Abstract: A player device includes a multi-media receiving device and a signal processing device. The signal processing device performs a mixing operation to mix a multi-media signal and a pseudo stereo signal to generate a mixed multi-media signal. The signal processing device further converts a system sound from a mono sound signal into the pseudo stereo signal by selecting at least one frequency component of the system sound as a modulation signal and combining a delayed version of the modulation signal and remaining frequency components of the system sound to generate the pseudo stereo signal. In the mixing operation, the signal processing device combines a multi-media sound component of a first channel and a system sound component of the first channel and combines a multi-media sound component of a second channel and a system sound component of the second channel to generate the mixed multimedia signal.

Abstract: A second media rendering device in a network is synchronized with a first media rendering device in the network. At a first time s1, the second media rendering device sends a request for a current audio sample index from the first media rendering device. The first media rendering device receives the request for the current audio sample index from the second media rendering device. The first media rendering device sends a message including a first current audio sample index equal to a value for a first media rendering device internal sample index. At a second time s2 the second media rendering device receives the first current sample index value from the first media rendering device. The second media rendering device determines a transmission latency, and estimates an updated first media rendering device internal sample index value.

Abstract: A multi-channel audio decoder for providing at least two output audio signals on the basis of an encoded representation is configured to render a plurality of decoded audio signals, which are obtained on the basis of the encoded representation, in dependence on one or more rendering parameters, to obtain a plurality of rendered audio signals. The multi-channel audio decoder is configured to derive one or more decorrelated audio signals from the rendered audio signals, and to combine the rendered audio signals, or a scaled version thereof, with the one or more decorrelated audio signals, to obtain the output audio signals. A multi-channel audio encoder provides a decorrelation method parameter to control an audio decoder.

Abstract: This encoding device comprises: a downmix circuit that switches mixing processing according to the characteristic of an input stereo signal to generate either a first stereo signal or a second stereo signal obtained by mixing processing of a left channel signal and a right channel signal; a first encoding circuit that encodes the first stereo signal; and a second encoding circuit that encodes two signals included in the second stereo signal. The second encoding circuit performs monaural encoding on the basis of the encoding mode of the first encoding circuit in a first section in which switching from the first stereo signal to the second stereo signal is performed and/or a second section in which switching from the second stereo signal to the first stereo signal is performed.

Abstract: Provided are a display device capable of processing a plurality of voice commands requested from a plurality of wireless sound devices, and an operating method thereof. The display device includes: a communication unit communicatively connected to a plurality of wireless sound devices; a voice acquisition unit configured to acquire a voice command requested from the plurality of wireless sound devices; and a processor configured to process the voice command, wherein the processor is configured to: perform communication connection to the plurality of wireless sound devices; when a voice transmission request is received from the plurality of wireless sound devices, identify the wireless sound devices based on addresses of the wireless sound devices; and determine a voice command processing sequence for the plurality of wireless sound devices based on a time sequence of the voice transmission request.

Abstract: An information processing device 100 of the present invention includes an analysis unit 121 and an encoding unit 122. The analysis unit 121 extracts a partial time-series data set obtained by dividing a time-series data set that is a set of time-series data including a plurality of elements at given time intervals, and calculates correlation data representing a correlation between elements of time-series data included in the partial time-series data set. The encoding unit 122 generates coded data based on the time-series data of the partial time-series data set and the correlation data.

Abstract: This application discloses a downmixed signal calculation method and apparatus. The method includes: when a current frame or a previous frame of the current frame of a stereo signal is not a switching frame and a residual signal in the current frame or the previous frame does not need to be encoded, obtaining a second downmixed signal in the current frame and a downmix compensation factor of the current frame, correcting the second downmixed signal in the current frame based on the downmix compensation factor of the current frame, to obtain the first downmixed signal in the current frame and determining the first downmixed signal in the current frame as a downmixed signal in the current frame in a preset frequency band.

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 multi-input, multi-output audio process is implemented as a linear system for use in an audio filterbank to convert a set of frequency-domain input audio signals into a set of frequency-domain output signals. A transfer function from one input to one output is defined as a frequency dependent gain function. In some implementations, the transfer function includes a direct component that is substantially defined as a frequency dependent gain, and one or more decorrelated components that have frequency-varying group phase response. The transfer function is formed from a set of sub-band functions, with each sub-band function being formed from a set of corresponding component transfer functions including direct component and one or more decorrelated components.

Abstract: There are disclosed several examples of encoding and decoding technique. In particular, an audio synthesizer for generating a synthesis signal from a downmix signal, includes: an input interface for receiving the downmix signal, the downmix signal having a number of downmix channels and side information, the side information including channel level and correlation information of an original signal, the original signal having a number of original channels; and a synthesis processor for generating, according to at least one mixing rule, the synthesis signal using: channel level and correlation information of the original signal; and covariance information associated with the downmix signal.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. For coding, portions of the original HOA representation are predicted from the directional signal components. This prediction provides side information which is required for a corresponding decoding. By using some additional specific purpose bits, a known side information coding processing is improved in that the required number of bits for coding that side information is reduced on average.

Abstract: A method for performing DRC on a HOA signal comprises transforming the HOA signal to the spatial domain, analyzing the transformed HOA signal, and obtaining, from results of said analyzing, gain factors that are usable for dynamic compression. The gain factors can be transmitted together with the HOA signal. When applying the DRC, the HOA signal is transformed to the spatial domain, the gain factors are extracted and multiplied with the transformed HOA signal in the spatial domain, wherein a gain compensated transformed HOA signal is obtained. The gain compensated transformed HOA signal is transformed back into the HOA domain, wherein a gain compensated HOA signal is obtained. The DRC may be applied in the QMF-filter bank domain.

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: 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: An apparatus with a base unit comprising a microphone array arranged to capture a plurality of local audio signals for producing a spatially encoded sound-field signal, and a remote microphone device with a microphone and an associated storage portion, wherein the remote microphone device is arranged to capture a remote audio signal associated with a sound source with the microphone and store said remote audio signal in the associated storage portion. The apparatus is further arranged to use the plurality of local audio signals to produce a spatially encoded sound-field signal with a plurality of components, determine a position of the remote microphone device, and generate a spatially encoded soundtrack using the spatially encoded sound-field signal and the stored remote audio signal in accordance with the determined position of the remote microphone device.

Abstract: Encoding and decoding devices for encoding the channels of an audio system having at least four channels are disclosed. The decoding device has a first stereo decoding component which subjects a first pair of input channels to a first stereo decoding, and a second stereo decoding component which subjects a second pair of input channels to a second stereo decoding. The results of the first and second stereo decoding components are crosswise coupled to a third and a fourth stereo decoding component which each performs stereo decoding on one channel resulting from the first stereo decoding component, and one channel resulting from the second stereo decoding component.

Abstract: Ambisonics audio such as may be used for computer simulations such as computer games is improved by using multi-order optimizations that frame an optimization problem that minimizes a cost function across a subset of Ambisonics orders for a chosen Ambisonics order “N”. In a simple form, this cost function minimizes error across all orders (0<=n<=N), and additional weighting is applied to emphasize or de-emphasize particular orders. The cost functions and optimization criteria may be different for binaural and speaker outputs.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: The present invention relates to a method for receiving data transmitted acoustically. The method includes receiving an acoustically transmitted signal encoding data; processing the received signal to minimise environmental interference within the received signal; and decoding the processed signal to extract the data. The data encoded within the signal using a sequence of tones. A method for encoding data for acoustic transmission is also disclosed. This method includes encoding data into an audio signal using a sequence of tones. The audio signal in this method is configured to minimise environmental interference. A system and software are also disclosed.

Abstract: A system for and a method of generating an audio image for use in rendering audio. The method comprises accessing an audio stream; accessing positional information, the positional information comprising a first position, a second position and a third position; and generating an audio image. In some embodiments, generating the audio image comprises generating, based on the audio stream, a first virtual wave front to be perceived by a listener as emanating from the first position; generating, based on the audio stream, a second virtual wave front to be perceived by the listener as emanating from the second position; and generating, based on the audio stream, a third virtual wave front to be perceived by the listener as emanating from the third position.

Abstract: Provided is an acoustic signal encoding method capable of encoding an acoustic signal having a large number of channels at a sufficient bit rate. In this acoustic signal encoding method, the acoustic signal of a plurality of channels are encoded by executing encoding device. Firstly, the masking threshold corresponding to the spatial masking effect of hearing is calculated. Then, the amount of information for allocating the acoustic signal of the plurality of channels to each channel is determined by the calculated masking threshold. Then, the acoustic signal of the plurality of channels are encoded with the amount of information allocated to each. This makes it possible to encode the acoustic signal of the plurality of channels at a sufficient bit rate.

Abstract: Multichannel audio playback devices and associated systems and methods are disclosed herein. In some examples, a first playback device is configured to receive a source stream of audio content comprising left, right and center input channels. In a first mode, the first playback device is configured to play back audio via a plurality of transducers based on the left, right, and center input channels. In a second mode, in which the first playback device is bonded to second and third playback devices, the first playback device is configured to (i) play back audio via the plurality of transducers based on at least the center input channel, (ii) cause audio to be played via the second playback device based on at least the right input channel, and (iii) cause audio to be played via the third playback device based on at least the left input channel.

Abstract: Provided are a method for generating a haptic feedback signal, an electronic device, and a storage medium. The method includes: acquiring an initial haptic feedback signal; mapping an initial haptic feedback signal from a time domain to a frequency domain to obtain an initial spectrogram of the initial haptic feedback signal in the frequency domain; adjusting an amplitude value of a harmonic component of any frequency band in the initial spectrogram to obtain a target spectrogram; and mapping a signal corresponding to the target spectrogram from the frequency domain to the time domain to obtain a target haptic feedback signal. In the present application, a large number of haptic feedback signals with different haptic feedback effects may be obtained by adjusting the amplitude value of the harmonic component of any frequency band in the initial spectrogram, thereby effectively improving the richness of the haptic feedback signals.

Abstract: An apparatus configured to: determine at least one spatial audio parameter for providing spatial audio reproduction; determine at least one transport signal; determine at least one coherence parameter for at least two frequency bands, wherein a sound scene is configured to be reproduced based, at least partially, on the at least one transport signal, the at least one spatial audio parameter, and the at least one coherence parameter; and provide the at least one transport signal, the at least one spatial audio parameter, and the at least one coherence parameter for encoding.

Abstract: The present invention relates to methods and apparatus for encoding an HOA signal representation (c(t)) of a sound field having an order of N and a number O=(N+1)2 of coefficient sequences to a mezzanine HOA signal representation (wMEZZ(t)). The present invention further relates to methods and apparatus for decoding a reconstructed HOA signal representation from the mezzanine HOA signal representation.

Abstract: A compensated deep neural network (compensated-DNN) is provided. A first vector having a set of components and a second vector having a set of corresponding components are received. A component of the first vector includes a first quantized value and a first compensation instruction, and a corresponding component of the second vector includes a second quantized value and a second compensation instruction. The first quantized value is multiplied with the second quantized value to compute a raw product value. The raw product value is compensated for a quantization error according to the first and second compensation instructions to produce a compensated product value. The compensated product value is added into an accumulated value for the dot product. The accumulated value is converted into an output vector of the dot product. The output vector includes an output quantized value and an output compensation instruction.

Abstract: A device may receive, from a user device associated with a user and located at a location, a request to access a service, and may provide, based on the request, an authorization request to an authentication system. The device may receive an authorization code from the authentication system, and may request an access token from an authorization system. The device may receive the access token, and may request user information from the authorization system. The device may receive the user information, a first confidence score, and a second confidence score, and may perform a geofence analysis of the location to generate a third confidence score. The device may determine whether the request is genuine or fraudulent based on the user information, the first confidence score, the second confidence score, and the third confidence score, and may approve or deny the request based on whether the request is genuine or fraudulent.

Abstract: A device includes a memory configured to store untransformed ambisonic coefficients at different time segments. The device includes one or more processors configured to obtain the untransformed ambisonic coefficients at the different time segments, where the untransformed ambisonic coefficients at the different time segments represent a soundfield at the different time segments. The one or more processors are configured to apply one adaptive network, based on a constraint that includes preservation of a spatial direction of one or more audio sources in the soundfield at the different time segments, to the untransformed ambisonic coefficients at the different time segments to generate transformed ambisonic coefficients at the different time segments, wherein the transformed ambisonic coefficients at the different time segments represent a modified soundfield at the different time segments, that was modified based on the constraint.

Abstract: An apparatus comprising means configured to: generate spatial audio signal directional metadata parameters for a block of time-frequencies; generate encoded spatial audio signal directional metadata parameters (108) for a block of time-frequencies based on a first quantization resolution (203); compare a number of bits used for the encoded spatial audio signal directional parameters (108) for the block of time-frequencies based on the first quantization resolution against a determined number of bits; output or store the encoded spatial audio signal directional metadata parameters for a block of time-frequencies (108) based on a first quantization resolution when the number of bits used for the encoded spatial audio signal directional parameters for the block of time-frequencies (108) based on the first quantization resolution is less than a determined number of bits (217); generate encoded spatial audio signal directional metadata parameters (108) for the block of time-frequencies based on a second quantizati

Abstract: Systems and methods for selecting crossover frequencies for 2.1 speaker systems integrated into electronic displays are described. In an embodiment, an electronic display may include: a left speaker coupled to a left side of an enclosure, a right speaker coupled to a right side of the enclosure, and a subwoofer coupled to the left or right side of the enclosure. In another embodiment, a method may include: determining a position of an asymmetric subwoofer integrated into a display, and selecting a crossover frequency between the asymmetric subwoofer and a set of stereo speakers integrated into the display based, at least in part, upon the position.

Abstract: An audio encoding and decoding method and a related apparatus are provided. The audio encoding method includes: determining a channel combination scheme for a current frame; when the channel combination scheme for the current frame is different from a channel combination scheme for a previous frame, performing segmented time-domain downmix processing on left and right channel signals in the current frame based on the channel combination scheme for the current frame and the channel combination scheme for the previous frame, to obtain a primary channel signal and a secondary channel signal in the current frame; and encoding the obtained primary channel signal and secondary channel signal in the current frame.

Abstract: An audio decoding method includes obtaining an encoded bitstream; performing bitstream demultiplexing on the encoded bitstream, to obtain a high frequency band parameter of a current frame of an audio signal, wherein the high frequency band parameter indicates a location, a quantity, and an amplitude or energy of a tone component comprised in a high frequency band signal of the current frame; obtaining a reconstructed high frequency band signal of the current frame based on the high frequency band parameter; and obtaining an audio output signal of the current frame based on the reconstructed high frequency band signal of the current frame.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. For coding, portions of the original HOA representation are predicted from the directional signal components. This prediction provides side information which is required for a corresponding decoding. By using some additional specific purpose bits, a known side information coding processing is improved in that the required number of bits for coding that side information is reduced on average.

Abstract: This application discloses a downmixed signal calculation method and apparatus. The method includes: when a current frame or a previous frame of the current frame of a stereo signal is not a switching frame and a residual signal in the current frame or the previous frame does not need to be encoded, obtaining a second downmixed signal in the current frame and a downmix compensation factor of the current frame, correcting the second downmixed signal in the current frame based on the downmix compensation factor of the current frame, to obtain the first downmixed signal in the current frame and determining the first downmixed signal in the current frame as a downmixed signal in the current frame in a preset frequency band.

Abstract: An apparatus for generating a sound field description from an input signal having at least two channels has: an input signal analyzer for obtaining direction data and diffuseness data from the input signal; an estimator for estimating a first energy- or amplitude-related measure for an omnidirectional component derived from the input signal and for estimating a second energy- or amplitude-related measure for a directional component derived from the input signal, and a sound component generator for generating sound field components of the sound field, wherein the sound component generator is configured to perform an energy compensation of the directional component using the first energy- or amplitude-related measure, the second energy- or amplitude-related measure, the direction data and the diffuseness data.

Abstract: A method for performing DRC on a HOA signal comprises transforming the HOA signal to the spatial domain, analyzing the transformed HOA signal, and obtaining, from results of said analyzing, gain factors that are usable for dynamic compression. The gain factors can be transmitted together with the HOA signal. When applying the DRC, the HOA signal is transformed to the spatial domain, the gain factors are extracted and multiplied with the transformed HOA signal in the spatial domain, wherein a gain compensated transformed HOA signal is obtained. The gain compensated transformed HOA signal is transformed back into the HOA domain, wherein a gain compensated HOA signal is obtained. The DRC may be applied in the QMF-filter bank domain.

Abstract: In one embodiment, an apparatus includes: a sensor to sense real world information; a digitizer coupled to the sensor to digitize the real world information into digitized information; a signal processor coupled to the digitizer to process the digitized information into a spectrogram; a neural engine coupled to the signal processor, the neural engine comprising an autoencoder to compress the spectrogram into a compressed spectrogram; and a wireless circuit coupled to the neural engine to send the compressed spectrogram to a remote destination, to enable the remote destination to process the compressed spectrogram.

Abstract: A schematic block diagram of an audio encoder for encoding a multichannel audio signal is shown. The audio encoder includes a linear prediction domain encoder, a frequency domain encoder, and a controller for switching between the linear prediction domain encoder and the frequency domain encoder. The controller is configured such that a portion of the multichannel signal is represented either by an encoded frame of the linear prediction domain encoder or by an encoded frame of the frequency domain encoder. The linear prediction domain encoder includes a downmixer for downmixing the multichannel signal to obtain a downmixed signal. The linear prediction domain encoder further includes a linear prediction domain core encoder for encoding the downmix signal and furthermore, the linear prediction domain encoder includes a first joint multichannel encoder for generating first multichannel information from the multichannel signal.

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: An apparatus for decoding an encoded multichannel signal of a current frame to obtain three or more current audio output channels is provided. A multichannel processor is adapted to select two decoded channels from three or more decoded channels depending on first multichannel parameters. Moreover, the multichannel processor is adapted to generate a first group of two or more processed channels based on the selected channels. A noise filling module is adapted to identify for at least one of the selected channels, one or more frequency bands, within which all spectral lines are quantized to zero, and to generate a mixing channel using, depending on side information, a proper subset of three or more previous audio output channels that have been decoded, and to fill the spectral lines of frequency bands, within which all spectral lines are quantized to zero, with noise generated using spectral lines of the mixing channel.

Abstract: A method for compressing a HOA signal being an input HOA representation with input time frames (C(k)) of HOA coefficient sequences comprises spatial HOA encoding of the input time frames and subsequent perceptual encoding and source encoding. Each input time frame is decomposed (802) into a frame of predominant sound signals (XPS(k?1)) and a frame of an ambient HOA component ({tilde over (C)}AMB(k?1)). The ambient HOA component ({tilde over (C)}AMB(k?1)) comprises, in a layered mode, first HOA coefficient sequences of the input HOA representation (cn(k?1)) in lower positions and second HOA coefficient sequences (cAMB,n(k?1)) in remaining higher positions. The second HOA coefficient sequences are part of an HOA representation of a residual between the input HOA representation and the HOA representation of the predominant sound signals.

Abstract: A first device obtains, from the array, several audio signals and processes the audio signals to produce a speech signal and one or more ambient signals. The first device processes the ambient signals to produce a sound-object sonic descriptor that has metadata describing a sound object within an acoustic environment. The first device transmits, over a communication data link, the speech signal and the descriptor to a second electronic device that is configured to spatially reproduce the sound object using the descriptor mixed with the speech signal, to produce several mixed signals to drive several speakers.

Abstract: A spectrum encoding method includes selecting an important spectral component in band units for a normalized spectrum and encoding information of the selected important spectral component for a band, based on a number, a position, a magnitude and a sign thereof. A spectrum decoding method includes obtaining from a bitstream, information about an important spectral component for a band of an encoded spectrum and decoding the obtained information of the important spectral component, based on a number, a position, a magnitude and a sign of the important spectral component.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: A device includes a memory configured to store untransformed ambisonic coefficients at different time segments. The device also includes one or more processors configured to obtain the untransformed ambisonic coefficients at the different time segments, where the untransformed ambisonic coefficients at the different time segments represent a soundfield at the different time segments. The one or more processors are also configured to apply one adaptive network, based on a constraint, to the untransformed ambisonic coefficients at the different time segments to generate transformed ambisonic coefficients at the different time segments, wherein the transformed ambisonic coefficients at the different time segments represent a modified soundfield at the different time segments, that was modified based on the constraint.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: The present technology relates to an encoding device and method, a decoding device and method, and a program, which are adapted to be capable of improving convenience. The decoding device is provided with: a decoding unit that decodes audio data including an object audio, the audio data being included in an encoded bit stream, and reads metadata of the object audio from an area in which arbitrary data of the encoded bit stream can be stored; and an output unit that outputs the decoded audio data on the basis of the metadata. The present technology can be applied to the decoding device.

Abstract: An audio output apparatus is disclosed. The audio output apparatus that outputs a multi-channel audio signal through a plurality of speakers disposed at different locations, the audio output apparatus includes an input interface, and a processor configured to, based on the multi-channel audio signal input through the inputter being received, obtain scene information on a type of audio included in the multi-channel audio signal and sound image angle information about an angle formed by sound image of the type of audio included in the multi-channel audio signal based on a virtual user, and generate an output signal to be output through the plurality of speakers from the multi-channel audio signal based on the obtained scene information and sound image angle information, wherein the type of audio includes at least one of sound effect, shouting sound, music, and voice, and a number of the plurality of speakers is equal to or greater than a number of channels of the multi-channel audio signal.