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.

Abstract: Systems and methods to transmit data over multiple communication channels in parallel with forward error correction. Original packets are evenly distributed to the channels as the initial systematically channel-encoded packets. Subsequent channel-encoded packets are configured to be linearly independent of their base sets of channel-encoded packets, where a base set for a subsequent channel-encoded packet includes those scheduled to be transmitted before the subsequent packet in the same channel as the subsequent packet, and optionally one or more initial packets from other channels. The compositions of the sequences of the encoded packets can be predetermined without the content of the packets; and the channel-encoded packets can be generated from the original packets on-the-fly by the transmitters of the channels during transmission. When a sufficient number of packets have been received via the channels, a recipient may terminate their transmissions.

Abstract: An audio processing method may involve receiving media input audio data corresponding to a media stream and headphone microphone input audio data, determining a media audio gain for at least one of a plurality of frequency bands of the media input audio data and determining a headphone microphone audio gain for at least one of a plurality of frequency bands of the headphone microphone input audio data. Determining the headphone microphone audio gain may involve determining a feedback risk control value, for at least one of the plurality of frequency bands, corresponding to a risk of headphone feedback between at least one external microphone of a headphone microphone system and at least one headphone speaker and determining a headphone microphone audio gain that will mitigate actual or potential headphone feedback in at least one of the plurality of frequency bands, based at least partly upon the feedback risk control value.

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: 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: Encoding/decoding techniques where multiple transform parameter sets are encoded together with a rendered playback presentation of an input audio content. The multiple transform parameters are used on the decoder side to transform the playback presentation to provide a personalized binaural playback presentation optimized for an individual listener with respect to their hearing profile. This may be achieved by selection or combination of the data present in the metadata streams.

Abstract: An apparatus and method of transmission control for an audio device. The audio device uses sources other than the microphone to determine nuisance, and uses this to calculate a gain as well as to make the transmit decision. Using the gain results in a more nuanced nuisance mitigation than using the transmit decision on its own.

Abstract: One or more context aware processing parameters and an ambient audio stream are received. One or more sound characteristics associated with the ambient audio stream are identified using a machine learning model. One or more actions to perform are determined using the machine learning model and based on the one or more context aware processing parameters and the identified one or more sound characteristics. The one or more actions are performed.

Abstract: Apparatus and methods are provided that employ one or more of a variety of techniques for reducing the time required to display high resolution images on a high dynamic range display having a light source layer and a display layer. In one technique, the image resolution is reduced, an effective luminance pattern is determined for the reduced resolution image, and the resolution of the effective luminance pattern is then increased to the resolution of the display layer. In another technique, the light source layer's point spread function is decomposed into a plurality of components, and an effective luminance pattern is determined for each component. The effective luminance patterns are then combined to produce a total effective luminance pattern. Additional image display time reduction techniques are provided.

Abstract: An apparatus may include a housing adapted for at least partial insertion into a concha bowl of a human ear, at least one speaker residing in or on the housing, a control system residing in or on the housing and a positioning element attached to the housing. The control system may be configured for controlling the speaker and configured for radio frequency (RF) communication. The positioning element may be configured to fit at least partially inside a concha of the human ear and may be configured to retain the housing at least partially within the concha bowl. The positioning element may include one or more wires Control System configured for communication with the control system. The one or more wires may be configured for at receiving and/or transmitting RF radiation. In some examples, the positioning element may be, or may include, a concha lock. The positioning element may include a loop antenna.

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: An integrated cinema amplifier comprises a power supply stage that distributes power over a plurality of channels for rendering immersive audio content in a surround sound listening environment. The amplifier automatically detects maximum and net power availability and requirements based on audio content by decoding audio metadata and dynamically adjusts gains to each channel or sets of channels based on content and operational/environmental conditions. A power supply stage provides power to drive a plurality of channels corresponding to speaker feeds to a plurality of speakers. The amplifier has a front panel having an LED array with each LED associated with a respective channel or group of channels of the multi-channel amplifier, and a control unit configured to light the LEDs according to display patterns based on operating status or error conditions of the amplifier.

Abstract: Individual loudspeaker dynamics processing configuration data, for each of a plurality of loudspeakers of a listening environment, may be obtained. Listening environment dynamics processing configuration data may be determined, based on the individual loudspeaker dynamics processing configuration data. Dynamics processing may be performed on received audio data based on the listening environment dynamics processing configuration data, to generate processed audio data. The processed audio data may be rendered for reproduction via a set of loudspeakers that includes at least some of the plurality of loudspeakers, to produce rendered audio signals. The rendered audio signals may be provided to, and reproduced by, the set of loudspeakers.

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

Abstract: An apparatus for processing an input audio signal relies on a cascade of filterbanks, the cascade having a synthesis filterbank for synthesizing an audio intermediate signal from the input audio signal, the input audio signal being represented by a plurality of first subband signals generated by an analysis filterbank, wherein a number of filterbank channels of the synthesis filterbank is smaller than a number of channels of the analysis filterbank. The apparatus furthermore has a further analysis filterbank for generating a plurality of second subband signals from the audio intermediate signal, wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank, so that a sampling rate of a subband signal of the plurality of second subband signals is different from a sampling rate of a first subband signal of the plurality of first subband signals.

Abstract: An integrated cinema amplifier comprises a power supply stage that distributes power over a plurality of channels for rendering immersive audio content in a surround sound listening environment. The amplifier automatically detects maximum and net power availability and requirements based on audio content by decoding audio metadata and dynamically adjusts gains to each channel or sets of channels based on content and operational/environmental conditions. A power supply stage provides power to drive a plurality of channels corresponding to speaker feeds to a plurality of speakers. The amplifier has a front panel having an LED array with each LED associated with a respective channel or group of channels of the multi-channel amplifier, and a control unit configured to light the LEDs according to display patterns based on operating status or error conditions of the amplifier.

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 invention proposes a method and a device for arithmetic encoding of a current spectral coefficient using preceding spectral coefficients. Said preceding spectral coefficients are already encoded and both, said preceding and current spectral coefficients, are comprised in one or more quantized spectra resulting from quantizing time-frequency-transform of video, audio or speech signal sample values.

Abstract: A method for selecting a device for audio processing may involve receiving a first wakeword confidence metric from a first device that includes at least a first microphone and receiving a second wakeword confidence metric from a second device that includes at least a second microphone. The first and second wakeword confidence metrics may correspond to a first local maximum of a first plurality of wakeword confidence values determined by the first device and a second local maximum of a second plurality of wakeword confidence values determined by the second device. The method may involve comparing the first wakeword confidence metric and the second wakeword confidence metric and selecting a device for subsequent audio processing based, at least in part, on a comparison of the first wakeword confidence metric and the second wakeword confidence metric.

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: 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 sequence of images in a first codeword representation, methods, processes, and systems are presented for integrating reshaping into a next generation video codec for encoding and decoding the images, wherein reshaping allows part of the images to be coded in a second codeword representation which allows more efficient compression than using the first codeword representation. A variety of architectures are discussed, including: an out-of-loop reshaping architecture, an in-loop-for intra pictures only reshaping architecture, an in-loop architecture for prediction residuals, and a hybrid in-loop reshaping architecture. Syntax methods for signaling reshaping parameters, and image-encoding methods optimized with respect to reshaping are also presented.

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: A method of processing of a sequence of video frames from a camera capturing a writing surface for subsequent transmission to at least one of a remote videoconferencing client and a remote videoconferencing server. The method comprises receiving the sequence of video frames from the camera; and selecting an image area of interest in the video frames, comprising selecting one of a sub-area of the video frames and an entire area of the video frames. The method also comprises, for each current video frame of the sequence of video frames, generating a pen stroke mask by applying adaptive thresholding to the image area of interest. The method also comprises generating an output video frame using the pen stroke mask. Corresponding systems and computer readable media are disclosed.

Abstract: Shaped glasses have curved surface lenses with spectrally complementary filters disposed thereon. The filters curved surface lenses are configured to compensate for wavelength shifts occurring due to viewing angles and other sources. Complementary images are projected for viewing through projection filters having passbands that pre-shift to compensate for subsequent wavelength shifts. At least one filter may have more than 3 primary passbands. For example, two filters include a first filter having passbands of low blue, high blue, low green, high green, and red, and a second filter having passbands of blue, green, and red. The additional passbands may be utilized to more closely match a color space and white point of a projector in which the filters are used. The shaped glasses and projection filters together may be utilized as a system for projecting and viewing 3D images.

Abstract: An audio session management method for an audio environment having multiple audio devices may involve receiving, from a first device implementing a first application and by a device implementing an audio session manager, a first route initiation request to initiate a first route for a first audio session. The first route initiation request may indicate a first audio source and a first audio environment destination. The first audio environment destination may correspond with at least a first person in the audio environment, but in some instances will not indicate an audio device. The method may involve establishing a first route corresponding to the first route initiation request. Establishing the first route may involve determining a first location of at least the first person in the audio environment, determining at least one audio device for a first stage of the first audio session and initiating or scheduling the first audio session.

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: Embodiments are disclosed for hybrid near/far-field speaker virtualization.

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 upmixing unit adapted to perform mono to stereo upmixing. The upmixing unit includes a decorrelator unit D adapted to apply a decorrelation filter to an input signal. The decorrelator unit is adapted to determine filter coefficients for the decorrelation filter by referring to pre-computed values. The present disclosure further relates to a an apparatus for encoding a Unified Audio and Speech stream, as well as to corresponding methods and storage media.

Abstract: Novel methods and systems for adapting a voice cloning synthesizer for a new speaker using real speech data are disclosed. Utterances from one or more target speakers are parameterized and are used to initialize an embedding vector for use with a voice synthesizer, by means of clustering the utterance data and determining the centroid of the data, using a speaker identification neural network, and/or by finding the closest stored embedded vector to the utterance data.

Abstract: A rendering mode may be determined for received audio data, including audio signals and associated spatial data. The audio data may be rendered for reproduction via a set of loudspeakers of an environment according to the rendering mode, to produce rendered audio signals. Rendering the audio data may involve determining relative activation of a set of loudspeakers in an environment. The rendering mode may be variable between a reference spatial mode and one or more distributed spatial modes. The reference spatial mode may have an assumed listening position and orientation. In the distributed spatial mode(s), one or more elements of the audio data may each be rendered in a more spatially distributed manner than in the reference spatial mode and spatial locations of remaining elements of the audio data may be warped such that they span a rendering space of the environment more completely than in the reference spatial mode.

Abstract: A method for estimating a user's location in an environment may involve receiving output signals from each microphone of a plurality of microphones in the environment. At least two microphones of the plurality of microphones may be included in separate devices at separate locations in the environment and the output signals may correspond to a current utterance of a user. The method may involve determining multiple current acoustic features from the output signals of each microphone and applying a classifier to the multiple current acoustic features. Applying the classifier may involve applying a model trained on previously-determined acoustic features derived from a plurality of previous utterances made by the user in a plurality of user zones in the environment. The method may involve determining, based at least in part on output from the classifier, an estimate of the user zone in which the user is currently located.

Abstract: Encoding and decoding devices for encoding the channels of an audio system having at least four channels are disclosed. The decoding device has a first stereo decoding component which subjects a first pair of input channels to a first stereo decoding, and a second stereo decoding component which subjects a second pair of input channels to a second stereo decoding. The results of the first and second stereo decoding components are crosswise coupled to a third and a fourth stereo decoding component which each performs stereo decoding on one channel resulting from the first stereo decoding component, and one channel resulting from the second stereo decoding component.

Abstract: A method is provided, including: defining a plurality of frequency bins; sending, during a training phase, a test signal at different amplitude levels to one or more speakers, and gathering resulting test voltage (V) and current (I) points for the different amplitude levels and for each frequency bin; for each frequency bin, applying a linear regression algorithm to the gathered test voltage and current points for the different amplitudes to obtain a reference electrical impedance of said one or more speakers; sending, during a monitoring phase subsequent to said training phase, an audio signal to said one or more speakers, and gathering resulting new voltage and current points to obtain an operating electrical impedance for said one or more speakers for each frequency bin, determining a deviation between the operating and the reference electrical impedance, and, if the deviation exceeds a defined tolerance, reporting the deviation to a user.

Abstract: An encoding device evaluates a plurality of processing and/or post-processing algorithms and/or methods to be applied to a video stream, and signals a selected method, algorithm, class or category of methods/algorithms either in an encoded bitstream or as side information related to the encoded bitstream. A decoding device or post-processor utilizes the signaled algorithm or selects an algorithm/method based on the signaled method or algorithm. The selection is based, for example, on availability of the algorithm/method at the decoder/post-processor and/or cost of implementation. The video stream may comprise, for example, downsampled multiplexed stereoscopic images and the selected algorithm may include any of upconversion and/or error correction techniques that contribute to a restoration of the downsampled images.

Abstract: Creative intent input describing emotion expectations and narrative information relating to media content is received. Expected physiologically observable states relating to the media content are generated based on the creative intent input. An audiovisual content signal with the media content and media metadata comprising the physiologically observable states is provided to a playback apparatus. The audiovisual content signal causes the playback device to use physiological monitoring signals to determine, with respect to a viewer, assessed physiologically observable states relating to the media content and generate, based on the expected physiologically observable states and the assessed physiologically observable states, modified media content to be rendered to the viewer.

Abstract: Methods, processes, and systems are presented for combining signal reshaping (also referred to as luma mapping chroma residuals scaling) with local illumination compensation (LIC) in video coding. Examples and trade-offs when the LIC model parameters are computed in the original domain, the reshaped domain, or a mixed domain, are presented.

Abstract: Some disclosed methods involve multi-band bass management. Some such examples may involve applying multiple high-pass and low-pass filter frequencies for the purpose of bass input management. Some disclosed methods treat at least some low-frequency signals as audio objects that can be panned. Some disclosed methods involve panning low and high frequencies separately. Following high-pass rendering, a power audit may determine a low-frequency deficit factor that is to be reproduced by subwoofers or other low-frequency-capable loudspeakers.

Abstract: Two corresponding color patches are displayed on two image displays until adjusted by a viewer to match visually to a common color. Two sets of code values rendered on the two corresponding color patches on the two image displays are identified. Two sets of tristimulus values for the viewer are determined based on the two sets of code values rendered on the two corresponding color patches on the two image displays. The viewer's color matching function are generated based on the two sets of tristimulus values. The viewer's CMF is used in image rendering operations on a target image display.

Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

Abstract: According to the present invention, an adaptive scheme is applied to an image encoding apparatus that includes an inter-predictor, an intra-predictor, a transformer, a quantizer, an inverse quantizer, and an inverse transformer, wherein input images are classified into two or more different categories, and two or more modules from among the inter-predictor, the intra-predictor, the transformer, the quantizer, and the inverse quantizer are implemented to perform respective operations in different schemes according to the category to which an input image belongs. Thus, the invention has the advantage of efficiently encoding an image without the loss of important information as compared to a conventional image encoding apparatus which adopts a packaged scheme.

Abstract: A device and method for video rendering. The device includes a memory and an electronic processor. The electronic processor is configured to receive, from a source device, video data including multiple reference viewpoints, determine a target image plane corresponding to a target viewpoint, determine, within the target image plane, one or more target image regions, and determine, for each target image region, a proxy image region larger than the corresponding target image region. The electronic processor is configured to determine, for each target image region, a plurality of reference pixels that fit within the corresponding proxy image region, project, for each target image region, the plurality of reference pixels that fit within the corresponding proxy image region to the target image region, producing a rendered target region from each target image region, and composite one or more of the rendered target regions to create video rendering.

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

Abstract: Systems and methods for adjusting video processing curves are disclosed herein. In one embodiment a method for applying an adjustment to an original curve derived from a set of input image data is disclosed, comprising: receiving a set of input image data to be adjusted; calculating an original curve from the set of input image data; receiving an adjustment curve, the adjustment curve based upon a desired image parameter; and applying the adjustment curve to the original curve to produce a resulting curve. In another embodiment, a display management unit (DMU) comprising a processor is disclosed that, upon receiving a set of input image data, processes the original curve according to: calculating an original curve from the set of input image data; receiving an adjustment curve, the adjustment curve based upon a desired image parameter; and applying the adjustment curve to the original curve to produce a resulting curve.

Abstract: Methods for performing dynamic range compression (DRC) on audio in a manner intended to produce output audio for playback by systems or devices with limited power handling capabilities and preferably also to reduce or prevent undesirable artifacts (e.g., pumping and/or breathing) in the output audio. Some embodiments perform the DRC so as to maximize average loudness (while preventing loss of quieter elements) during playback, and also to reduce or prevent distortion. Other aspects are systems or devices configured to perform embodiments of the method. In some embodiments, reduced DRC is applied when average loudness of the input audio approaches (or matches or exceeds) a target (e.g., a knee point for DRC, or a signal level near to a maximum playback level of the intended playback system), since such input audio is assumed to have already been compressed, and otherwise applying full DRC to the input audio.

Abstract: An apparatus and method of pre-conditioning audio for machine perception. Machine perception differs from human perception, and different processing parameters are used for machine perception applications (e.g., speech to text processing) as compared to those used for human perception applications (e.g., voice communications). These different parameters may result in pre-conditioned audio that is worsened for human perception yet improved for machine perception.

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: Methods for rendering audio for playback by two or more speakers are disclosed. The audio includes one or more audio signals, each with an associated intended perceived spatial position. Relative activation of the speakers may be a cost function of a model of perceived spatial position of the audio signals when played back over the speakers, a measure of proximity of the intended perceived spatial position of the audio signals to positions of the speakers, and one or more additional dynamically configurable functions. The dynamically configurable functions may be based on at least one or more properties of the audio signals, one or more properties of the set of speakers and/or one or more external inputs.

Abstract: A method and a device for encoding and decoding infra 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.