Abstract: Disclosed is an audio signal encoding/decoding method that uses an encoding downmix strategy applied at an encoder that is different than a decoding re-mix/upmix strategy applied at a decoder. Based on the type of downmix coding scheme, the method comprises: computing input downmixing gains to be applied to the input audio signal to construct a primary downmix channel; determining downmix scaling gains to scale the primary downmix channel; generating prediction gains based on the input audio signal, the input downmixing gains and the downmix scaling gains; determining residual channel(s) from the side channels by using the primary downmix channel and the prediction gains to generate side channel predictions and subtracting the side channel predictions from the side channels; determining decorrelation gains based on energy in the residual channels; encoding the primary downmix channel, the residual channel(s), the prediction gains and the decorrelation gains; and sending the bitstream to a decoder.

Abstract: A method for modifying object reconstruction information, comprising obtaining a set of N spatial audio objects, each spatial audio object including an audio signal and spatial metadata, obtaining an audio presentation representing the N spatial audio objects, obtaining object reconstruction information configured to reconstruct the N spatial audio objects from the audio presentation, applying the reconstruction information to the audio presentation to form a set of N reconstructed spatial audio objects, using a first rendering configuration, rendering the N spatial audio objects to obtain a first rendered presentation, and rendering the N reconstructed spatial audio objects to obtain a second rendered presentation, and modifying the reconstruction information based on a difference between the first rendered presentation and the second rendered presentation, thereby forming modified reconstruction information.

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

Abstract: Described herein is a method for generating a modified bitstream on a source device, wherein the method includes the steps of: a) receiving, by a receiver, a bitstream including coded media data; b) generating, by an embedder, payload of additional media data and embedding the payload in the bitstream for obtaining, as an output from the embedder, a modified bitstream including the coded media data and the payload of the additional media data; and c) outputting the modified bitstream to a sink device. Described is further a method for processing said modified bitstream on a sink device. Described are moreover a respective source device and sink device as well as a system of a source device and a sink device and respective computer program products.

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

Abstract: Methods and systems for advanced stereo processing of an audio signal are disclosed. The methods and systems include selecting a coding mode of either transform coding or linear predictive coding and performing advanced stereo processing when in the selected coding mode. Both encoding and decoding operations are provided.

Abstract: Conventional audio compression technologies perform a standardized signal transformation, independent of the type of the content. Multi-channel signals are decomposed into their signal components, subsequently quantized and encoded. This is disadvantageous due to lack of knowledge on the characteristics of scene composition, especially for e.g. multi-channel audio or Higher-Order Ambisonics (HOA) content. A method for decoding an encoded bitstream of multi-channel audio data and associated metadata is provided, including transforming the first Ambisonics format of the multi-channel audio data to a second Ambisonics format representation of the multi-channel audio data, wherein the transforming maps the first Ambisonics format of the multi-channel audio data into the second Ambisonics format representation of the multi-channel audio data.

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: Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent.

Abstract: First foviated images are streamed to a streaming client. The first foviated images with first image metadata sets are used to generate first display mapped images for rendering to a viewer at first time points. View direction data is collected and used to determine a second view direction of the viewer at a second time point. A second foviated image and a second image metadata set are generated from a second HDR source image in reference to the second view direction of the viewer and used to generate a second display mapped image for rendering to the viewer at the second time point. The second image metadata set comprises a display management metadata portions for adapting a focal-vision and peripheral-vision image portions to corresponding image portions in the second display mapped image. The focal-vision display management metadata portion is generated with a predicted light adaptation level of the viewer for the second time point.

Abstract: A method for adaptive streaming of media content with bitrate switching is described, wherein the media content comprising a plurality of consecutive media segments. The method comprising, at a media streaming server: transmitting a segment of the media content encoded in a first coding mode having a first bitrate; receiving an indication for a coding mode switch to a second coding mode having a second bitrate and in response transmitting a transition segment for transitioning between the first coding mode and the second coding mode; and transmitting another segment of the media content encoded in the second coding mode.

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

Abstract: A set of tensor-product B-Spline (TPB) basis functions is determined. A set of selected TPB prediction parameters to be used with the set of TPB basis functions for generating predicted image data in mapped images from source image data in source images of a source color grade is generated. The set of selected TPB prediction parameters is generated by minimizing differences between the predicted image data in the mapped images and reference image data in reference images of a reference color grade. The reference images correspond to the source images and depict same visual content as depicted by the source images. The set of selected TPB prediction parameters is encoded in a video signal as a part of image metadata along with the source image data in the source images. The mapped images are caused to be reconstructed and rendered with a recipient device of the video signal.

Abstract: A method is provided for coding at least one image split up into partitions, a current partition to be coded containing data, at least one data item of which is allotted a sign. The coding method includes, for the current partition, the following steps: calculating the value of a function representative of the data of the current partition with the exclusion of the sign; comparing the calculated value with a predetermined value of the sign; as a function of the result of the comparison, modifying or not modifying at least one of the data items of the current partition, in the case of modification, coding the at least one modified data item.

Abstract: Methods and systems for canvas size scalability across the same or different bitstream layers of a video coded bitstream are described. Offset parameters for a conformance window, a reference region of interest (ROI) in a reference layer, and a current ROI in a current layer are received. The width and height of a current ROI and a reference ROI are computed based on the offset parameters and they are used to generate a width and height scaling factor to be used by a reference picture resampling unit to generate an output picture based on the current ROI and the reference ROI.

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: 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: Methods, systems, and devices implement intra-prediction for hexagonally-sampled compression and decompression of videos and images having a regular grid of hexagonally-shaped pixels. For encoding, a prediction unit (PU) shape is selected at a sequence level from the group consisting of parallelogram, zigzag-square, hexagonal super-pixel, a rectangular zigzag and an arrow, and the hexagonally-sampled image is divided into regions based on the PU shape. For each region: a prediction mode and a PU size are determined; reference pixels are determined for each predicted pixel in the PU shape based on the prediction mode; a weighted factor is determined for each of the reference pixels based on a distance between the reference pixel and the predicted pixel; and a predicted value of each of the predicted pixels in the PU shape is determined using the corresponding reference pixels and the weighted factors.

Abstract: Some examples involve rendering received audio data by determining a first relative activation of a set of loudspeakers in an environment according to a first rendering configuration corresponding to a first set of speaker activations, receiving a first rendering transition indication indicating a transition from the first rendering configuration to a second rendering configuration and determining a second set of speaker activations corresponding to a simplified version of the second rendering configuration. Some examples involve performing a first transition from the first set of speaker activations to the second set of speaker activations, determining a third set of speaker activations corresponding to a complete version of the second rendering configuration and performing a second transition to the third set of speaker activations without requiring completion of the first transition.

Abstract: Some methods involve receiving, by a control system that is configured for implementing a plurality of renderers, audio data and listening configuration data for a plurality of listening configurations, each listening configuration of the plurality of listening configurations corresponding to a listening position and a listening orientation in an audio environment, and rendering, by each renderer and according to the listening configuration data, the received audio data to obtain a set of renderer-specific loudspeaker feed signals for a corresponding listening configuration. Each renderer may be configured to render the audio data for a different listening configuration. Some such methods may involve decomposing each set of renderer-specific loudspeaker feed signals into a renderer-specific set of frequency bands and combining the renderer-specific frequency bands of each renderer to produce an output set of loudspeaker feed signals.

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: An multi-input, multi-output audio process is implemented as a linear system for use in an audio filterbank to convert a set of frequency-domain input audio signals into a set of frequency-domain output audio signals. A transfer function from one input to one output is defined as a frequency dependent gain function. In some implementations, the transfer function includes a direct component that is substantially defined as a frequency dependent gain, and one or more decorrelated components that have frequency-varying group phase response. The transfer function is formed from a set of sub-band functions, with each sub-band function being formed from a set of corresponding component transfer functions including direct component and one or more decorrelated components.

Abstract: A first image and a second image of different dynamic ranges are derived from the same source image. Based on a chroma sampling format of the first image, it is determined whether edge preserving filtering is to be used to generate chroma upsampled image data in a reconstructed image. If so, image metadata for performing the edge preserving filtering is generated. The first image, the second image and the image metadata are encoded into an image data container to enable a recipient device to generate the reconstructed image.

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 loudspeaker equalization. The method combines default tunings (generated based on a default listening environment) and room tunings (generated based on an end user listening environment) to result in combined tunings that account for differences between the end user listening environment and the default listening environment.

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: Soundfield signals such as e.g. Ambisonics carry a representation of a desired sound field. The Ambisonics format is based on spherical harmonic decomposition of the soundfield, and Higher Order Ambisonics (HOA) uses spherical harmonics of at least 2nd order. However, commonly used loudspeaker setups are irregular and lead to problems in decoder design. A method for improved decoding an audio soundfield representation for audio playback comprises calculating a panning function (W) using a geometrical method based on the positions of a plurality of loudspeakers and a plurality of source directions, calculating a mode matrix (?) from the loudspeaker positions, calculating a pseudo-inverse mode matrix (?+) and decoding the audio soundfield representation. The decoding is based on a decode matrix (D) that is obtained from the panning function (W) and the pseudo-inverse mode matrix (?+).

Abstract: A method for generating a bitstream indicative of an object based audio program is described. The bitstream comprises a sequence of containers. A first container of the sequence of containers comprises a plurality of substream entities for a plurality of substreams of the object based audio program and a presentation section. The method comprises determining a set of object channels. The method further comprises providing a set of object related metadata for the set of object channels. In addition, the method comprises inserting a first set of object channel frames and a first set of object related metadata frames into a respective set of substream entities of the first container. Furthermore, the method comprises inserting presentation data into the presentation section.

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: 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: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: Described is picture segmentation through columns and slices in video encoding and decoding. A video picture is divided into a plurality of columns, each column covering only a part of the video picture in a horizontal dimension. All coded tree blocks (“CTBs”) belonging to a slice may belong to one or more columns. The columns may be used to break the same or different prediction or in-loop filtering mechanisms of the video coding, and the CTB scan order used for encoding and/or decoding may be local to a column. Column widths may be indicated in a parameter set and/or may be adjusted at the slice level. At the decoder, column width may be parsed from the bitstream, and slice decoding may occur in one or more columns.

Abstract: An attenuation or “gap” may be inserted into at least a first frequency range of at least first and second audio playback signals of a content stream during at least a first time interval to generate at least first and second modified audio playback signals. Corresponding audio device playback sound may be provided by at least first and second audio devices. At least one microphone may detect at least the first audio device playback sound and the second audio device playback sound and may generate corresponding microphone signals. Audio data may be extracted from the microphone signals in at least the first frequency range, to produce extracted audio data. A far-field audio environment impulse response and/or audio environment noise may be estimated based, at least in part, on the extracted audio data.

Abstract: Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.

Abstract: The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length La, extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length Ls, generating a frame of the output signal.

Abstract: A method for encoding an input audio stream including the steps of obtaining a first playback stream presentation of the input audio stream intended for reproduction on a first audio reproduction system, obtaining a second playback stream presentation of the input audio stream intended for reproduction on a second audio reproduction system, determining a set of transform parameters suitable for transforming an intermediate playback stream presentation to an approximation of the second playback stream presentation, wherein the transform parameters are determined by minimization of a measure of a difference between the approximation of the second playback stream presentation and the second playback stream presentation, and encoding the first playback stream presentation and the set of transform parameters for transmission to a decoder.

Abstract: Embodiments are disclosed for spatial noise filling in multi-channel codecs. In an embodiment, a method of regenerating background noise ambience in a multi-channel codec by generating spatial hole filling noise comprises: computing noise estimates based on a primary downmix channel generated from an input audio signal representing a spatial audio scene with background noise ambience; computing spectral shaping filter coefficients based on the noise estimates; spectrally shaping the multi-channel noise signal using the spectral shaping filter coefficients and a noise distribution, the spectral shaping resulting in a diffused, multi-channel noise signal with uncorrelated channels; spatially shaping the diffused, uncorrelated multi-channel noise signal with uncorrelated channels based on a noise ambience of the spatial audio scene; and adding the spatially and spectrally shaped multi-channel noise to a multi-channel codec output to synthesize the background noise ambience of the spatial audio scene.

Abstract: Some methods involve receiving, by a control system configured for implementing a plurality of Tenderers, audio data and listening configuration data for a plurality of listening configurations, each listening configuration of the plurality of listening configurations corresponding to a listening position and a listening orientation in an audio environment, and rendering, by each Tenderer and according to the listening configuration data, the received audio data to obtain a set of Tenderer-specific loudspeaker feed signals for a corresponding listening configuration. Each Tenderer may be configured to render the audio data for a different listening configuration. Some such methods may involve decomposing each set of renderer-specific loudspeaker feed signals into a Tenderer-specific set of frequency bands and combining the renderer-specific frequency bands of each Tenderer to produce an output set of loudspeaker feed signals.

Abstract: A method for tracking a reference picture on an electronic device is described. The method includes receiving a bitstream. The method also includes decoding a portion of the bitstream to produce a decoded reference picture. The method further includes tracking the decoded reference picture in a decoded picture buffer (DPB) with reduced overhead referencing. The method additionally includes decoding a picture based on the decoded reference picture.

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: An apparatus and method of rendering audio objects with multiple types of renderers. The weighting between the selected renderers depends upon the position information in each audio object. As each type of renderer has a different output coverage, the combination of their weighted outputs results in the audio being perceived at the position according to the position information.

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: A method for generating a high-dynamic-range (HDR) image includes (a) denoising a short-exposure-time image, wherein the denoising comprises applying a first guided filter to the short-exposure-time image, the guided filter utilizing a long exposure-time-image as its guide, (b) after the step of denoising, scaling at least one of the short-exposure-time image and the long-exposure-time image to place the short-exposure-time image and the long-exposure-time image on a common radiance scale, and (c) after the step of scaling, merging the short-exposure-time image with the long-exposure-time image to generate the HDR image.

Abstract: The disclosure relates to methods of processing a spatial audio signal for generating a compressed representation of the spatial audio signal. The methods include analyzing the spatial audio signal to determine directions of arrival for one or more audio elements; for at least one frequency subband, determining respective indications of signal power associated with the directions of arrival; generating metadata including direction information that includes indications of the directions of arrival of the audio elements, and energy information that includes respective indications of signal power; generating a channel-based audio signal with a predefined number of channels based on the spatial audio signal; and outputting, as the compressed representation, the channel-based audio signal and the metadata.

Abstract: A method for performing DRC on a HOA signal comprises transforming the HOA signal to the spatial domain, analyzing the transformed HOA signal, and obtaining, from results of said analyzing, gain factors that are usable for dynamic compression. The gain factors can be transmitted together with the HOA signal. When applying the DRC, the HOA signal is transformed to the spatial domain, the gain factors are extracted and multiplied with the transformed HOA signal in the spatial domain, wherein a gain compensated transformed HOA signal is obtained. The gain compensated transformed HOA signal is transformed back into the HOA domain, wherein a gain compensated HOA signal is obtained. The DRC may be applied in the QMF-filter bank domain.

Abstract: A first reshaping mapping is performed on a first image represented in a first domain to generate a second image represented in a second domain. The first domain is of a first dynamic range different from a second dynamic range of which the second domain is. A second reshaping mapping is performed on the second image represented in the second domain to generate a third image represented in the first domain. The third image is perceptually different from the first image in at least one of: global contrast, global saturation, local contrast, local saturation, etc. A display image is derived from the third image and rendered on a display device.

Abstract: Images are acquired through image sensors operating in conjunction with a media consumption system. The acquired images are used to determine a user's movement in a plurality of degrees of freedom. Sound images depicted in spatial audio rendered by audio speakers operating in conjunction with the media consumption system are adapted based at least in part on the user's movement in the plurality of degrees of freedom.

Abstract: The present invention is directed to methods and apparatus for translating a first plurality of audio input channels to a second plurality of audio output channels. This includes determining that there is pair-wise coding among any of the first plurality of audio input channels, determining an input/output-mapping matrix for mapping at least a first set of the first plurality of audio input channels to at least a second set of the second plurality of audio output channels; and deriving the second plurality of audio output channels based on first plurality of audio input channels, the input/output-mapping matrix and the determined pair-wise coding. The first plurality of audio input channels represent the same soundfield represented by the second plurality of audio output channels.

Abstract: The present invention relates to a block partitioning structure in video coding technology, and a video encoding and decoding method and apparatus using the same, wherein the video encoding and decoding method includes the steps of: acquiring quad-partitioning information of a block; acquiring bi-partitioning information of the block when the acquired quad-partitioning information of the block does not indicate four partitions; acquiring partitioning direction information for bi-partitioning of the block when the acquired bi-partitioning information of the block indicates two partitions; acquiring information on whether to perform any other type of partitioning, when the acquired bi-partitioning information of the block does not indicate two partitions; and acquiring additional information required for the any other type of partitioning, when the acquired information on whether to perform any other type of partitioning indicates that the any other type of partitioning is performed.