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: An apparatus and method of generating personalized HRTFs. The system is prepared by calculating a model for HRTFs described as the relationship between a finite example set of input data, namely anthropometric measures and demographic information for a set of individuals, and a corresponding set of output data, namely HRTFs numerically simulated using a high-resolution database of 3D scans of the same set of individuals. At the time of use, the system queries the user for their demographic information, and then from a series of images of the user, the system detects and measures various anthropometric characteristics. The system then applies the prepared model to the anthropometric and demographic data as part of generating a personalized HRTF. In this manner, the personalized HRTF can be generated with more convenience than by performing a high-resolution scan or an acoustic measurement of the user, and with less computational complexity than by numerically simulating their HRTF.

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: A computing device comprises a device image display outputting device display light; an optical configuration for a viewer of the computing device to view external display images rendered with external display light from an external image display and device display images rendered with the device display light; a display light combiner to combine the external display light and the device display light to reach the viewer's vision field. The external display light and the device display light are of different light properties. The display light combiner selectively reflects the device display light toward the viewer's vision field and selectively transmits the external display light toward the viewer's vision field.

Abstract: A forward reshaping mapping is generated to map a source image to a corresponding forward reshaped image of a lower dynamic range. The source image is spatially downsampled to generate a resized image into which noise is injected to generate a noise injected image. The forward reshaping mapping is applied to map the noise injected image to generate a noise embedded image of the lower dynamic range. A video signal is encoded with the noise embedded image and delivered to a recipient device for the recipient device to render a display image generated from the noise embedded image.

Abstract: The present document relates to audio source coding systems. In particular, the present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a first sample (615) of a first subband signal in a first subband of an audio signal is described. The first subband signal of the audio signal is determined using an analysis filterbank (612) comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the audio signal, respectively.

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: Methods for generating an AV bitstream (e.g., an MPEG-2 transport stream or bitstream segment having adaptive streaming format) such that the AV bitstream includes at least one video I-frame synchronized with at least one audio I-frame, e.g., including by re-authoring at least one video or audio frame (as a re-authored I-frame or a re-authored P-frame). Typically, a segment of content of the AV bitstream which includes the re-authored frame starts with an I-frame and includes at least one subsequent P-frame. Other aspects are methods for adapting such an AV bitstream, audio/video processing units configured to perform any embodiment of the inventive method, and audio/video processing units which include a buffer memory which stores at least one segment of an AV bitstream generated in accordance with any embodiment of the inventive method.

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: Methods for generating an object based audio program, renderable in a personalizable manner, and including a bed of speaker channels renderable in the absence of selection of other program content (e.g., to provide a default full range audio experience). Other embodiments include steps of delivering, decoding, and/or rendering such a program. Rendering of content of the bed, or of a selected mix of other content of the program, may provide an immersive experience. The program may include multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects), the bed of speaker channels, and other speaker channels. Another aspect is an audio processing unit (e.g., encoder or decoder) configured to perform, or which includes a buffer memory which stores at least one frame (or other segment) of an object based audio program (or bitstream thereof) generated in accordance with, any embodiment of the method.

Abstract: A computer implemented system for rendering captured audio soundfields to a listener comprises apparatus to deliver the audio soundfields to the listener. The delivery apparatus delivers the audio soundfields to the listener with first and second audio elements perceived by the listener as emanating from first and second virtual source locations, respectively, and with the first audio element and/or the second audio element delivered to the listener from a third virtual source location. The first virtual source location and the second virtual source location are perceived by the listener as being located to the front of the listener, and the third virtual source location is located to the rear or the side of the listener.

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: Light from an array of laser light sources are spread to cover the modulating face of a DMD or other modulator. The spread may be performed, for example, by a varying curvature array of lenslets, each laser light directed at one of the lenslets. Light from neighboring and/or nearby light sources overlap at a modulator. The lasers are energized at different energy/brightness levels causing the light illuminating the modulator to itself be modulated (locally dimmed). The modulator then further modulates the locally dimmed lights to produce a desired image. A projector according to the invention may utilize, for example, a single modulator sequentially illuminated or separate primary color modulators simultaneously illuminated.

Abstract: A non-random-access video stream is received. A first image block is encoded after second image blocks according to a non-random-access processing order. View direction data is received to indicate a viewer's view direction coinciding with a location covered by the first image block. The first image block is encoded into the random-access video stream before the second image blocks in a random-access processing order. The random-access video stream is delivered to a recipient decoding device operated by the viewer to cause the first image block to be processed and rendered before the second image blocks according to the random-access processing order.

Abstract: A multi-view image stream encoded with primary and secondary image is accessed. Each primary image stream comprises groups of pictures (GOPs). Each secondary image stream comprises I-frames generated from a corresponding primary image stream. Viewpoint data collected in real time is received from a recipient decoding device to indicate that the viewer's viewpoint has changed from a specific time point. A camera is selected based on the viewer's changed viewpoint. It is determined whether the specific time point corresponds to a non-I-frame in a GOP of a primary image stream of the selected camera. If so, an I-frame from a secondary image stream corresponding to the primary image stream is transmitted to the recipient decoding device.

Abstract: Methods and systems for generating a set of forward and backward reshaping functions for the efficient coding of high-dynamic range (HDR) images are provided. Given an initial set of forward reshaping functions, output forward reshaping functions are constructed by a) using the forward reshaping functions to generate a first set of corresponding backward reshaping functions b) generating a second set of backward reshaping functions using a multi-segment polynomial representation with a common set of pivot points c) generating an output set of backward reshaping functions by optimizing the polynomial representation of the second set of backward reshaping functions to minimize gap values between consecutive segments and d) using the output set of backward reshaping functions to generate the output set of forward reshaping functions by minimizing the distance between original input HDR codewords and reconstructed HDR codewords.

Abstract: The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

Abstract: 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: 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: 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: Smaller halftone tiles are implemented on a first modulator of a dual modulation projection system. This technique uses multiple halftones per frame in the pre-modulator synchronized with a modified bit sequence in the primary modulator to effectively increase the number of levels provided by a given tile size in the halftone modulator. It addresses the issue of reduced contrast ratio at low light levels for small tile sizes and allows the use of smaller PSFs which reduce halo artifacts in the projected image and may be utilized in 3D projecting and viewing.

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

Abstract: The present invention provides a method and a device for deriving an inter-view motion merging candidate. A method for deriving an inter-view motion merging candidate, according to an embodiment of the present invention, can comprise the steps of: on the basis of encoding information of an inter-view reference block derived by means of a variation vector of a current block, determining whether or not inter-view motion merging of the current block is possible; and, if inter-view motion merging of the current block is not possible, generating an inter-view motion merging candidate of the current block by using encoding information of an adjacent block that is spatially adjacent to the inter-view reference block.

Abstract: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder.

Abstract: In a cloud-based system for encoding high dynamic range (HDR) video, a computing node is assigned to be a dispatcher node, segmenting the input video into scenes and generating a scene to segment allocation to be used by other computing nodes. The scene to segment allocation process includes one or more iterations with an initial random assignment of scenes to computing nodes, followed by a refined assignment based on optimizing the allocation cost across all the computing nodes. Methods to generate scene-based forward and backward reshaping functions to optimize video coding and improve the coding efficiency of reshaping-related metadata are also examined.

Abstract: Systems and methods of rendering DCI-compliant image data on Enhanced Dynamic Range (EDR) display systems are disclosed. One embodiment of an EDR projector system comprises a first modulator and a second modulator. One method for rendering DCI-compliant image data on an EDR projector system comprises: receiving input image data, said image data comprising a plurality of image formats; determining whether the input image data comprises DCI image data; if the input image data comprises DCI image data, then performing dynamic range (DR) processing on the DCI image data; and rendering the dynamic range processed DCI image data on the EDR projector system. One DR processing method is to set the first modulator to a desired luminance level—e.g., fully ON or a ratio of DCI max luminance to the EDR max luminance. In addition, a desired minimum level of luminance may be set for the EDR projector.

Abstract: Described are methods of processing audio data for hum noise detection and/or removal. The audio data comprises a plurality of frames. One method incudes: classifying frames of the audio data as either content frames or noise frames, using one or more content activity detectors; determining a noise spectrum from one or more frames of the audio data that are classified as noise frames; determining one or more hum noise frequencies based on the determined noise spectrum; generating an estimated hum noise signal based on the one or more hum noise frequencies; and removing hum noise from at least one frame of the audio data based on the estimated hum noise signal. Also described are apparatus for carrying out the methods, as well as corresponding programs and computer-readable storage media.

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. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

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: Images of an actual rendering environment are acquired through image sensors operating in conjunction with a media consumption system. The acquired images of the actual rendering environment are used to predict audio characteristics of objects present in the actual rendering environment. Spatial audio rendered, to a user in the actual rendering environment, by audio speakers operating in conjunction with the media consumption system is adjusted or modified based at least in part on the audio characteristics of the objects present in the actual rendering environment.

Abstract: At a first time point, a first light capturing device at a first spatial location in a three-dimensional (3D) space captures first light rays from light sources located at designated spatial locations on a viewer device in the 3D space. At the first time point, a second light capturing device at a second spatial location in the 3D space captures second light rays from the light sources located at the designated spatial locations on the viewer device in the 3D space. Based on the first light rays captured by the first light capturing device and the second light rays captured by the second light capturing device, at least one of a spatial position and a spatial direction, at the first time point, of the viewer device is determined.

Abstract: In some embodiments, a pitch filter for filtering a preliminary audio signal generated from an audio bitstream is disclosed. The pitch filter has an operating mode selected from one of either: (i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal, and (ii) an inactive mode where the pitch filter is disabled. The preliminary audio signal is generated in an audio encoder or audio decoder having a coding mode selected from at least two distinct coding modes, and the pitch filter is capable of being selectively operated in either the active mode or the inactive mode while operating in the coding mode based on control information.

Abstract: Novel methods and systems for compensating for ambient light around displays are disclosed. A shift in the PQ curve applied to an image can compensate for sub-optimal ambient light conditions for a display, with the PQ shift being either an addition to a compensation value in PQ space followed by a subtraction of the compensation value in linear space, or an addition to the compensation value in linear space followed by a subtraction of the compensation value in PQ space. Further adjustments to the PQ curve can also be made to provide an improved image quality with respect to image luminance.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding of a vector of parameters in an audio coding system. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system. According to the disclosure, a modulo differential approach for coding and encoding a vector of a non-periodic quantity may improve the coding efficiency and provide encoders and decoders with less memory requirements. Moreover, an efficient method for encoding and decoding a sparse matrix is provided.

Abstract: One or more media contents are received. A viewer's light adaptive states are predicted as a function of time as if the viewer is watching display mapped images derived from the one or more media contents. The viewer's light adaptive states are used to detect an excessive change in luminance in a specific media content portion of the one or more media contents. The excessive change in luminance in the specific media content portion of the one or more media contents is caused to be reduced while the viewer is watching one or more corresponding display mapped images derived from the specific media content portion of the one or more media contents.

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: Methods, systems, and computer program products for network-based processing and distribution of multimedia content of a live performance are disclosed. In some implementations, recording devices can be configured to record a multimedia event (e.g., a musical performance). The recording devices can provide the recordings to a server while the event is ongoing. The server automatically synchronizes, mixes and masters the recordings. The server performs the automatic mixing and mastering using reference audio data previously captured during a rehearsal. The server streams the mastered recording to multiple end users through the Internet or other public or private network. The streaming can be live streaming.

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 (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: 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: A novel projection system includes a light source, a phase modulator, an amplitude modulator, and a controller having temporal lightfield simulation capabilities. The phase modulator spatially modulates a lightfield from the light source to generate an intermediate image on the amplitude modulator. The amplitude modulator spatially modulates the intermediate image to form a final image. The controller models the phase state of the phase modulator during transitions between phase modulator frames and generates lightfield simulations of the intermediate image during the transition. The controller utilizes the lightfield simulations to generate and provide sets of amplitude drive values to the amplitude modulator at a faster rate than that at which the phase modulator is capable of switching.

Abstract: An audio processing system (100) accepts an audio bitstream having one of a plurality of predefined audio frame rates. The system comprises a front-end component (110), which receives a variable number of quantized spectral components, corresponding to one audio frame in any of the predefined audio frame rates, and performs an inverse quantization according to predetermined, frequency-dependent quantization levels. The front-end component may be agnostic of the audio frame rate. The audio processing system further comprises a frequency-domain processing stage (120) and a sample rate converter (130), which provide a reconstructed audio signal sampled at a target sampling frequency independent of the audio frame rate. By its frame-rate adaptability, the system can be configured to operate frame-synchronously in parallel with a video processing system that accepts plural video frame rates.

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

Abstract: Apparatus and methods for providing solutions to the problem of preserving original creative intent for video playback on a target display are presented herein. According to one aspect, a video bitstream includes metadata with a flag indicative of creative intent for a target display. This metadata may include numerous fields that denote characteristics such as content type, content sub-type, intended white point, whether or not to use the video in Reference Mode, intended sharpness, intended noise reduction, intended MPEG noise reduction, intended Frame Rate Conversion, intended Average Picture Level, and intended color. This metadata is designed to make it effortless for the content creators to tag their content. The metadata can be added to the video content at multiple points, the status of the flag is set to TRUE or FALSE to indicate whether the metadata was added by the content creator or a third party.

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 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: A method (910) for rendering an audio signal in a virtual reality rendering environment (180) is described. The method (910) comprises rendering (911) an origin audio signal of an audio source (311, 312, 313) from an origin source position on an origin sphere (114) around an origin listening position (301) of a listener (181). Furthermore, the method (900) comprises determining (912) that the listener (181) moves from the origin listening position (301) to a destination listening position (302). In addition, the method (900) comprises determining (913) a destination source position of the audio source (311, 312, 313) on a destination sphere (114) around the destination listening position (302) based on the origin source position, and determining (914) a destination audio signal of the audio source (311, 312, 313) based on the origin audio signal.

Abstract: Media input audio data corresponding to a media stream and microphone input audio data from at least one microphone may be received. A first level of at least one of a plurality of frequency bands of the media input audio data, as well as a second level of at least one of a plurality of frequency bands of the microphone input audio data, may be determined. Media output audio data and microphone output audio data may be produced by adjusting levels of one or more of the first and second plurality of frequency bands based on the perceived loudness of the microphone input audio data, of the microphone output audio data, of the media output audio data and the media input audio data. One or more processes may be modified upon receipt of a mode-switching indication.

Abstract: Described are methods of processing an audio signal for packet loss concealment. The audio signal comprises a sequence of frames, each frame containing representations of a plurality of audio channels and reconstruction parameters for upmixing the plurality of audio channels to a predetermined channel format. One method includes: receiving the audio signal; and generating a reconstructed audio signal in the predefined channel format based on the received audio signal. Generating the reconstructed audio signal comprises: determining whether at least one frame of the audio signal has been lost; and if a number of consecutively lost frames exceeds a first threshold, fading the reconstructed audio signal to a predefined spatial configuration. Also described is a method of encoding an audio signal. Yet further described are apparatus for carrying out the methods, as well as corresponding programs and computer-readable storage media.

Abstract: Multiple virtual source locations may be defined for a volume within which audio objects can move. A set-up process for rendering audio data may involve receiving reproduction speaker location data and pre-computing gain values for each of the virtual sources according to the reproduction speaker location data and each virtual source location. The gain values may be stored and used during “run time,” during which audio reproduction data are rendered for the speakers of the reproduction environment. During run time, for each audio object, contributions from virtual source locations within an area or volume defined by the audio object position data and the audio object size data may be computed. A set of gain values for each output channel of the reproduction environment may be computed based, at least in part, on the computed contributions. Each output channel may correspond to at least one reproduction speaker of the reproduction environment.