Abstract: An audio signal coding method is provided that includes: obtaining a current frame of an audio signal; obtaining a coding parameter based on a power spectrum ratio of a current frequency in a current frequency area of at least a part of signals of the current frame, where the coding parameter indicates tonal component information of the at least a part of signals, the tonal component information includes at least one of location information of a tonal component, quantity information of tonal components, amplitude information of the tonal component, or energy information of the tonal component, and the power spectrum ratio of the current frequency is a ratio of a value of a power spectrum of the current frequency to a mean value of power spectrums of the current frequency area; and performing bitstream multiplexing on the coding parameter to obtain a coded bitstream.

Abstract: The technology generally relates to spatial audio communication between devices. For example, a first device and a second device may be connected via a communication link. The first device may capture audio signals in an environment through two or more microphones. The first device may encode the captured audio with spatial configuration data. The first device may transmit the encoded audio via the communication link to the second device. The second device may decode the encoded audio into binaural or ambisonic audio to be output by one or more speakers of the second device. The binaural or ambisonic audio may be converted into spatial audio to be output. The second device may output the binaural or spatial audio to create an immersive listening experience.

Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: An audio encoder for encoding audio input data has: a preprocessor for preprocessing the audio input data to obtain audio data to be coded; a coder processor for coding the audio data to be coded; and a controller for controlling the coder processor so that, depending on a first signal characteristic of a first frame of the audio data to be coded, a number of audio data items of the audio data to be coded by the coder processor for the first frame is reduced compared to a second signal characteristic of a second frame, and a first number of information units used for coding the reduced number of audio data items for the first frame is stronger enhanced compared to a second number of information units for the second frame.

Abstract: A vehicle control system includes a transmitting electronic control unit for transmitting multiplexed protocol data units. The transmitting electronic control unit is enabled to transmit first and second protocol data units with corresponding first and second selector field values. A receiving electronic control unit is in electronic communication with the transmitting electronic control unit. The receiving electronic control unit is enabled to be programmed to operate in a first mode of operation or in a second mode of operation. The receiving electronic control unit is enabled to receive and process the first protocol data unit based on receipt of the first selector field value when operating in the first mode of operation and is enabled to receive and process the second protocol data unit based on receipt of the second selector field value when operating in the second mode of operation.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: A time-domain stereo parameter encoding method and a related product are provided. The time-domain stereo parameter encoding method includes: determining a channel combination scheme for a current frame; determining a time-domain stereo parameter of the current frame based on the channel combination scheme for the current frame; and encoding the determined time-domain stereo parameter of the current frame, where the time-domain stereo parameter includes at least one of a channel combination ratio factor or an inter-channel time difference.

Abstract: An audio decoder decodes a bit stream of encoded audio data, which bit stream represents a sequence of audio sample values and includes a plurality of frames, wherein each frame includes associated encoded audio sample values. The audio decoder includes a determiner configured to determine whether a frame of the encoded audio data is a special frame including encoded audio sample values associated with the special frame and additional information, wherein the additional information include encoded audio sample values of a number of frames preceding the special frame, wherein the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, wherein the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame upon start-up of the decoder.

Abstract: A downscaled version of an audio decoding procedure may more effectively and/or at improved compliance maintenance be achieved if the synthesis window used for downscaled audio decoding is a downsampled version of a reference synthesis window involved in the non-downscaled audio decoding procedure by downsampling by the downsampling factor by which the downsampled sampling rate and the original sampling rate deviate, and downsampled using a segmental interpolation in segments of ¼ of the frame length.

Abstract: An embodiment method comprises applying domain transformation processing to a time-series of signal samples, received from a sensor coupled to a dynamical system, to produce a dataset of transformed signal samples therefrom, buffering the transformed signal samples, obtaining a data buffer having transformed signal samples as entries, computing statistical parameters of the data buffer, producing a drift signal indicative of the evolution of the dynamical system as a function of the computed statistical parameters, selecting transformed signal samples buffered in the data buffer as a function of the drift signal, applying normalization processing to the buffered transformed signal samples, applying auto-encoder artificial neural network processing to a dataset of resealed signal samples, and producing a dataset of reconstructed signal samples and calculating an error of reconstruction.

Abstract: A method is described that processes an audio signal. A discontinuity between a filtered past frame and a filtered current frame of the audio signal is removed using linear predictive filtering.

Abstract: There is inter alia disclosed an apparatus for spatial audio encoding which can receive or determine for one or more audio signals (102), spatial audio parameters (106) on a sub band basis for providing spatial audio reproduction, the spatial audio parameters can comprise a coherence value (112) for each sub band of a plurality of subbands (202) of a frame. The apparatus then determines a significance measure for the coherence values (401) of the plurality of sub bands of the frame and uses the significance measure to determine whether to encode (403) the coherence values of the plurality of sub bands of the frame.

Abstract: An audio decoder decodes a bit stream of encoded audio data, which bit stream represents a sequence of audio sample values and includes a plurality of frames, wherein each frame includes associated encoded audio sample values. The audio decoder includes a determiner configured to determine whether a frame of the encoded audio data is a special frame including encoded audio sample values associated with the special frame and additional information, wherein the additional information include encoded audio sample values of a number of frames preceding the special frame, wherein the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, wherein the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame upon start-up of the decoder.

Abstract: An audio encoder for encoding audio input data has: a preprocessor for preprocessing the audio input data to obtain audio data to be coded; a coder processor for coding the audio data to be coded; and a controller for controlling the coder processor so that, depending on a first signal characteristic of a first frame of the audio data to be coded, a number of audio data items of the audio data to be coded by the coder processor for the first frame is reduced compared to a second signal characteristic of a second frame, and a first number of information units used for coding the reduced number of audio data items for the first frame is stronger enhanced compared to a second number of information units for the second frame.

Abstract: A downscaled version of an audio decoding procedure may more effectively and/or at improved compliance maintenance be achieved if the synthesis window used for downscaled audio decoding is a downsampled version of a reference synthesis window involved in the non-downscaled audio decoding procedure by downsampling by the downsampling factor by which the downsampled sampling rate and the original sampling rate deviate, and downsampled using a segmental interpolation in segments of ¼ of the frame length.

Abstract: A downscaled version of an audio decoding procedure may more effectively and/or at improved compliance maintenance be achieved if the synthesis window used for downscaled audio decoding is a downsampled version of a reference synthesis window involved in the non-downscaled audio decoding procedure by downsampling by the downsampling factor by which the downsampled sampling rate and the original sampling rate deviate, and downsampled using a segmental interpolation in segments of ¼ of the frame length.

Abstract: A system and method code an object-based audio signal comprising audio objects in response to audio streams with associated metadata. In the system and method, a metadata processor codes the metadata and generates information about bit-budgets for the coding of the metadata of the audio objects. An encoder codes the audio streams while a bit-budget allocator is responsive to the information about the bit-budgets for the coding of the metadata of the audio objects from the metadata processor to allocate bitrates for the coding of the audio streams by the encoder.

Abstract: An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.

Abstract: A downscaled version of an audio decoding procedure may more effectively and/or at improved compliance maintenance be achieved if the synthesis window used for downscaled audio decoding is a downsampled version of a reference synthesis window involved in the non-downscaled audio decoding procedure by downsampling by the downsampling factor by which the downsampled sampling rate and the original sampling rate deviate, and downsampled using a segmental interpolation in segments of 1/4 of the frame length.

Abstract: Some implementations involve receiving a content stream that includes audio data, receiving at least one type of level adjustment indication relating to playback of the audio data and controlling a level of the input audio data, based on the at least one type of level adjustment indication, to produce level-adjusted audio data. Some examples involve determining, based at least in part on the type(s) of level adjustment indication, a multiband limiter configuration, applying the multiband limiter to the level-adjusted audio data, to produce multiband limited audio data and providing the multiband limited audio data to one or more audio reproduction transducers of an audio environment.

Abstract: Methods and apparatus provide digital media with spatial audio to a blockchain. The blockchain network executes a decentralized application (Dapp) with a user interface (UI) that enables a user to select audio for spatialization and uploading to the blockchain. The spatial audio transmits to the blockchain network to reduce processing and transmission of network data.

Abstract: A method of processing speech includes: providing a first set of audio data having audio features in a first bandwidth; down-sampling the first set of audio data to a second bandwidth lower than the first bandwidth; producing, by a high frequency reconstruction network (HFRN), an estimate of audio features in the first bandwidth for the first set of audio data, based on at least the down-sampled audio data; inputting, into the HFRN, a second set of audio data having audio features in the second bandwidth; producing, by the HFRN, based on a second set of audio data having audio features in the second bandwidth, an estimate of audio features in the first bandwidth for the second set of audio data; and training a speech processing system (SPS) using the estimates of audio features in the first bandwidth for the first and second sets of audio data.

Abstract: Systems and processes for operating an intelligent automated assistant are provided. An example process for determining user intent includes receiving a natural language input and detecting an event. The process further includes, determining, at a first time, based on the natural language input, a first value for a first node of a parsing structure; and determining, at a second time, based on the detected data event, a second value for a second node of the parsing structure. The process further includes in accordance with a determination that the first time and the second time are within the predetermined time: determining, using the parsing structure, the first value, and the second value, a user intent associated with the natural language input; initiating a task based on the determined intent; and providing an output indicative of the task.

Abstract: A method for encoding an input audio stream including the steps of obtaining a first playback stream presentation of the input audio stream intended for reproduction on a first audio reproduction system, obtaining a second playback stream presentation of the input audio stream intended for reproduction on a second audio reproduction system, determining a set of transform parameters suitable for transforming an intermediate playback stream presentation to an approximation of the second playback stream presentation, wherein the transform parameters are determined by minimization of a measure of a difference between the approximation of the second playback stream presentation and the second playback stream presentation, and encoding the first playback stream presentation and the set of transform parameters for transmission to a decoder.

Abstract: An apparatus for encoding directional audio coding parameters comprising diffuseness parameters and direction parameters having a parameter calculator (100) for calculating the diffuseness parameters with a first time or frequency resolution and for calculating the direction parameters with a second time or frequency resolution; and a quantizer and encoder processor (200) for generating a quantized and encoded representation of the diffuseness parameters and the direction parameters.

Abstract: A method for spatial audio signal encoding comprising: obtaining a plurality of audio direction parameters, wherein each parameter comprises an elevation value and an azimuth value and wherein each parameter has an ordered position; deriving for each of the plurality of audio direction parameters a corresponding derived audio direction parameter (SP) comprising an elevation and an azimuth value, corresponding derived audio direction parameters (SP) being arranged in a manner determined by a spatial utilization defined by the elevation values and the azimuth values of the plurality of audio direction parameters; rotating each derived audio direction parameter (SP) by the azimuth value (?0) of an audio direction parameter in the first position of the plurality of audio direction parameters and quantizing the rotation to determine for each a corresponding quantized rotated derived audio direction parameter; changing the ordered position of an audio direction parameter to a further position coinciding with a posi

Abstract: Disclosed is a dialogue emotion correction method based on a graph neural network, including: extracting acoustic features, text features, and image features from a video file to fuse them into multi-modal features; obtaining an emotion prediction result of each sentence of a dialogue in the video file by using the multi-modal features; fusing the emotion prediction result of each sentence with interaction information between talkers in the video file to obtain interaction information fused emotion features; combining, on the basis of the interaction information fused emotion features, with context-dependence relationship in the dialogue to obtain time-series information fused emotion features; correcting, by using the time-series information fused emotion features, the emotion prediction result of each sentence that is obtained previously as to obtain a more accurate emotion recognition result.

Abstract: An audio decoder decodes a bit stream of encoded audio data, which bit stream represents a sequence of audio sample values and includes a plurality of frames, wherein each frame includes associated encoded audio sample values. The audio decoder includes a determiner configured to determine whether a frame of the encoded audio data is a special frame including encoded audio sample values associated with the special frame and additional information, wherein the additional information include encoded audio sample values of a number of frames preceding the special frame, wherein the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, wherein the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame upon start-up of the decoder.

Abstract: An audio decoder decodes a bit stream of encoded audio data, which bit stream represents a sequence of audio sample values and includes a plurality of frames, wherein each frame includes associated encoded audio sample values. The audio decoder includes a determiner configured to determine whether a frame of the encoded audio data is a special frame including encoded audio sample values associated with the special frame and additional information, wherein the additional information include encoded audio sample values of a number of frames preceding the special frame, wherein the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, wherein the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame upon start-up of the decoder.

Abstract: Embodiments are disclosed for channel-based audio (CBA) (e.g., 22.2-ch audio) to object-based audio (OBA) conversion. The conversion includes converting CBA metadata to object audio metadata (OAMD) and reordering the CBA channels based on channel shuffle information derived in accordance with channel ordering constraints of the OAMD. The OBA with reordered channels is rendered in a playback device using the OAMD or in a source device, such as a set-top box or audio/video recorder. In an embodiment, the CBA metadata includes signaling that indicates a specific OAMD representation to be used in the conversion of the metadata. In an embodiment, pre-computed OAMD is transmitted in a native audio bitstream (e.g., AAC) for transmission (e.g., over HDMI) or for rendering in a source device. In an embodiment, pre-computed OAMD is transmitted in a transport layer bitstream (e.g., ISO BMFF, MPEG4 audio bitstream) to a playback device or source device.

Abstract: An encoder for encoding a parametric spectral representation (f) of auto-regressive coefficients that partially represent an audio signal. The encoder includes a low-frequency encoder configured to quantize elements of a part of the parametric spectral representation that correspond to a low-frequency part of the audio signal. It also includes a high-frequency encoder configured to encode a high-frequency part (fH) of the parametric spectral representation (f) by weighted averaging based on the quantized elements ({circumflex over (f)}L) flipped around a quantized mirroring frequency ({circumflex over (f)}m), which separates the low-frequency part from the high-frequency part, and a frequency grid determined from a frequency grid codebook in a closed-loop search procedure. Described are also a corresponding decoder, corresponding encoding/decoding methods and UEs including such an encoder/decoder.

Abstract: In a stereo encoding method, a channel combination encoding solution of a current frame is first obtained, and then a quantized channel combination ratio factor of the current frame and an encoding index of the quantized channel combination ratio factor are obtained based on the obtained channel combination encoding solution, so that an obtained primary channel signal and secondary channel signal of the current frame meet a characteristic of the current frame.

Abstract: An audio decoder decodes a bit stream of encoded audio data, which bit stream represents a sequence of audio sample values and includes a plurality of frames, wherein each frame includes associated encoded audio sample values. The audio decoder includes a determiner configured to determine whether a frame of the encoded audio data is a special frame including encoded audio sample values associated with the special frame and additional information, wherein the additional information include encoded audio sample values of a number of frames preceding the special frame, wherein the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, wherein the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame upon start-up of the decoder.

Abstract: Systems and methods of the present disclosure are directed toward digital signal processing using machine-learned differentiable digital signal processors. For example, embodiments of the present disclosure may include differentiable digital signal processors within the training loop of a machine-learned model (e.g., for gradient-based training). Advantageously, systems and methods of the present disclosure provide high quality signal processing using smaller models than prior systems, thereby reducing energy costs (e.g., storage and/or processing costs) associated with performing digital signal processing.

Abstract: Embodiments of the disclosure provide methods and systems for processing a neural network associated with an input matrix having a first number of elements. The method can include: dividing the input matrix into a plurality of vectors, each vector having a second number of elements; grouping the plurality of vectors into a first group of vectors and a second group of vectors; and pruning the first group of vectors and the second group of vectors.

Abstract: Implementations set forth herein relate providing and/or detecting an automated assistant suppressing output for limiting responsiveness of an automated assistant that is available via another computing device. The output can be rendered as an inaudible tone in an environment, in order to pre-emptively suppress responses from automated assistants that may detect a forthcoming spoken utterance from a user. Limiting responsiveness in this way can preserve computational resources in environments, such as vehicles, in which multiple users may be carrying devices that are capable of being invoked via a spoken utterance. For instance, pre-emptively limiting responsiveness of other automated assistants can eliminate interruptions at devices that may other be inadvertently invoked as a result of detecting certain spoken utterances.

Abstract: In general, techniques are described for psychoacoustic audio coding of ambisonic audio data. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may store the bitstream that includes an encoded audio object and a corresponding spatial component that defines spatial characteristics of the encoded foreground audio signal. The encoded foreground audio signal may include a coded gain and a coded shape. The one or more processors may perform a gain and shape synthesis with respect to the coded gain and the coded shape to obtain a foreground audio signal, and reconstruct, based on the foreground audio signal and the spatial component, the ambisonic audio data.

Abstract: An audio coding method and apparatus are provided. The audio coding method includes: obtaining first audio data; obtaining a target bit rate and a Bluetooth packet type, where the target bit rate and the Bluetooth packet type correspond to a current status of a Bluetooth channel; obtaining one or more of a bit pool parameter set, a psychoacoustic parameter set, and a spectrum bandwidth parameter set by using a neural network obtained through pre-training based on the first audio data, the target bit rate, and the Bluetooth packet type; and coding the first audio data based on one or more of the bit pool parameter set, the psychoacoustic parameter set, and the spectrum bandwidth parameter set to obtain a to-be-sent bit stream. The status of the Bluetooth channel can be adaptively matched, and continuous audio listening experience is provided when audio quality is maximally ensured.

Abstract: A Bluetooth communication method is disclosed, and relates to the field of short-range wireless communications technologies. The method includes: A terminal receives a play operation performed by a user on first audio data. The terminal sends first indication information to a Bluetooth device when a service type of the first audio data is a first service type, where the first indication information is used by the Bluetooth device to set a buffer time length for the audio data to first duration. The Bluetooth device receives the first audio data sent by the terminal via Bluetooth, and buffers the first audio data. The Bluetooth device starts to play the buffered first audio data when the buffer time length for the first audio data reaches the first duration. In this way, the terminal can perform targeted delay control on audio data playback of the Bluetooth device in different application scenarios.

Abstract: An audio processing system reduces perception of audible artifacts due to changes in an element in an audio channel of the audio processing system. The system reproduces an audio input signal and produces an audio output signal with the audio channel. The channel has an adjustable or selectable element that, responsive to a control signal, changes a characteristic of the audio processing channel, which generates a transient in the audio output signal. The systems include a level detector for measuring a signal level of the audio input signal and a controller responsive to an output of the level detector to determine a masking time interval available from the audio output signal due to signal content in the audio input signal. The controller generates the control signal to change the characteristic of the audio processing channel so that at least a portion of the transient occurs in the masking time interval.

Abstract: An audio encoder for encoding audio input data has: a preprocessor for preprocessing the audio input data to obtain audio data to be coded; a coder processor for coding the audio data to be coded; and a controller for controlling the coder processor so that, depending on a first signal characteristic of a first frame of the audio data to be coded, a number of audio data items of the audio data to be coded by the coder processor for the first frame is reduced compared to a second signal characteristic of a second frame, and a first number of information units used for coding the reduced number of audio data items for the first frame is stronger enhanced compared to a second number of information units for the second frame.

Abstract: Disclosed are system (100) and method (300) for optimizing data transmission in communication network. System comprises Internet of Thing (IoT) device node (102) having sensors for capturing sensor data and controller (106) configured at the IoT device node for constructing set of attributes from the sensor data, such that each set of attribute comprises sensor data based on correlation. Support Vector Regression (SVR) models for pruning the sensor data at IoT device node (102) is defined. Values for each attribute in set of attributes predicted based on predicting run-time errors in values according to the SVR models defined for each of base attribute and the non-base attribute. The run-time errors for each of base attributes and non-base attributes gets compared with threshold error values. The IoT device node (102) transmits either model parameters for attributes or training data along with the set of attributes by transceiver (108) to the data collector node (104).

Abstract: The present document describes a method (500) for generating a bitstream (101), wherein the bitstream (101) comprises a sequence of superframes (400) for a sequence of frames of an immersive audio signal (111). The method (500) comprises, repeatedly for the sequence of superframes (400), inserting (501) coded audio data (206) for one or more frames of one or more downmix channel signals (203) derived from the immersive audio signal (111), into data fields (411, 421, 412, 422) of a superframe (400); and inserting (502) metadata (202, 205) for reconstructing one or more frames of the immersive audio signal (111) from the coded audio data (206), into a metadata field (403) of the superframe (400).

Abstract: A method is described that processes an audio signal. A discontinuity between a filtered past frame and a filtered current frame of the audio signal is removed using linear predictive filtering.

Abstract: Audio encoder for encoding audio input data to obtain audio output data includes an input interface for receiving a plurality of audio channels, a plurality of audio objects and metadata related to one or more of the plurality of audio objects; a mixer for mixing the plurality of objects and the plurality of channels to obtain a plurality of pre-mixed channels, each pre-mixed channel including audio data of a channel and audio data of at least one object; a core encoder for core encoding core encoder input data; and a metadata compressor for compressing the metadata related to the one or more of the plurality of audio objects, wherein the audio encoder is configured to operate in at least one mode of the group of two modes.

Abstract: A method of generating a residual signal performed by an encoder includes identifying an input signal including an audio sample, generating a first residual signal from the input signal using linear predictive coding (LPC), generating a second residual signal having a less information amount than the first residual signal by transforming the first residual signal, transforming the second residual signal into a frequency domain, and generating a third residual signal having a less information amount than the second residual signal from the transformed second residual signal using frequency-domain prediction (FDP) coding.

Abstract: Systems, methods, and apparatuses to restore degraded speech via a modified diffusion model are described. An exemplary system is specially configured to train a diffusion-based vocoder containing an upsampler, based on pairing original speech x and degraded speech mel-spectrum mT samples; train a deep convoluted neural network (CNN) upsampler based on a mean absolute error loss to match the estimated original speech {circumflex over (x)}? outputted by the diffusion-based vocoder by extracting the upsampler, generating a reference conditioner, and generating a weighted altered conditioner cTn?. The system further optimizes speech quality to invert non-linear transformation and estimate lost data by feeding the degraded mel-spectrum mT through the CNN upsampler and feeding the degraded mel-spectrum mT through the diffusion-based vocoder. The system then generates estimated original speech {circumflex over (x)}? based on the corresponding degraded speech mel-spectrum mT. Other related embodiments are described.

Abstract: An example audio decoding device includes a memory configured to store at least a portion of a coded audio bitstream; and one or more processors configured to: decode, based on the coded audio bitstream, a representation of a soundfield; decode, based on the coded audio bitstream, a syntax element indicating a selection of either a head-related transfer function (HRTF) or a binaural room impulse response (BRIR); and render, using the selected HRTF or BRIR, speaker feeds from the soundfield.

Abstract: To enable a set of media access control to be favorably performed on the reception side. A container having a predetermined format is transmitted, the container including a media stream. A predetermined number of pieces of media access information associated for the set of media access control, are sequentially inserted into a layer of the media stream or a layer of the container. For example, the media access information includes identification information for making a distinction from different media access information and identification information for making an association with the different media access information.

Abstract: The present document relates to a method of layered encoding of a compressed sound representation of a sound or sound field. The compressed sound representation comprises a basic compressed sound representation comprising a plurality of components, basic side information for decoding the basic compressed sound representation to a basic reconstructed sound representation of the sound or sound field, and enhancement side information including parameters for improving the basic reconstructed sound representation.