Abstract: An audio session management method may involve: determining, by an audio session manager, one or more first media engine capabilities of a first media engine of a first smart audio device, the first media engine being configured for managing one or more audio media streams received by the first smart audio device and for performing first smart audio device signal processing for the one or more audio media streams according to a first media engine sample clock; receiving, by the audio session manager and via a first application communication link, first application control signals from the first application; and controlling the first smart audio device according to the first media engine capabilities, by the audio session manager, via first audio session management control signals transmitted to the first smart audio device via a first smart audio device communication link and without reference to the first media engine sample clock.

Abstract: Methods and systems for performing at least one audio activity (e.g., conducting a phone call or playing music or other audio content) in an environment including by determining an estimated location of a user in the environment in response to sound uttered by the user (e.g., a voice command), and controlling the audio activity in response to determining the estimated user location. The environment may have zones which are indicated by a zone map and estimation of the user location may include estimating in which of the zones the user is located. The audio activity may be performed using microphones and loudspeakers which are implemented in or coupled to smart audio devices.

Abstract: Methods are disclosed for adaptive display management using one or more viewing environment parameters. Given the one or more viewing environment parameters, an effective luminance range for a target display, and an input image, a tone-mapped image is generated based on a tone-mapping curve, an original PQ luminance mapping function, and the effective luminance range of the display. Corrected PQ (PQ?) luminance mapping functions are generated according to the viewing environment parameters and, optionally, the transmissivity properties and reflectivity properties of the target display.

Abstract: The present disclosure relates to a method and audio processing system for performing dynamic range adjustment of spatial audio objects. The method comprises obtaining (step S1) a plurality of spatial audio objects (10), obtaining (step S2) at least one rendered audio presentation of the spatial audio objects (10) and determining (step S3) signal level data associated with each presentation audio channel in said set of presentation audio channels. The method further comprises obtaining (step S31) a threshold value and, for each time segment, selecting (step S4) a selected presentation audio channel which is associated with a highest or a lowest signal level, determining (step S5) a gain based on the threshold value and the representation of the signal level of the selected audio channel, and applying (step S6) the gain of each time segment to corresponding time segments of the spatial audio objects.

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. The high frequency regeneration is performed as a post-processing operation with a delay of 3010 samples per audio channel.

Abstract: A computing device system including a computing device having a housing and electronic components disposed within the housing, where the electronic components include a controller, a memory, and a power source. A display screen is supported on the housing, and a socket extends into the housing. A removable speaker is selectively received within the socket, where the removable speaker includes a power source that is automatically charged when received within the socket.

Abstract: Methods, systems, and bitstream syntax are described for the entropy modeling of latent features in image and video coding using a combination of probability density functions. Using high-level syntax elements, an encoder may signal to compliant decoders the multi-distribution entropy model using: the number of one or more PDFs being used, an identifier of each PDF being used among a list of available PDFs, the number of PDF parameters in each PDF, and syntax elements indicating which PDF parameters across two or more PDFs being used are being shared.

Abstract: Described is a method of audio processing in a HbbTV terminal device. The method includes receiving a decoded broadcast feed including a first audio track, receiving HbbTV content relating to the broadcast feed, the HbbTV content including a second audio track, extracting level-related information from the decoded broadcast feed, wherein the level-related information is embedded in the decoded broadcast feed and enables to obtain an indication of an original audio level of the first audio track, analyzing the first audio track for determining an actual audio level of the first audio track, determining a gain factor based on the actual audio level and the original audio level, and generating a third audio track for output by the HbbTV terminal device based on the first audio track, the second audio track, and the gain factor. Also described is an apparatus for carrying out the method, as well as corresponding programs and computer-readable storage media.

Abstract: Embodiments are disclosed for speaker diarization supporting episodical content. In an embodiment, a method comprises: receiving media data including one or more utterances; dividing the media data into a plurality of blocks; identifying segments of each block of the plurality of blocks associated with a single speaker; extracting embeddings for the identified segments in accordance with a machine learning model, wherein extracting embeddings for identified segments further comprises statistically combining extracted embeddings for identified segments that correspond to a respective continuous utterance associated with a single speaker; clustering the embeddings for the identified segments into clusters; and assigning a speaker label to each of the embeddings for the identified segments in accordance with a result of the clustering. In some embodiments, a voiceprint is used to identify a speaker and the speaker identity for a speaker label.

Abstract: Described is a method of processing an audio signal. The method includes a first step for applying enhancement to a first component of the audio signal and/or applying suppression to a second component of the audio signal relative to the first component, and a second step of modifying an output of the first step by applying a deep learning based model to the output of the first step, for perceptually improving the first component of the audio signal. Also described is an apparatus for carrying out the method, as well as corresponding programs and computer-readable storage media.

Abstract: A method for encoding envelope information is provided. In some implementations, the method involves determining a first downmixed signal associated with a downmixed channel associated with an audio signal to be encoded. In some implementations, the method involves determining energy levels of the first downmixed signal for a plurality of frequency bands. In some implementations, the method involves determining whether to encode information indicative of the energy levels in a bitstream. In some implementations, the method involves encoding the determined energy levels. In some implementations, the method involves generating an energy control value indicating that energy levels are encoded. In some implementations, the method involves generating the bitstream, wherein the energy control value and the information indicative of the energy levels are usable by the decoder to adjust energy levels associated with the first downmixed signal.

Abstract: A projection system includes a light source configured to emit a light in response to an image data, a phase light modulator configured to receive the light from the light source and to apply a spatially-varying phase modulation on the light, thereby generating a projection light and steering the light on a reconstruction field, wherein the reconstruction field is a complex plane on which a reconstruction image is formed, and a controller configured to control the light source, control the phase light modulator, initialize (401) the reconstruction field to an initial value, and iteratively for each of a plurality of subframes within a frame of the image data: set (402) the reconstruction field to the initial value for the first iteration or set (402) the reconstruction field to a subsequent-iteration reconstruction field value for any subsequent-iteration, map (403) the reconstruction field to a modulation field, wherein the modulation field is a complex plane of the phase light modulator which modulates a ph

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: A method for decoding an encoded audio bitstream in an audio processing system is disclosed. The method includes extracting from the encoded audio bitstream a first waveform-coded signal comprising spectral coefficients corresponding to frequencies up to a first cross-over frequency for a time frame and performing parametric decoding at a second cross-over frequency for the time frame to generate a reconstructed signal. The second cross-over frequency is above the first cross-over frequency and the parametric decoding uses reconstruction parameters derived from the encoded audio bitstream to generate the reconstructed signal. The method also includes extracting from the encoded audio bitstream a second waveform-coded signal comprising spectral coefficients corresponding to a subset of frequencies above the first cross-over frequency for the time frame and interleaving the second waveform-coded signal with the reconstructed signal to produce an interleaved signal for the time frame.

Abstract: A method for performing calibration of magnetometers is provided. In some embodiments, the method involves obtaining a sequence of gyroscope measurements from one or more gyroscopes and a sequence of magnetometer measurements from one or more magnetometers. In some embodiments, the method involves determining a sequence of angular velocity estimates based on the sequence of gyroscope measurements. In some embodiments, the method involves determining a first estimate of a derivative of an external magnetic field based on the sequence of magnetometer measurements. In some embodiments, the method involves determining a second estimate of the derivative of the external magnetic field based on the sequence of angular velocity estimates. In some embodiments, the method involves identifying magnetometer calibration constants based on a difference between the first estimate of the derivative and the second estimate of the derivative.

Abstract: A method for performing gain control on audio signals is provided. In some implementations, the method involves determining downmixed signals associated with one or more downmix channels associated with a current frame of an audio signal to be encoded. In some implementations, the method involves determining whether an overload condition exists for an encoder. In some implementation, the method involves determining a gain parameter. In some implementations, the method involves determining at least one gain transition function based on the gain parameter and a gain parameter associated with a preceding frame of the audio signal. In some implementations, the method involves applying the at least one gain transition function to one or more of the downmixed signals. In some implementations, the method involves encoding the downmixed signals in connection with information indicative of gain control applied to the current frame.

Abstract: The present disclosure provides methods, devices and computer program products for non-uniform quantization of parameters. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system taking the non-uniformly quantized parameters into account. According to the disclosure, such an approach renders it possible to reduce bit consumption without substantially reducing the quality of the reconstructed audio object.

Abstract: Sampled data is packaged in checkerboard format for encoding and decoding. The sampled data may be quincunx sampled multi-image video data (e.g., 3D video or a multi-program stream), and the data may also be divided into sub-images of each image which are then multiplexed, or interleaved, in frames of a video stream to be encoded and then decoded using a standardized video encoder. A system for viewing may utilize a standard video decoder and a formatting device that de-interleaves the decoded sub-images of each frame reformats the images for a display device. A 3D video may be encoded using a most advantageous interleaving format such that a preferred quality and compression ratio is reached. In one embodiment, the invention includes a display device that accepts data in multiple formats.

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: 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: A volumetric image of a scene can be created, in one embodiment, by recording, through a camera in a device, a series of images of the scene as the camera is moved along a path relative to the scene; during the recording, the device stores motion path metadata about the path, and the series of images is associated with the motion path metadata and a metadata label is associated with the series of images, the metadata label indicating that the recorded series of images represent a volumetric image of the scene. The series of images, the motion path metadata and the metadata label can be assembled into a package for distribution to devices that can view the volumetric image, which may be referred to as a limited volumetric image. The devices that receive the volumetric image can display the individual images in the series of images or as a video.

Abstract: A method for delivering media to a playback device including outputting first test media to be viewed by a first user. The method further includes receiving a first user input related to a first perception of the first test media by the first user and indicating a first personalized quality of experience of the first user with respect to the first test media. The method further includes generating a first personalized sensitivity profile including one or more viewing characteristics of the first user based on the first user input, and determining, based at least in part on the first personalized sensitivity profile, a first media parameter. The first media parameter is determined in order to increase an efficiency of media delivery to the first playback device over a network while preserving the first personalized quality of experience of the first user.

Abstract: Disclosed is a portable computing device (1) comprising a keyboard (13) and an acoustic transducer (16, 17), the keyboard (13) comprising a key (14, 15), wherein the acoustic transducer (16, 17) is placed in the key (14, 15), and wherein the key (14, 15) comprises a sound port (150) allowing sound generated by the transducer (16, 17) to propagate.

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: Embodiments are disclosed for context aware soundscape control. In an embodiment, an audio processing method comprises: capturing, using a first set of microphones on a mobile device, a first audio signal from an audio scene; capturing, using a second set of microphones on a pair of earbuds, a second audio signal from the audio scene; capturing, using a camera on the mobile device, a video signal from a video scene; generating, with at least one processor, a processed audio signal from the first audio signal and the second audio signal, the processed audio signal generated with adaptive soundscape control based on context information; and combining, with the at least one processor, the processed audio signal and the captured video signal as multimedia output.

Abstract: A method (700) for rendering an audio signal of an audio source (211, 212, 213) in a virtual reality rendering environment (180) is described. The method (700) comprises determining (701) whether or not a directivity pattern (232) of the audio source (211, 212, 213) is to be taken into account for a listening situation of a listener (181) within the virtual reality rendering environment (180). Furthermore, the method (700) comprises rendering (702) an audio signal of the audio source (211, 212, 213) without taking into account the directivity pattern (232) of the audio source (211, 212, 213), if it is determined that the directivity pattern (232) of the audio source (211, 212, 213) is not to be taken into account for the listening situation of the listener (181).

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.

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: 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: Methods and systems for frame rate scalability are described. Support is provided for input and output video sequences with variable frame rate and variable shutter angle across scenes, or for input video sequences with fixed input frame rate and input shutter angle, but allowing a decoder to generate a video output at a different output frame rate and shutter angle than the corresponding input values. Techniques allowing a decoder to decode more computationally-efficiently a specific backward compatible target frame rate and shutter angle among those allowed are also presented.

Abstract: Methods and systems for frame rate scalability are described. Support is provided for input and output video sequences with variable frame rate and variable shutter angle across scenes, or for input video sequences with fixed input frame rate and input shutter angle, but allowing a decoder to generate a video output at a different output frame rate and shutter angle than the corresponding input values. Techniques allowing a decoder to decode more computationally-efficiently a specific backward compatible target frame rate and shutter angle among those allowed are also presented.

Abstract: A handheld imaging device has a data receiver that is configured to receive reference encoded image data. The data includes reference code values, which are encoded by an external coding system. The reference code values represent reference gray levels, which are being selected using a reference grayscale display function that is based on perceptual non-linearity of human vision adapted at different light levels to spatial frequencies. The imaging device also has a data converter that is configured to access a code mapping between the reference code values and device-specific code values of the imaging device. The device-specific code values are configured to produce gray levels that are specific to the imaging device. Based on the code mapping, the data converter is configured to transcode the reference encoded image data into device-specific image data, which is encoded with the device-specific code values.

Abstract: A method and apparatus for decompressing a Higher Order Ambisonics (HOA) signal representation is disclosed. The apparatus includes an input interface that receives an encoded directional signal and an encoded ambient signal and an audio decoder that perceptually decodes the encoded directional signal and encoded ambient signal to produce a decoded directional signal and a decoded ambient signal, respectively. The apparatus further includes an extractor for obtaining side information related to the directional signal and an inverse transformer for converting the decoded ambient signal from a spatial domain to an HOA domain representation of the ambient signal. The apparatus also includes a synthesizer for recomposing a Higher Order Ambisonics (HOA) signal from the HOA domain representation of the ambient signal and the decoded directional signal. The side information includes a direction of the directional signal selected from a set of uniformly spaced directions.

Abstract: The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel on the basis of the first and second frequency-domain representations of the first input channel, the first frequency-domain representation of the second input channel and a complex prediction coefficient. The upmixing can be suspended responsive to control data.

Abstract: A projection system for etendue utilization includes a first light source configured to emit a light, the light including a first etendue component and a second etendue component, wherein the first etendue component has a lower etendue than the second etendue component, a first projection optics configured to project a first image on a screen, a second projection optics configured to project a second image on the screen, and an etendue splitter component. The etendue splitter component is configured to receive the light from the light source, extract, from the light, the first etendue component and the second etendue component, provide the first etendue component to the first projection optics, and provide the second etendue component to the second projection optics.

Abstract: The present document relates to audio coding systems. In particular, the present document relates to efficient methods and systems for parametric multi-channel audio coding. An audio encoding system configured to generate a bitstream indicative of a downmix signal and spatial metadata for generating a multi-channel upmix signal from the downmix signal is described. The system comprises a downmix processing unit configured to generate the downmix signal from a multi-channel input signal; wherein the downmix signal comprises m channels and wherein the multi-channel input signal comprises n channels; n, m being integers with m<n. Furthermore, the system comprises a parameter processing unit configured to determine the spatial metadata from the multi-channel input signal.

Abstract: Systems, methods, and computer program products implementing a sensor data prediction algorithm are disclosed. An example method comprises receiving motion data representing motions of a head-mounted listening device; transforming the motion data into quaternion domain; predicting, by one or more processors, future motions of the head-mounted listening device, the predicting including creating angular acceleration data from the transformed motion data and applying one or more smoothing filters to the angular acceleration data, the predicted future motions including rotation angles around corresponding axes in the quaternion domain; and providing the predicted future motions of the head-mounted listening device to a processor for adjusting a sound field presented by the listening device such that the sound field follows predicted movements of the head-mounted listening device.

Abstract: Methods and systems for improving signal processing by smoothing the covariance matrix of a multi-channel signal by setting a forgetting factor based on the bins of a band. A method and system for resetting the smoothing based on transient detection is also disclosed. A method and system for resampling for the smoothing during a banding transition is also disclosed.

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 method for decoding an intra-picture prediction mode includes the steps of: determining whether the intra-picture prediction mode of a current prediction unit is identical to a first intra-picture prediction mode candidate or a second intra-picture prediction mode candidate based on bit information: and when the intra-picture prediction mode of the current prediction unit is identical to the first intra-picture prediction mode candidate and/or to the second intra-picture prediction mode candidate, determining whether the first intra-picture prediction mode candidate or the second intra-picture prediction mode candidate is identical to the intra-picture prediction mode of the current prediction unit on the basis of additional bit information, and decoding the intra-picture prediction mode of the current prediction unit.

Abstract: Described herein is an audio decoder for decoding a bitstream of encoded audio data, wherein the bitstream of encoded audio data represents a sequence of audio sample values and comprises a plurality of frames, wherein each frame comprises associated encoded audio sample values, the audio decoder comprising: a determiner configured to determine whether a frame of the bitstream of encoded audio data is an immediate playout frame comprising encoded audio sample values associated with a current frame and additional information; and an initializer configured to initialize the decoder if the determiner determines that the frame is an immediate playout frame, wherein initializing the decoder comprises decoding the encoded audio sample values comprised by the additional information before decoding the encoded audio sample values associated with the current frame.

Abstract: Sampled data is packaged in checkerboard format for encoding and decoding. The sampled data may be quincunx sampled multi-image video data (e.g., 3D video or a multi-program stream), and the data may also be divided into sub-images of each image which are then multiplexed, or interleaved, in frames of a video stream to be encoded and then decoded using a standardized video encoder. A system for viewing may utilize a standard video decoder and a formatting device that de-interleaves the decoded sub-images of each frame reformats the images for a display device. A 3D video may be encoded using a most advantageous interleaving format such that a preferred quality and compression ratio is reached. In one embodiment, the invention includes a display device that accepts data in multiple formats.

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

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

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

Abstract: An audio session management method may involve: determining, by an audio session manager, one or more first media engine capabilities of a first media engine of a first smart audio device, the first media engine being configured for managing one or more audio media streams received by the first smart audio device and for performing first smart audio device signal processing for the one or more audio media streams according to a first media engine sample clock; receiving, by the audio session manager and via a first application communication link, first application control signals from the first application; and controlling the first smart audio device according to the first media engine capabilities, by the audio session manager, via first audio session management control signals transmitted to the first smart audio device via a first smart audio device communication link and without reference to the first media engine sample clock.

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

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

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

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