Abstract: Methods and apparatus for decoding a compressed Higher Order Ambisonics (HOA) representation of a sound or soundfield. The method may include receiving a bit stream containing the compressed HOA representation and decoding, based on a determination that there are multiple layers, the compressed HOA representation from the bitstream to obtain a sequence of decoded HOA representations. A first subset of the sequence of decoded HOA representations is determined based only on corresponding ambient HOA components. A second subset of the sequence of decoded HOA representations is determined based on corresponding ambient HOA components and corresponding predominant sound components.

Abstract: A system and method of adjusting an audio output in one location so that its propagation into another location is reduced. As a first device in a first location generates sound, a second device in a second location detects the propagated sound. The first device then adjusts its output based on the detected sound.

Abstract: A novel projection system includes first and second light sources (e.g., sets of lasers), a spatial light modulator (SLM) that receives light from the first light source, and a beam steering device that receives light from the second light source and steers the light to highlight regions of the SLM. The SLM then modulates the light from both light sources to generate a highlighted imaging beam which can then be projected on a viewing surface. The highlighted imaging beam can represent a highlighted 2D image or a highlighted left- or right-eye view of a 3D image. The projection system thus improves peak brightness in the displayed highlighted images without incorporating a separate highlight projector or other expensive equipment. Methods for highlighting projected images are also described.

Abstract: A display system comprises light sources configured to emit first light with a first spectral power distribution; light regeneration layers configured to be stimulated by the first light and to convert at least a portion of the first light and recycled light into second light, the second light comprising (a) primary spectral components that correspond to primary colors and (b) secondary spectral components that do not correspond to the primary colors; and notch filter layers configured to receive a portion of the second light and to filter out the secondary spectral components from the portion of the second light. The portion of the second light can be directed to a viewer of the display system and configured to render images viewable to the viewer.

Abstract: Methods and systems for improving coding decoding efficiency of video by providing a syntax modeler, a buffer, and a decoder. The syntax modeler may associate a first sequence of symbols with syntax elements. The buffer may store tables, each represented by a symbol in the first sequence, and each used to associate a respective symbol in a second sequence of symbols with encoded data. The decoder decodes the data into a bitstream using the second sequence retrieved from a table.

Abstract: The invention discloses rendering sound field signals, such as Higher-Order Ambisonics (HOA), for arbitrary loudspeaker setups, where the rendering results in highly improved localization properties and is energy preserving. This is obtained by rendering an audio sound field representation for arbitrary spatial loudspeaker setups and/or by a a decoder that decodes based on a decode matrix (D). The decode matrix (D) is based on smoothing and scaling of a first decode matrix {circumflex over (D)} with smoothing coefficients. The first decode matrix {circumflex over (D)} is based on a mix matrix G and a mode matrix {tilde over (?)}, where the mix matrix G was determined based on L speakers and positions of a spherical modelling grid related to a HOA order N, and the mode matrix {tilde over (?)} was determined based on the spherical modelling grid and the HOA order N.

Abstract: Improved methods and/or apparatus for decoding an encoded audio signal in soundfield format for L loudspeakers. The method and/or apparatus can render an Ambisonics format audio signal to 2D loudspeaker setup(s) based on a rendering matrix. The rendering matrix has elements based on loudspeaker positions and wherein the rendering matrix is determined based on weighting at least an element of a first matrix with a weighting factor g = 1 L . The first matrix is determined based on positions of the L loudspeakers and at least a virtual position of at least a virtual loudspeaker that is added to the positions of the L loudspeakers.

Abstract: An encoding system encodes an N-channel audio signal (X), wherein N?3, as a single-channel downmix signal (Y) together with dry and wet upmix parameters ({tilde over (C)}, {tilde over (P)}). In a decoding system, a decorrelating section outputs, based on the downmix signal, an (N?1)-channel decorrelated signal (Z); a dry upmix section maps the downmix signal linearly in accordance with dry upmix coefficients (C) determined based on the dry upmix parameters; a wet upmix section populates an intermediate matrix based on the wet upmix parameters and knowing that the intermediate matrix belongs to a predefined matrix class, obtains wet upmix coefficients (P) by multiplying the intermediate matrix by a predefined matrix, and maps the decorrelated signal linearly in accordance with the wet upmix coefficients; and a combining section combines outputs from the upmix sections to obtain a reconstructed signal ({circumflex over (X)}) corresponding to the signal to be reconstructed.

Abstract: An apparatus and method of rendering audio. A binaural signal is split on an amplitude weighting basis into a front binaural signal and a rear binaural signal, based on perceived position information of the audio. In this manner, the front-back differentiation of the binaural signal is improved.

Abstract: Disclosed are a method for determining a color difference component quantization parameter and a device using the method. Method for decoding an image can comprise the steps of: decoding a color difference component quantization parameter offset on the basis of size information of a transform unit; and calculating a color difference component quantization parameter index on the basis of the decoded color difference component quantization parameter offset. Therefore, the present invention enables effective quantization by applying different color difference component quantization parameters according to the size of the transform unit when executing the quantization.

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: Systems, methods, and computer program products for frequency-domain estimation of latency between audio signals. In some embodiments, the estimation is performed on first blocks of data indicative of samples of a first audio signal and second blocks of data indicative of samples of a second audio signal, and includes determining a coarse latency estimate, including by determining gains which, when applied to some of the second blocks, determine estimates of one of the first blocks, and identifying one of the estimates as having a best spectral match to said one of the first blocks. A refined latency estimate is determined from the coarse estimate and some of the gains. Optionally, at least one metric indicative of confidence in the refined latency estimate is generated. Audio processing (e.g., echo cancellation) may be performed on the frequency-domain data, including by performing time alignment based on the refined latency estimate.

Abstract: Disclosed are a method for determining a color difference component quantization parameter and a device using the method. Method for decoding an image can comprise the steps of: decoding a color difference component quantization parameter offset on the basis of size information of a transform unit; and calculating a color difference component quantization parameter index on the basis of the decoded color difference component quantization parameter offset. Therefore, the present invention enables effective quantization by applying different color difference component quantization parameters according to the size of the transform unit when executing the quantization.

Abstract: A method for decoding a video bitstream is disclosed. The method comprises: entropy decoding a first portion of a video bitstream, wherein first portion of video bitstream is associated with a video frame, thereby producing a first portion of decoded data; entropy decoding a second portion of video bitstream, wherein second portion of video bitstream is associated with video frame, thereby producing a second portion of decoded data, wherein entropy decoding second portion of video bitstream is independent of entropy decoding first portion of video bitstream; and reconstructing a first portion of video frame associated with video bitstream using first portion of decoded data and second portion of decoded data.

Abstract: Different candidate image data feature types are evaluated to identify one or more specific image data feature types to be used in training a prediction model for optimizing one or more image metadata parameters. A plurality of image data features of the one or more selected image data feature types is extracted from one or more images. The plurality of image data features of the one or more selected image data feature types is reduced into a plurality of significant image data features. A total number of image data features in the plurality of significant image data features is no larger than a total number of image data features in the plurality of image data features of the one or more selected image data feature types. The plurality of significant image data features is applied to training the prediction model for optimizing one or more image metadata parameters.

Abstract: The present disclosure relates to methods, apparatus and systems for encoding an audio signal into a bitstream, in particular at an encoder, comprising: encoding or including audio signal data associated with 3DoF audio rendering into one or more first bitstream parts of the bitstream, and encoding or including metadata associated with 6DoF audio rendering into one or more second bitstream parts of the bitstream. The present disclosure further relates to methods, apparatus and systems for decoding an audio signal and audio rendering based on the bitstream.

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

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

Abstract: A novel light source for a 3D display system includes a plurality of left eye light emitters and a plurality of right eye light emitters. The left eye emitters include a broad spectral distribution emitter and an overlapping narrow spectral distribution emitter in each of the blue, green, and red color bands. Similarly, the right eye emitters include a broad spectral distribution emitter and an overlapping narrow spectral distribution emitter in each of the blue, green, and red color bands. The combined spectral distributions of each of the broad and narrow emitters provide a primary light for each color and for each eye that has a desirable spectral shape, including wide bandwidth and short tail(s). The invention thus minimizes cross-talk and speckling in left- and right-eye images of 3D display systems.

Abstract: Various embodiments are disclosed for (possibly simultaneously) applying EQ and DRC to audio signals. In an embodiment, a method comprises: dividing an input audio signal into n frames, where n is a positive integer greater than one; dividing each frame of the input audio signal into Nb frequency bands, where Nb is a positive integer greater than one; for each frame n: computing an input level of the input audio signal in each band f, resulting in a input audio level distribution for the input audio signal; computing a gain for each band f based at least in part on a mapping of one or more properties of the input audio level distribution to a reference N audio level distribution computed from one or more reference audio signals; and applying each computed gain for each band f to each corresponding band f of the input audio signal.

Abstract: A method for representing a second presentation of audio channels or objects as a data stream, the method comprising the steps of: (a) providing a set of base signals, the base signals representing a first presentation of the audio channels or objects; (b) providing a set of transformation parameters, the transformation parameters intended to transform the first presentation into the second presentation; the transformation parameters further being specified for at least two frequency bands and including a set of multi-tap convolution matrix parameters for at least one of the frequency bands.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: Embodiments are described for a method of simultaneously localizing a set of speakers and microphones, having only the times of arrival between each of the speakers and microphones. An autodiscovery process uses an external input to set: a global translation (3 continuous parameters), a global rotation (3 continuous parameters), and discrete symmetries, i.e., an exchange of any axis pairs and/or reversal of any axis. Different time of arrival acquisition techniques may be used, such as ultrasonic sweeps or generic multitrack audio content. The autodiscovery algorithm is based in minimizing a certain cost function, and the process allows for latencies in the recordings, possibly linked to the latencies in the emission.

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: Embodiments relate to an audio processing unit that includes a bitstream payload deformatter and a decoding subsystem. The decoding subsystem is coupled to the bitstream payload deformatter and configured to decode at least a portion of a block of an encoded audio bitstream. The block includes a fill element with an identifier indicating a start of the fill element and fill data after the identifier. The fill data includes at least one flag identifying whether a base form of spectral band replication or an enhanced form of spectral band replication is to be performed on audio content of the block. The identifier is a three bit unsigned integer transmitted most significant bit first and having a value of 0x6.

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 data structure defining a high dynamic range image comprises a tone map having a reduced dynamic range and HDR information. The high dynamic range image can be reconstructed from the tone map and the HDR information. The data structure can be backwards compatible with legacy hardware or software viewers. The data structure may comprise a JFIF file having the tone map encoded as a JPEG image with the HDR information in an application extension or comment field of the JFIF file, or a MPEG file having the tone map encoded as a MPEG image with the HDR information in a video or audio channel of the MPEG file. Apparatus and methods for encoding or decoding the data structure may apply pre- or post correction to compensate for lossy encoding of the high dynamic range information.

Abstract: The disclosed embodiments enable converting audio signals captured in various formats by various capture devices into a limited number of formats that can be processed by an audio codec (e.g., an Immersive Voice and Audio Services (IVAS) codec). In an embodiment, a simplification unit of the audio device receives an audio signal captured by one or more audio capture devices coupled to the audio device. The simplification unit determines whether the audio signal is in a format that is supported/not supported by an encoding unit of the audio device. Based on the determining, the simplification unit, converts the audio signal into a format that is supported by the encoding unit. In an embodiment, if the simplification unit determines that the audio signal is in a spatial format, the simplification unit can convert the audio signal into a spatial “mezzanine” format supported by the encoding.

Abstract: A wearable device for augmented media content experiences can be formed with a mountable physical structure that has removably mountable positions and component devices that are removably mounted through the removably mountable positions. The component devices can be specifically selected based on a specific type of content consumption environment in which the wearable device is to operate. The mountable physical structure may be subject to a device washing process to which the component devices are not subject to, after the wearable device including the mountable physical structure and the component devices is used by a viewer in a content consumption session in the specific type of content consumption environment, so long as the component devices are subsequently removed from the mountable physical structure.

Abstract: The present invention relates to an intra prediction mode mapping method and a device using the method. The intra prediction mode includes: decoding flag information providing information regarding whether an intra prediction mode of a plurality of candidate intra prediction modes for the current block is the same as the intra prediction mode for the current block, and decoding a syntax component including information regarding the intra prediction mode for the current block in order to induce the intra prediction mode for the current block if the intra prediction mode from among the plurality of candidate intra prediction modes for the current block is not the same as the intra prediction mode for the current block. Thus, it is possible to increase the efficiency with which are images are decoded.

Abstract: There are provided decoding and encoding methods for encoding and decoding of multichannel audio content for playback on a speaker configuration with N channels. The decoding method comprises decoding, in a first decoding module, M input audio signals into M mid signals which are suitable for playback on a speaker configuration with M channels; and for each of the N channels in excess of M channels, receiving an additional input audio signal corresponding to one of the M mid signals and decoding the input audio signal and its corresponding mid signal so as to generate a stereo signal including a first and a second audio signal which are suitable for playback on two of the N channels of the speaker configuration.

Abstract: In an implementation, a supplemental sequence parameter set (“SPS”) structure is provided that has its own network abstraction layer (“NAL”) unit type and allows transmission of layer-dependent parameters for non-base layers in an SVC environment. The supplemental SPS structure also may be used for view information in an MVC environment. In a general aspect, a structure is provided that includes (1) information (1410) from an SPS NAL unit, the information describing a parameter for use in decoding a first-layer encoding of a sequence of images, and (2) information (1420) from a supplemental SPS NAL unit having a different structure than the SPS NAL unit, and the information from the supplemental SPS NAL unit describing a parameter for use in decoding a second-layer encoding of the sequence of images. Associated methods and apparatuses are provided on the encoder and decoder sides, as well as for the signal.

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: An auto exposure method for a spatially-multiplexed-exposure (SME) high-dynamic-range (HDR) image sensor includes (a) retrieving raw image data from an exposure of the spatially-multiplexed-exposure high-dynamic-range image sensor, (b) preprocessing long-exposure pixel values and short-exposure pixel values of the raw image data to remove therefrom long-exposure pixel values and short-exposure pixel values failing to meet one or more quality requirements, (c) synthesizing, into an high-dynamic-range histogram, the long-exposure pixel values remaining after the step of preprocessing and the short-exposure pixel values remaining after the step of preprocessing, (d) deriving a goodness metric from the high-dynamic-range histogram, (e) adjusting at least one of the long exposure time and the short exposure time, based at least in part upon the goodness metric, and (f) outputting the at least one of the long exposure time and the short exposure time, as adjusted, to the spatially-multiplexed-exposure high-dynamic-

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

Abstract: An audio processing unit (APU) is disclosed. The APU includes a buffer memory configured to store at least one frame of an encoded audio bitstream, where the encoded audio bitstream includes audio data and a metadata container. The metadata container includes a header and one or more metadata payloads after the header. The one or more metadata payloads include dynamic range compression (DRC) metadata, and the DRC metadata is or includes profile metadata indicative of whether the DRC metadata includes dynamic range compression (DRC) control values for use in performing dynamic range compression in accordance with at least one compression profile on audio content indicated by at least one block of the audio data.

Abstract: Given a sequence of images in a first codeword representation, methods, processes, and systems are presented for image reshaping using rate distortion optimization, wherein reshaping allows the images to be coded in a second codeword representation which allows more efficient compression than using the first codeword representation. Syntax methods for signaling reshaping parameters are also presented.

Abstract: Bordering pixels delineating a texture hole region in an image are identified. Depth values of the bordering pixels are recorded. The depth values are automatically clustered into two depth value clusters with a depth value threshold separating the two depth value clusters. A subset of bordering background pixels is identified in the bordering pixels as those with depth values in one of the two depth value clusters that is declared as a background depth value cluster. The subset of bordering background pixels is used to predict texture hole pixel values in the texture hole region based on multiple candidate prediction directions. Quality indicator values are computed for the multiple candidate prediction directions and used to select a specific candidate prediction direction for filling in final texture hole pixel values in the texture hole region of the image.

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 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 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: Methods (700, 800, 900), systems (200, 300, 400, 500, 600) and computer program products are provided. Location metadata (620) associated with an audio object is received (801). The location metadata defines a position of the audio object in an audio scene. It is estimated (630, 802), based on the location metadata, whether the audio object includes dialog. A value representative of a result of the estimation is assigned (803) to an object type parameter (231). In some example embodiments, audio objects are selected (661, 662, 804) based on values of their respective of object type parameters. In some example embodiments, at least one of the selected audio objects is submitted to dialog enhancement (690, 807).

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: Three dimensional (3D) glasses suited for wearers with varying facial geometries may include a frame adapted to position spectrally filtering lenses at a particular distance from the eyes of the wearer. The 3D glasses may include a means for adjusting the distance between the lenses and the eyes of the wearer. The lenses may include positive runout.

Abstract: A system for decoding a video bitstream includes receiving a frame of the video that includes at least one slice and at least one tile and where each of the at least one slice and the at least one tile are not all aligned with one another.

Abstract: Methods and systems for generating an interpolated reshaping function for the efficient coding of high-dynamic range images are provided. The interpolated reshaping function is constructed based on a set of pre-computed basis reshaping functions. Interpolation schemes are derived for pre-computed basis reshaping functions represented as look-up tables, multi-segment polynomials, or matrices of coefficients in a multivariate, multi-regression representation. Encoders and decoders using asymmetric reshaping and interpolated reshaping functions for mobile applications are also presented.

Abstract: A display has a screen which incorporates a light modulator. The screen may be a front projection screen or a rear-projection screen. The screen is illuminated with light from a light source comprising an array of controllable light-emitters. The controllable-emitters and elements of the light modulator may be controlled to adjust the intensity of light emanating from corresponding areas on the screen. The display may provide a high dynamic range.

Abstract: Disclosed is a prediction method adopting in-loop filtering. According to the present invention, a prediction method for encoding and decoding video comprises the following steps: generating a residual block of the current block through an inverse quantization and inverse transform; generating a prediction block of the current block through an intra-prediction; performing in-loop filtering on the current block in which the residual block and the prediction block are combined; and storing the current block, on which the in-loop filtering is performed, in a frame buffer for an intra-prediction of the next block to be encoded. As described above, prediction is performed using an in-loop filter during processes for encoding and decoding video, thereby improving the accuracy of prediction and reducing errors in prediction, thus improving the efficiency of video compression and reducing the amount of data to be transmitted.