last segments, and for generating a decoded bitstream using the adjusted N-PWL reshaping representation. Syntax methods for signaling reshaping parameters are also presented.

Abstract: The present disclosure relates to an apparatus for decoding an encoded Unified Audio and Speech stream. The apparatus comprises a core decoder for decoding the encoded Unified Audio and Speech stream. The core decoder includes an eSBR unit for extending a bandwidth of an input signal, the eSBR unit including a QMF based harmonic transposer. The QMF based harmonic transposer is configured to process the input signal in the QMF domain, in each of a plurality of synthesis subbands, to extend the bandwidth of the input signal. The QMF based harmonic transposer is configured to operate at least in part based on pre-computed information. The present disclosure further relates to corresponding methods and storage media.

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 invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal.

Abstract: Some disclosed methods may involve receiving audio reproduction data and determining, based on the audio reproduction data, a sound source location at which a sound is to be rendered. A near-field gain and a far-field gain may be based, at least in part, on a sound source distance between the sound source location and a reproduction environment location. Room speaker feed signals may be based, at least in part, on room speaker positions, the sound source location and the far-field gain. Near-field speaker feed signals may be based, at least in part, on the near-field gain, the sound source location and a position of near-field speakers.

Abstract: Described herein is a method (30) of rendering an audio signal (17) for playback in an audio environment (27) defined by a target loudspeaker system (23), the audio signal (17) including audio data relating to an audio object and associated position data indicative of an object position. Method (30) includes the initial step (31) of receiving the audio signal (17). At step (32) loudspeaker layout data for the target loudspeaker system (23) is received. At step (33) control data is received that is indicative of a position modification to be applied to the audio object in the audio environment (27). At step (38) in response to the position data, loudspeaker layout data and control data, rendering modification data is generated. Finally, at step (39) the audio signal (17) is rendered with the rendering modification data to output the audio signal (17) with the audio object at a modified object position that is between loudspeakers within the audio environment (27).

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: Described herein is a method for processing of media data for playback, wherein the method includes the steps of: (a) fetching, by a web proxy, from two or more media servers media data and a media manifest file including metadata information relating to the fetched media data, and merging, by the web proxy, the media data; (b) modifying, by said web proxy, the content of the media manifest file and/or the content of the media data; and (c) providing, by said web proxy, the media manifest file and the media data as modified in step (b) to a media retrieval element for receiving and processing the media manifest file and the media data for decoding or playback, wherein a localhost address is assigned to the web proxy and the web proxy acts as a server for said media retrieval element. Described are further a respective device and computer program product.