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.

Abstract: An apparatus for encoding an audio signal includes: a core encoder for core encoding first audio data in a first spectral band; a parametric coder for parametrically coding second audio data in a second spectral band being different from the first spectral band, wherein the parametric coder includes: an analyzer for analyzing first audio data in the first spectral band to obtain a first analysis result and for analyzing second audio data in the second spectral band to obtain a second analysis result; a compensator for calculating a compensation value using the first analysis result and the second analysis result; and a parameter calculated for calculating a parameter from the second audio data in the second spectral band using the compensation value.

Abstract: A memory system includes a non-volatile memory and a memory controller. The memory controller is configured to queue write commands received from a host. The commands may include a first write command to write first write data associated with a first stream and a second write command to write second write data associated with a second stream. When the first and second write commands are queued, the memory controller repeatedly performs a sequence of a first operation of acquiring a predetermined amount of the first write data from the host and then transmitting to the non-volatile memory, and a second operation of acquiring the predetermined amount of the second write data from the host and then transmitting to the non-volatile memory. The second operation in the sequence is started after completion of the first operation in the sequence.

Abstract: A method and device for compressing audio signals forming, over time, a succession of sample frames, in each of N channels of an ambisonic representation of order higher than 0. The method includes: forming, based on the channels and for a current frame, a matrix of inter-channel covariance, and searching for eigenvectors of the covariance matrix with a view to obtaining a matrix of eigenvectors; testing the matrix of eigenvectors to verify that it represents a rotation in an N-dimensional space, and if not, correcting the matrix of eigenvectors until a rotation matrix is obtained, for the current frame; and applying the rotation matrix to the signals of the N channels before separate-channel encoding of the signals.

Abstract: A base transmission signal and an auxiliary transmission signal may be used to (e.g., concurrently) improve a harvesting efficiency of an energy receiver while maintaining a prespecified performance level of an information receiver. An auxiliary transmission signal may be constructed. The construction of the auxiliary transmission signal may be signaled. An auxiliary transmission signal may be adapted. The adaptation of the auxiliary transmission signal may be signaled. An impact of auxiliary transmission signals on an information receiver's performance may be mitigated. One or more narrowband energy harvesting (EH) signals may be used, for example to provide a high peak to average power ratio (PARR) for EH devices (e.g., without significantly impacting overall transmitted signal PARR) characteristics. One or more narrowband auxiliary signals may be used, for example to enhance the PARR characteristics of an information signal sub-band.

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

Abstract: An inter-channel phase difference (IPD) parameter extraction method includes obtaining a parameter for obtaining an information extraction manner for a current frame of a multi-channel signal; obtaining an IPD parameter extraction manner for the current frame based on the parameter for obtaining the information extraction manner, where the obtained IPD parameter extraction manner is one of at least two preset IPD parameter extraction manners; and obtaining an IPD parameter of the current frame based on the obtained IPD parameter extraction manner for the current frame.

Abstract: An apparatus for selecting one of a first encoding algorithm having a first characteristic and a second encoding algorithm having a second characteristic for encoding a portion of an audio signal to obtain an encoded version of the portion of the audio signal has a first estimator for estimating a first quality measure for the portion of the audio signal, which is associated with the first encoding algorithm, without actually encoding and decoding the portion of the audio signal using the first encoding algorithm. A second estimator is provided for estimating a second quality measure for the portion of the audio signal, which is associated with the second encoding algorithm, without actually encoding and decoding the portion of the audio signal using the second encoding algorithm. The apparatus has a controller for selecting the first or second encoding algorithms based on a comparison between the first and second quality measures.

Abstract: A multi-channel signal encoding method includes determining a downmixed signal of a first channel signal and a second channel signal, determining an initial reverberation gain parameter of the first channel signal and the second channel signal, determining a target reverberation gain parameter of the first channel signal and the second channel signal based on a correlation between the first channel signal and the downmixed signal, a correlation between the second channel signal and the downmixed signal, and the initial reverberation gain parameter, quantizing the first channel signal and the second channel signal based on the downmixed signal and the target reverberation gain parameter, and writing a quantized first channel signal and a quantized second channel signal into a bitstream.

Abstract: A signal processing method and device includes obtaining spectral coefficients of a current frame of an audio signal, in which N sub-bands of the current frame comprises at least one of the spectral coefficients. A total energy of M successive sub-bands of the N sub-bands, a total energy of K successive sub-bands of the N sub-bands, and an energy of a first sub-band are obtained to determine whether to modify original envelope values of the M sub-bands. When the original envelope values of the M sub-bands are modified, encoding bits are allocated to each of the N sub-bands according to the modified envelope values of the M sub-bands.

Abstract: Provided is a method for audio peak reduction using an all-pass filter, including: determining a delay parameter m and a gain parameter g based on a formula (1): arg ? min m , g max n ? "\[LeftBracketingBar]" y m , g ( n ) ? "\[RightBracketingBar]" , ( 1 ) absolute peak map Y ? ( m , g ) = max n ? "\[LeftBracketingBar]" y m , g ( n ) ? "\[RightBracketingBar]" , ym,g(n) represents a processed signal with a time-domain response function and is calculated based on a formula (2): ym,g(n)=(hs*x)(n)(2), where hs represents an impulse response function, x (n) represents an input signal, and hs is calculated based on formula (3): H S ( z ) = g + z - m 1 + gz - m . ( 3 ) This method is widely used in the reproduction, storage and broadcasting of sound, and the computational complexity is small, which is a supplement to the traditional nonlinear compression algorithm.

Abstract: A method, system, and device are directed to audio input bit-size conversion for compatibility to audio processing systems with an expected input sample bit-size.

Abstract: A method and apparatus for audio signal processing in an audio chain to correct a non-linearity of the electroacoustic transducers in the audio chain by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain using an approximation of the quadratic function. The method accommodates the psychoacoustical characteristics of the human ear by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain approximating by a non-linear fifth degree polynomial function for a pressure change by the human ear up to p_?. The method and apparatus reduce non-linearities of the entire audio chain with the human ear, by adding non-linearities in the audio chain so that an audio chain characteristic reduces the non-linearity of the human ear polynomial approximation to the pressure change p_?=±1 Pa.

Abstract: Embodiments relate to audio processing unit(s) and methods for decoding an encoded audio bitstream, that includes a fill element with an identifier indicating a start of the fill element and fill data which includes a flag identifying whether to perform a base form of spectral band replication or an enhanced form of spectral band replication, wherein the base form of spectral band replication includes spectral patching, the enhanced form of spectral band replication includes harmonic transposition, one value of the flag indicates that said enhanced form of spectral band replication should be performed on the audio content, and another indicates that said base form of spectral band replication but not said harmonic transposition should be performed on the audio content, wherein the fill data further includes a parameter indicating whether pre-flattening is to be performed after spectral patching for avoiding spectral discontinuities.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: Provided is a data processing system which includes a processor and a storage device, and inputs/outputs data using a learned compander that compresses and expands data, wherein the data processing system comprises an estimation unit which uses learning data and estimates a region of interest to a data model, and a learning unit which causes the compander to learn according to an evaluation function in which each region was weighted based on the region of interest, and a result of the compander compressing and expanding the learning 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: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A method of processing an audio signal includes the following operations. A system receives an audio signal. The system determines that a first frame of the audio signal includes a sparse transient signal. The system determines that a second frame of the audio signal includes a dense transient signal. The system compresses/expands (compands) the audio signal using a companding rule that applies a first companding exponent to the first frame of the audio signal and applies a second companding exponent to the second frame of the audio signal, each companding exponent being used to derive a respective degree of dynamic range compression and expansion for a corresponding frame. The system then provides the companded audio signal to a downstream device.

Abstract: An example method of operation includes identifying a venue geometry from a stored value, determining a venue polar representation based on the venue geometry, selecting one or more loudspeaker types to place in the venue based on the venue geometry and one or more target sound pressure levels (SPLs), selecting one or more locations in the venue to place the one or more loudspeakers, identifying one or more angles to position the one or more loudspeakers in the venue at the selected locations, and generating a display model of a loudspeaker arrangement comprising a selected number of speakers and one or more filters to apply to the one or more loudspeakers.

Abstract: An audio encoder has a first information sink oriented encoding branch such as a spectral domain encoding branch, a second information source or SNR oriented encoding branch such as an LPC-domain encoding branch, and a switch for switching between the first and second encoding branches, the second encoding branch having a converter into a specific domain different from the spectral domain such as an LPC analysis stage generating an excitation signal, and the second encoding branch having a specific domain coding branch such as LPC domain processing branch, and a specific spectral domain coding branch such as LPC spectral domain processing branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch. An audio decoder has a first domain decoder, a second domain decoder, and a third domain decoder as well as two cascaded switches for switching between the decoders.

Abstract: An apparatus configured to: receive at least two audio signals; determine at least one parameter associated with at least two audio signals, wherein the at least one parameter is configured to represent an ambiance energy distribution of the at least two audio signals, wherein the at least one parameter is associated with, at least, a respective energy of ambient sound of the at least two audio signals in a plurality of directions; determine at least one directional parameter; and provide, using at least one of the at least two audio signals, at least one output audio signal based on, at least, the at least one directional parameter and the at least one parameter, wherein the at least one parameter controls ambiance energy distribution of the at least one output signal according to the ambience energy distribution of the at least two audio signals in the plurality of directions.

Abstract: Disclosed are a method of encoding and decoding an audio signal and an encoder and a decoder performing the method. The method of encoding an audio signal includes identifying an input signal, and generating a bitstring of each encoding layer by applying, to the input signal, an encoding model including a plurality of successive encoding layers that encodes the input signal, in which a current encoding layer among the encoding layers is trained to generate a bitstring of the current encoding layer by encoding an encoded signal which is a signal encoded in a previous encoding layer and quantizing an encoded signal which is a signal encoded in the current encoding layer.

Abstract: A headset, during play of a particular game, receives a plurality of audio channels carrying game sounds and determines whether a first of the game sounds overwhelms a second of the game sounds. In response to the determination, the headset adjusts a level of one or more of the audio channels such that perceptibility of the second game sound is improved relative to perceptibility of the first game sound prior to the adjustment, wherein for the adjustment, a level of one or more of the audio channels carrying the first game sound is decreased while a level of one or more of the audio channels carrying the second game sound is maintained or increased. The audio channels include three or more audio channels and the adjustment of the level of the audio channels is performed while the three or more audio channels are combined into two stereo channels.

Abstract: The present invention provides a signal processing method of a speaker, wherein the signal processing method includes the steps of: receiving a plurality of packets, wherein the plurality of packets correspond to at least two audio frames, each audio frame has at least two of the packets, and each packet comprises a sequence number indicating an order of the packet in the plurality of packets; sorting the plurality of packets to generate reordered packets; and rendering the reordered packets to a middleware to regenerate the at least two audio frames.

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: A spoofing detection apparatus 100 includes a multi-channel spectrogram creation unit 10 and an evaluation unit 40. The multi-channel spectrogram creation unit 10 extracts different type of spectrograms from speech data and integrates the different type of spectrograms to create a multi-channel spectrogram. The evaluation unit 40 evaluates the created multi-channel spectrogram by applying the created multi-channel spectrogram to a classifier constructed using labeled multi-channel spectrograms as training data and classifies it to either genuine or spoof.

Abstract: Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.

Abstract: An apparatus for decoding an encoded audio signal having an encoded representation of a first set of first spectral portions and an encoded representation of parametric data indicating spectral energies for a second set of second spectral portions, has: an audio decoder for decoding the encoded representation of the first set of the first spectral portions to obtain a first set of first spectral portions and for decoding the encoded representation of the parametric data to obtain a decoded parametric data for the second set of second spectral portions indicating, for individual reconstruction bands, individual energies; a frequency regenerator for reconstructing spectral values in a reconstruction band having a second spectral portion using a first spectral portion of the first set of the first spectral portions and an individual energy for the reconstruction band, the reconstruction band having a first spectral portion and the second spectral portion.

Abstract: A computer-implemented sound signal generation method includes: obtaining a first sound source spectrum of a sound signal to be generated; obtaining a first spectral envelope of the sound signal; and estimating fragment data representative of samples of the sound signal based on the obtained first sound source spectrum and the obtained first spectral envelope.

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

Abstract: Systems and methods are presented for cross-fading (or other multiple clip processing) of information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Multiple clip processing can be accomplished at a client end according to directions sent from a service provider that specify a combination of (i) the clips involved; (ii) the device on which the cross-fade or other processing is to occur and its parameters; and (iii) the service provider system. For example, a consumer device with only one decoder, can utilize that decoder (typically hardware) to decompress one or more elements that are involved in a cross-fade at faster than real time, thus pre-fetching the next element(s) to be played in the cross-fade at the end of the currently being played element.

Abstract: This disclosure provides a decoding method, and a decoding apparatus for a stereo signal. The decoding method includes: decoding a bitstream to obtain a first channel signal, a second channel signal, and a first ITD of a current frame of a stereo signal; performing a mixing processing on the first channel signal and the second channel signal, to obtain a third channel reconstructed signal and a fourth channel reconstructed signal; performing interpolation processing based on the first ITD and a second ITD of a previous frame previous to the current frame, to obtain a third ITD; and adjusting a delay of the third channel reconstructed signal and the fourth channel reconstructed signal based on the third ITD.

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: A method for interpolating a sound field captured by a plurality of N microphones each outputting the encoded sound field in a form including at least one captured pressure and an associated pressure gradient vector. Such a method includes an interpolation of the sound field at an interpolation position outputting an interpolated encoded sound field as a linear combination of the N encoded sound fields each weighted by a corresponding weighting factor. The interpolation includes an estimation of the N weighting factors at least from: the interpolation position; a position of each of the N microphones; the N pressures captured by the N microphones; and an estimated power of the sound field at the interpolation position.

Abstract: One or more embodiments include techniques for generating immersive audio for an acoustic system. The techniques include determining an apparent location associated with a portion of audio; calculating, for each speaker included in a plurality of speakers of the acoustic system, a perceptual distance between the speaker and the apparent location; selecting a subset of speakers included in the plurality of speakers based on the perceptual distances between the plurality of speakers and the apparent location; generating a set of filters based on the subset of speakers and one or more target characteristics of the acoustic system; and generating, for each speaker included in the subset of speakers, a speaker signal using one or more filters included in the set of filters.

Abstract: The present disclosure relates to time-domain stereo parameter encoding methods and apparatus. One example 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: Methods for generating an object based audio program which is renderable in a personalizable manner, e.g., to provide an immersive, perception of audio content of the program. Other embodiments include steps of delivering (e.g., broadcasting), decoding, and/or rendering such a program. Rendering of audio objects indicated by the program may provide an immersive experience. The audio content of the program may be indicative of multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects, and typically also a default set of objects which will be rendered in the absence of a selection by a user) and a bed of speaker channels. Another aspect is an audio processing unit (e.g., encoder or decoder) configured to perform, or which includes a buffer memory which stores at least one frame (or other segment) of an object based audio program (or bitstream thereof) generated in accordance with, any embodiment of the method.

Abstract: In accordance with an example embodiment of the present invention, disclosed is a method and an apparatus thereof for selecting a packet loss concealment procedure for a lost audio frame of a received audio signal. A method for selecting a packet loss concealment procedure comprises detecting an audio type of a received audio frame and determining a packet loss concealment procedure based on the audio type. In the method, detecting an audio type comprises determining a stability of a spectral envelope of signals of received audio frames.

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: Aspects of the disclosure include methods, apparatuses, and non-transitory computer-readable storage mediums for decoding audio data of an audio scene. One apparatus includes processing circuitry that receives first audio source data and second audio source data. The first audio source data corresponds to a space of interest in the audio scene and the second audio source data does not correspond to the space of interest in the audio scene. The space of interest in the audio scene is represented by at least one of a listener space, an audio channel, or an audio object. The processing circuitry decodes the first audio source data based on the space of interest.

Abstract: An apparatus for generating an enhanced signal from an input signal, wherein the enhanced signal has spectral values for an enhancement spectral region, the spectral values for the enhancement spectral regions not being contained in the input signal, includes a mapper for mapping a source spectral region of the input signal to a target region in the enhancement spectral region, the source spectral region including a noise-filling region; and a noise filler configured for generating first noise values for the noise-filling region in the source spectral region of the input signal and for generating second noise values for a noise region in the target region, wherein the second noise values are decorrelated from the first noise values or for generating second noise values for a noise region in the target region, wherein the second noise values are decorrelated from first noise values in the source region.

Abstract: An illustrative frequency-domain encoder system transforms time-domain data representative of a content instance into frequency-domain data representative of the content instance. The frequency-domain data includes a plurality of complex coefficients each representing different frequency components of a plurality of frequency components incorporated by the content instance. The frequency-domain encoder system generates a frequency-domain data container that includes the complex coefficients of the frequency-domain data and metadata descriptive of the frequency-domain data. Additionally, within the frequency-domain data container, the frequency-domain encoder system integrates the complex coefficients of the frequency-domain data with timing data representative of a time-dependent feature of the content instance. Corresponding systems and methods are also disclosed.

Abstract: The present document describes a method (700) for encoding a multi-channel input signal (201). The method (700) comprises determining (701) a plurality of downmix channel signals (203) from the multi-channel input signal (201) and performing (702) energy compaction of the plurality of downmix channel signals (203) to provide a plurality of compacted channel signals (404). Furthermore, the method (700) comprises determining (703) joint coding metadata (205) based on the plurality of compacted channel signals (404) and based on the multi-channel input signal (201), wherein the joint coding metadata (205) is such that it allows upmixing of the plurality of compacted channel signals (404) to an approximation of the multi-channel input signal (201). In addition, the method (700) comprises encoding (704) the plurality of compacted channel signals (404) and the joint coding metadata (205).

Abstract: The invention provides an audio encoder for encoding an audio signal so as to produce therefrom an encoded signal, the audio encoder including: a framing device configured to extract frames from the audio signal; a quantizer configured to map spectral lines of a spectrum signal derived from the frame of the audio signal to quantization indices, wherein the quantizer has a dead-zone, in which the input spectral lines are mapped to quantization index zero; and a control device configured to modify the dead-zone; wherein the control device includes a tonality calculating device configured to calculate at least one tonality indicating value for at least one spectrum line or for at least one group of spectral lines, wherein the control device is configured to modify the dead-zone for the at least one spectrum line or the at least one group of spectrum lines depending on the respective tonality indicating value.

Abstract: A high-band encoding/decoding method and device for bandwidth extension are provided. A high-band encoding method comprising the steps of: generating sub band-specific bit allocation information on the basis of a low-band envelope; determining, on the basis of the sub band-specific bit allocation information, the sub band requiring an envelope update in a high band; and generating, for the determined sub band, refinement data relating to the envelope update. A high-band decoding method comprising the steps of: generating sub band-specific bit allocation information on the basis of a low-band envelope; determining, on the basis of the sub band-specific bit allocation information, the sub band requiring an envelope update in a high band; and decoding, for the determined sub band, refinement data relating to the envelope update, thereby updating the envelope.

Abstract: The present invention relates to a method of triggering an event. The method includes receiving a signal stream, detecting a trigger point within the signal stream using a fingerprint associated with the trigger point and triggering an event associated with the detected trigger point.

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 apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part.