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: 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: Methods and systems of improving bass response for a speaker in a portable computing device are described. One portable computing device includes first and second cover parts that are joined together to form a casing of the portable computing device, wherein a speaker volume is formed between portions of the first and second cover parts; a speaker arranged within the speaker volume; and one or more elastic spacers arranged between the first and second cover parts. The one or more elastic spacers are arranged to counteract, by their elastic recoil forces, a compression of the speaker volume when the first and second cover parts are under external compressing forces. The one or more elastic spacers are arranged between the first and second cover parts to be partially compressed by the first and second cover parts in the absence of external compressing forces on the first and second cover parts.

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: 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: Some implementations involve receiving, from a first subband domain acoustic echo canceller (AEC) of a first audio device in an audio environment, first adaptive filter management data from each of a plurality of first adaptive filter management modules, each first adaptive filter management module corresponding to a subband of the first subband domain AEC, each first adaptive filter management module being configured to control a first plurality of adaptive filters. The first plurality of adaptive filters may include at least a first adaptive filter type and a second adaptive filter type. Some implementations involve extracting, from the first adaptive filter management data, a first plurality of extracted features corresponding to a plurality of subbands of the first subband domain AEC and estimating a current local acoustic state based, at least in part, on the first plurality of extracted features.

Abstract: Embodiments are disclosed for timestamp smoothing to remove jitter. In some embodiments, a method of smoothing timestamps associated with audio packets comprises: receiving, using at least one processor, a series of input timestamps for audio packets and their respective packet lengths; estimating, using the at least one processor, an initial timestamp based on the series of input timestamps, the packet lengths and a sample time; calculating, using the at least one processor, a predicted timestamp based on the estimated initial timestamp; and smoothing, using the at least one processor, the predicted timestamp.

Abstract: Methods for generating an object based audio program which is renderable in a personalizable manner, e.g., to provide an immersive, perception of audio content of the program. Other embodiments include steps of delivering (e.g., broadcasting), decoding, and/or rendering such a program. Rendering of audio objects indicated by the program may provide an immersive experience. The audio content of the program may be indicative of multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects, and typically also a default set of objects which will be rendered in the absence of a selection by a user) and a bed of 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: 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: 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: In an embodiment, a method comprises: receiving bands of power spectra of an input audio signal and a microphone covariance, and for each band: estimating, using a classifier, respective probabilities of speech and noise; estimating, using a directionality model, a set of means for speech and noise, or a set of means and covariances for speech and noise, based on the microphone covariance for the band and the probabilities; estimating, using a level model, a mean and covariance of noise power based on the probabilities and the power spectra; determining a first noise suppression gain based on the directionality model; determining a second noise suppression gain based on the level model; selecting the first or second noise suppression gain or their sum based on a signal-to-noise ratio of the input audio signal; and scaling a time-frequency representation of the input signal by the selected noise suppression gain.

Abstract: A distributed amplification and packetized audio transmission system for clock synchronization and alignment between an audio/power source and endpoints with dedicated amplifiers and speakers. An Ethernet audio signal is combined with a Power-Line Communications (PLC) signal for transmission from the source to the endpoints over a common conductor. A single master clock in the source synchronizes the Ethernet audio transmitter with the PLC transmitter. Each end-point has a PLC receiver to recover the master clock for use by its Ethernet audio receiver to provide reliable clock synchronization between the source clock and the endpoint clocks. The endpoints can adjust and re-timestamp the PTP packetized clock based upon symbol and timing information from the PLC receiver.

Abstract: Systems, methods, and computer program products for audio processing based on convolutional neural network (CNN) are described. A first CNN architecture may comprise a contracting path of a U-net, a multi-scale CNN, and an expansive path of a U-net. The contracting path may comprise a first encoding layer and may be configured to generate an output representation of the contracting path. The multi-scale CNN may be configured to generate, based on the output representation of the contracting path, an intermediate representation. The multi-scale CNN may comprise at least two parallel convolution paths. The expansive path may comprise a first decoding layer and may be configured to generate a final representation based on the intermediate representation generated by the multi-scale CNN.

Abstract: A neural network system is provided, implementing a generative model for autoregressively generating a distribution for a plurality of current filter-bank samples of an audio signal, wherein the current samples correspond to a current time slot, and each current sample corresponds to a channel of the filter-bank. The system includes a hierarchy of a plurality of neural network processing tiers ordered from a top to a bottom tier, each tier trained to generate conditioning information based on previous filter-bank samples and, for at least each tier but the top tier, also on the conditioning information from a tier higher up in the hierarchy, and an output stage trained to generate the probability distribution based on previous samples for one or more previous time slots and the conditioning information from the lowest processing tier.

Abstract: Described herein is a method of processing an audio signal using a neural network or using a first and a second neural network. Described is further a method of training said neural network or of jointly training a set of said first and said second neural network. Moreover, described is a method of obtaining and transmitting a latent feature space representation of a perceptual domain audio signal using a neural network and a method of obtaining an audio signal from a latent feature space representation of a perceptual domain audio signal using a neural network. Described are also respective apparatuses and computer program products.

Abstract: A neural network system for predicting frequency coefficients of a media signal, the neural network system comprising a time predicting portion including at least one neural network trained to predict a first set of output variables representing a specific frequency band of a current time frame given coefficients of one or several previous time frames, and a frequency predicting portion including a at least one neural network trained to predict a second set of output variables representing a specific frequency band given coefficients of one or several frequency bands adjacent to the specific frequency band in said current time frame. Such a neural network system forms a predictor capable of capturing both temporal and frequency dependencies occurring in time-frequency tiles of a media signal.

Abstract: A projection system and calibration method therefore relate to a light source configured to emit a light in response to an image data, an illumination optical system configured to steer the light, the illumination optical system including a fold mirror and an integrating rod, a digital micromirror device (DMD) including a plurality of micromirrors respectively configured to reflect the steered light to a predetermined location as on-state light or to reflect the steered light as off-state light to a light dump; determining a deviation between an actual angle of orientation and an expected angle of orientation of a respective micromirror of the plurality of micromirrors; calculating a first amount of rotational adjustment corresponding to the fold mirror and a second amount of lateral adjustment corresponding to the integrating rod, and actuating the fold minor and integrating rod according to the corresponding first and second amount.

Abstract: Embodiments are described for a high-frequency waveguide that improves the performance of large-scale surround sound and immersive audio environments. A horn waveguide is configured to be asymmetric about one of a vertical axis and horizontal axis of the waveguide to form an asymmetric horn waveguide. A spherical enclosure surrounds the asymmetric horn waveguide to form a horn speaker, and a three-axis mounting system is configured to fix the horn speaker to one of a wall or ceiling surface of the venue, wherein the mounting system facilitates rotating the horn speaker to a location that provides maximum coverage of the venue within the passband of the asymmetric horn waveguide.

Abstract: In a cloud-based system for encoding high dynamic range (HDR) video, each node receives a video segment and bumper frames. Each segment is subdivided into primary scenes and secondary scenes to derive scene-based forward reshaping functions that minimize the amount of reshaping-related metadata when coding the video segment. When a parent scene of a secondary scene is processed by two or more neighboring nodes, initial forward reshaping functions and trim-pass correction parameters are adjusted using reference tone-mapping functions and updated scene-based trim-pass correction parameters.

Abstract: A projection system and calibration method therefor relate to a light source configured to emit a light in response to an image data, an illumination optical system configured to steer the light, the illumination optical system including a first lens group and a second lens group, a digital micromirror device (DMD) including a plurality of micromirrors respectively configured to reflect the steered light to a predetermined location as on-state light or to reflect the steered light as off-state light to a light dump; determining a deviation between an actual angle of orientation and an expected angle of orientation of a respective micromirror of the plurality of micromirrors; calculating a first amount of lateral adjustment corresponding to the first lens group and a second amount of lateral adjustment corresponding to the second lens group, and actuating the first and second lens groups according to the corresponding first and second amount.

Abstract: Embodiments are described for a method of rendering audio for playback through headphones comprising receiving digital audio content, receiving binaural rendering metadata generated by an authoring tool processing the received digital audio content, receiving playback metadata generated by a playback device, and combining the binaural rendering metadata and playback metadata to optimize playback of the digital audio content through the headphones.

Abstract: The present invention relates to a method for predicting transform coefficients representing frequency content of an adaptive block length media signal, by receiving a frame and receiving block length information indicating a number of quantized transform coefficients for each block in the frame, the number of quantized transform coefficients being one of a first or second number, wherein the first number is greater than the second number, determining a first block has the second number of quantized transform coefficients, converting the first block into a converted block having the first number of quantized transform coefficients, conditioning a main neural network trained to predict at least one output variable given at least one conditioning variable, the at least one conditioning variable being based on information regarding the converted block and block length information for the first block, providing at least one predicted transform coefficients from an output stage of the main neural network.

Abstract: Improved tools for authoring and rendering audio reproduction data are provided. Some such authoring tools allow audio reproduction data to be generalized for a wide variety of reproduction environments. Audio reproduction data may be authored by creating metadata for audio objects. The metadata may be created with reference to speaker zones. During the rendering process, the audio reproduction data may be reproduced according to the reproduction speaker layout of a particular reproduction environment.

Abstract: Systems, methods, and computer program products for audio processing based on convolutional neural network (CNN) are described. The CNN architecture may comprise a multi-scale input block and a multi-scale nested block. The multi-scale input block may be configured to receive input data and to generate a first downsampled input data set by downsampling the input data. The multi-scale nested block may comprise a first encoding layer configured to generate a first encoded data set by performing a convolution based on the input data. The multi-scale nested block may comprise a second encoding layer configured to generate a second encoded data set by performing a convolution based on the first downsampled input data set. Furthermore, the multi-scale nested block may comprise a first convolutional layer configured to generate a first output data set by upsampling the second encoded data set, concatenating the first encoded data set and the upsampled second encoded data set, and performing a convolution.

Abstract: A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

Abstract: The present disclosure relates to a method and system for performing packet loss concealment using a neural network system. The method comprises obtaining a representation of an incomplete audio signal, inputting the representation of the incomplete audio signal to an encoder neural network and outputting a latent representation of a predicted complete audio signal. The latent representation is input to a decoder neural network which outputs a representation of a predicted complete audio signal comprising a reconstruction of the original portion of the complete audio signal, wherein said encoder neural network and said decoder neural network have been trained with an adversarial neural network.

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

Abstract: A system for suppressing noise and enhancing speech and a related method are disclosed. The system trains a neural network model that takes banded energies corresponding to an original noisy waveform and produces a speech value indicating the amount of speech present in each band at each frame. The neural model comprises a feature extraction block that implements some lookahead. The feature extraction block is followed by an encoder with steady down-sampling along the frequency domain forming a contracting path. The encoder is followed by a corresponding decoder with steady up-sampling along the frequency domain forming an expanding path. The decoder receives scaled output feature maps from the encoder at a corresponding level. The decoder is followed by a classification block that generates a speech value indicating an amount of speech present for each frequency band of the plurality of frequency bands at each frame of the plurality of frames.

Abstract: Using a standard-based RGB to YCbCr color transform a new RGB to YCC 3×3 transformation matrix and a 3×1 offset vector are derived under a set of coding-efficiency constraints. The new RGB to YCC 3×3 transform comprises a luminance scaling factor and a 2×2 chroma sub-matrix that preserves the energy of the standard-based RGB to YCbCr transform while maintaining or improving coding efficiency. It also adds support for an authorization or watermarking mechanism in streaming video applications. Examples of using the new color transform using image reshaping are also provided.

Abstract: A global index value is generated for selecting a global reshaping function for an input image of a relatively low dynamic range using luma codewords in the input image. Image filtering is applied to the input image to generate a filtered image. The filtered values of the filtered image provide a measure of local brightness levels in the input image. Local index values are generated for selecting specific local reshaping functions for the input image using the global index value and the filtered values of the filtered image. A reshaped image of a relatively high dynamic range is generated by reshaping the input image with the specific local reshaping functions selected using the local index values.

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: The present invention relates to a method and device for processing a first and a second audio signal representing an input binaural audio signal acquired by a binaural recording device. The present invention further relates to a method for rendering a binaural audio signal on a speaker system. The method for processing a binaural signal comprising extracting audio information from the first audio signal, computing a band gain for reducing noise in the first audio signal and applying the band gains to respective frequency bands of the first audio signal in accordance with a dynamic scaling factor, to provide a first output audio signal. Wherein the dynamic scaling factor has a value between zero and one and is selected so as to reduce quality degradation for the first 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: 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: A deep-learning-based system for performing automated multitrack mixing based on a plurality of input audio tracks is described herein. The system comprises one or more instances of a deep-learning-based first network and one or more instances of a deep-learning-based second network. Particularly, the first network is configured to, based on the 5 input audio tracks, generate parameters for use in the automated multitrack mixing. The second network is configured to, based on the parameters, apply signal processing and at least one mixing gain to the input audio tracks, for generating an output mix of the audio tracks.

Abstract: A method for coding includes; segmenting an image into blocks; grouping blocks into a number of subsets; coding, using an entropy coding module, each subset, by associating digital information with symbols of each block of a subset, including, for the first block of the image, initializing state variables of the coding module; and generating a data sub-stream representative of at least one of the coded subsets of blocks. Where a current block is the first block to be coded of a subset, symbol occurrence probabilities for the first current block are determined based on those for a coded and decoded predetermined block of at least one other subset. Where the current block is the last coded block of the subset: writing, in the sub-stream representative of the subset, the entire the digital information associated with the symbols during coding of the blocks of the subset, and implementing the initializing sub-step.

Abstract: A method of coding at least one image comprising the steps of splitting the image into a plurality of blocks, of grouping said blocks into a predetermined number of subsets of blocks, of coding each of said subsets of blocks in parallel, the blocks of a subset considered being coded according to a predetermined sequential order of traversal. The coding step comprises, for a current block of a subset considered, the sub-step of predictive coding of said current block with respect to at least one previously coded and decoded block, and the sub-step of entropy coding of said current block on the basis of at least one probability of appearance of a symbol.

Abstract: Several embodiments of scalable image processing systems and methods are disclosed herein whereby color management processing of source image data to be displayed on a target display is changed according to varying levels of metadata.

Abstract: Diffuse or spatially large audio objects may be identified for special processing. A decorrelation process may be performed on audio signals corresponding to the large audio objects to produce decorrelated large audio object audio signals. These decorrelated large audio object audio signals may be associated with object locations, which may be stationary or time-varying locations. For example, the decorrelated large audio object audio signals may be rendered to virtual or actual speaker locations. The output of such a rendering process may be input to a scene simplification process. The decorrelation, associating and/or scene simplification processes may be performed prior to a process of encoding the audio data.

Abstract: A projection system and calibration method therefore relate to a light source configured to emit a light in response to an image data, an optical system configured to project the light emitted by the light source; receiving an input associated with a plurality of light values corresponding to a plurality of primary lightfields; converting the input associated with the plurality of light values to a plurality of projector primary color values; determining a gain map based on the plurality Values of projector primary color values; applying the gain map to an image to perform a chromaticity uniformity correction by adjusting levels of the plurality of primary lightfields so that a primary mixture is the same over an image frame, and projecting the image with the optical system in the image frame, wherein the second image is corrected by the gain map.

Abstract: A method of generating a substitution frame for a lost audio frame of an audio signal is presented. The method may comprise determining an audio filter based on samples of a valid audio frame preceding the lost audio frame. The method may comprise generating the substitution frame based on the audio filter and the samples of the valid audio frame preceding the lost audio frame. The method may be advantageously applied to a low frequency effects (LFE) channel of a multi-channel audio signal.

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: In a method to improve backwards compatibility when decoding high-dynamic range images coded in a wide color gamut (WCG) space which may not be compatible with legacy color spaces, hue and/or saturation values of images in an image database are computed for both a legacy color space (say, YCbCr-gamma) and a preferred WCG color space (say, IPT-PQ). Based on a cost function, a reshaped color space is computed so that the distance between the hue values in the legacy color space and rotated hue values in the preferred color space is minimized. HDR images are coded in the reshaped color space. Legacy devices can still decode standard dynamic range images assuming they are coded in the legacy color space, while updated devices can use color reshaping information to decode HDR images in the preferred color space at full dynamic range.

Abstract: A method and apparatus for reconstructing N audio channels from M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal representing the M audio channels and decoding the encoded audio signal to obtain a frequency domain representation of the M audio channels. The method further includes extracting a parameter from the bitstream and reconstructing at least one of the N audio channels using the parameter. The parameter represents an angle between two signals, at least one of which is included in the M audio channels.

Abstract: In a cloud-based system for encoding high dynamic range (HDR) video, each node receives a video segment and bumper frames. Each segment is subdivided into primary scenes and secondary scenes to derive scene-based forward reshaping functions that minimize the amount of reshaping-related metadata when coding the video segment, while maintaining temporal continuity among scenes processed by multiple 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: A novel spatial light modulator (SLM) includes a cover glass, and modulation layer, and a plurality of pixel minors, and separates unwanted, reflected light from desired, modulated light. In one embodiment, a geometrical relationship exists between the cover glass and the pixel minors, such that light that reflects from the cover glass is separated from light that reflects from the pixel minors and is transmitted from the SLM. In one example, one of the cover glass or the pixel minors is angled with respect to the modulation layer. In another example embodiment, the cover glass has a particular thickness, which introduces destructive interference between light that reflects from the top and bottom surfaces of the cover glass. In another embodiment antireflective coatings are disposed between optical interfaces of the SLM. In another embodiment, light from the SLM is directed through an optical filter to remove unwanted light.

Abstract: Methods for dialogue enhancing audio content, comprising providing a first audio signal presentation of the audio components, providing a second audio signal presentation, receiving a set of dialogue estimation parameters configured to enable estimation of dialogue components from the first audio signal presentation, applying said set of dialogue estimation parameters to said first audio signal presentation, to form a dialogue presentation of the dialogue components; and combining the dialogue presentation with said second audio signal presentation to form a dialogue enhanced audio signal presentation for reproduction on the second audio reproduction system, wherein at least one of said first and second audio signal presentation is a binaural audio signal presentation.

Abstract: Described is a method of frame-wise encoding metadata for an input signal, the metadata comprising a plurality of at least partially interrelated parameters calculable from the input signal. The method comprises, for each frame: iteratively performing, by using a looping process, steps of: determining a processing strategy from a plurality of processing strategies for calculating and quantizing the parameters; calculating and quantizing the parameters based on the determined processing strategy to obtain quantized parameters; and encoding the quantized parameters. In particular, each of the plurality of processing strategies comprises a respective first indication indicative of an ordering related to the calculation and quantization of individual parameters; and the processing strategy is determined based on at least one bitrate threshold.

Abstract: A method comprising receiving a first input bit stream for a first parametrically coded input audio signal, the first input bit stream including data representing a first input core audio signal and a first set including at least one spatial parameter relating to the first parametrically coded input audio signal. A first covariance matrix of the first parametrically coded audio signal is determined based on the spatial parameter(s) of the first set. A modified set including at least one spatial parameter is determined based on the determined first covariance matrix, wherein the modified set is different from the first set. An output core audio signal is determined, which is based on, or constituted by, the first input core audio signal. An output bit stream for a parametrically coded output audio signal is generated, the output bit stream including data representing the output core audio signal and the modified set.

Abstract: Embodiments are disclosed for a matrix coded stereo signal with periphonic elements. A mixing matrix, suitable for processing a multi-channel audio input signal, is constructed so that the resulting multi-channel output signal contains the same audio elements from the input signal, wherein the spatial relationships between audio elements, as defined by panning rules associated with the input signal format, are preserved in the output signal, as defined by matrix encoding rules associated with the output signal format. The choice of the coefficients of the mixing matrix is governed by a phase-preference rule that is used to determine the appropriate phase to apply to each input signal channel.