Abstract: The application relates to HFR (High Frequency Reconstruction/Regeneration) of audio signals. In particular, the application relates to a method and system for performing HFR of audio signals having large variations in energy level across the low frequency range which is used to reconstruct the high frequencies of the audio signal. A system configured to generate a plurality of high frequency subband signals covering a high frequency interval from a plurality of low frequency subband signals is described.

Abstract: Noise filling of a spectrum of an audio signal is improved in quality with respect to the noise filled spectrum so that the reproduction of the noise filled audio signal is less annoying, by performing the noise filling in a manner dependent on a tonality of the audio signal.

Abstract: To enable multichannel audio data to be transmitted favorably. Multichannel audio data of a predetermined number of channels is acquired. The multichannel audio data has a sampling frequency corresponding to the predetermined number of channels. Audio data of the respective channels configuring the multichannel audio data are sequentially transmitted to a reception side via a predetermined transmission channel for each unit audio data. Information indicating the sampling frequency is added to the transmission audio data.

Abstract: A computing system that facilitates decoding a spherical harmonics (SH) representation of a three-dimensional sound signal to a binaural sound signal is described herein. The computing system generates a binaural sound signal based upon the SH representation, a tapering window function that is selected based on an SH encoding order of the SH representation, and a coloration compensation filter that incorporates the tapering window function. The computing system causes the binaural sound signal to be played over at least two speakers.

Abstract: An encoder for encoding a parametric spectral representation (ƒ) 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 (ƒH) of the parametric spectral representation (ƒ) by weighted averaging based on the quantized elements ({circumflex over (ƒ)}L) flipped around a quantized mirroring frequency ({circumflex over (ƒ)}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: Methods, apparatus, systems and articles of manufacture are disclosed for audio equalization. Example apparatus disclosed herein include a volume adjuster to apply equalization adjustments to an audio signal to generate an equalized audio signal, the equalization adjustments output from a neural network in response to an input feature set; a thresholding controller to: detect an irregularity in a frequency representation of the audio signal after application of the equalization adjustments, the irregularity corresponding to a change in volume between adjacent frequency values exceeding a threshold; and adjust a volume at a first frequency value of the adjacent frequency values to reduce the irregularity; an equalization (EQ) curve generator to generate an EQ curve to apply to the audio signal when the irregularity has been reduced; and a frequency to time domain converter to output the equalized audio signal in a time domain based on the EQ curve.

Abstract: Aspects of the present disclosure provide methods, apparatuses, and systems for closed-loop sleep protection and/or sleep regulation. According to an aspect, sleep disturbing noises are predicted and a biosignal parameter is measured to dynamically mask predicted disturbing environmental noises in the sleeping environment with active attenuation. Environmental noises in a sleeping environment of a subject are detected, input, or predicted based on historical data of the sleeping environment collected over a period of time. The biosignal parameter is used to determine sleep physiology of a subject. Based on the environmental noises in the sleeping environment and the determined sleep physiology, the noises are predicted to be disturbing or non-disturbing noises. For predicted disturbing noises, one or more actions are taken to regulate sleep and avoid sleep disruption by using sound masking prior to or concurrently with the occurrence of the predicted disturbing noises.

Abstract: The present disclosure relates to a communication method and system for converging a 5th-Generation (5G) communication system for supporting higher data rates beyond a 4th-Generation (4G) system with a technology for Internet of Things (IoT). The present disclosure may be applied to intelligent services based on the 5G communication technology and the IoT-related technology, such as smart home, smart building, smart city, smart car, connected car, health care, digital education, smart retail, security and safety services. A method for performing radio link monitoring (RLM) in a wireless communication system is provided. The method includes determining at least one subband for RLM by a UE restricted to use a subband corresponding to a part of a system transmission bandwidth, wherein the subband is a preconfigured part of the system transmission bandwidth, performing RLM in the determined at least one subband, and determining a radio link quality of the at least one subband based on the RLM.

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: A system for resolving a resource deadlock between processes. A shared data structure is maintained that includes process records of the processes. Process states and process priorities are defined for each of the processes. A respective state of the process is determined for each process based on the process states and process priorities of the process records maintained in the shared data structure. The respective state is used to allocate and deallocate resources to the process to mitigate and resolve the resource deadlock between the processes.

Abstract: Various implementations include wearable audio devices and related methods for controlling such devices. In some particular implementations, a computer-implemented method of controlling a wearable audio device includes: receiving an initiation command to initiate a spatial audio mode; providing a plurality of audio choices corresponding with spatially delineated zones in an array defined relative to a physical position of the wearable audio device, in response to the initiation command, wherein each audio choice is associated with a prescribed action; receiving a selection command selecting one of the plurality of audio choices; and taking one of the prescribed actions in response to receiving the selection command.

Abstract: A music synthesis method, a system, a terminal and a computer-readable storage medium are provided. The method includes: receiving a track selected by a user; obtaining a text; receiving speech data recorded by the user on the basis of the text; and forming a music file in accordance with the selected track and the speech data. The speech of a user can be combined with the track through the music synthesis method of the present application and an optimal effect of music can be simulated such that the user can participate in the singing and presentation of a music, thereby making music more entertaining.

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: A method for secure device-to-device communication using multilayered ciphers is provided. A selected cipher is employed to generate a pair of encryption/decryption keystreams for enabling multilayered encryption/decryption on a pulsed-index communication (PIC) packet(s). In examples discussed herein, a first layer encryption/decryption is performed by encrypting/decrypting a PIC data(s) (PD(s)) in the PIC packet(s) based on a first of the pair of encryption/decryption keystreams. In addition, a second layer encryption/decryption is performed by encrypting/decrypting selected control information (e.g., information related to encoding/decoding the PD(s)) in the PIC packet(s) based on a second of the pair of encryption/decryption keystreams.

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 (a) 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: The present disclosure relates to a communication method and system for converging a 5th-Generation (5G) communication system for supporting higher data rates beyond a 4th-Generation (4G) system with a technology for Internet of Things (IoT). The present disclosure may be applied to intelligent services based on the 5G communication technology and the IoT-related technology, such as smart home, smart building, smart city, smart car, connected car, health care, digital education, smart retail, security and safety services. A method for performing radio link monitoring (RLM) in a wireless communication system is provided. The method includes determining at least one subband for RLM by a UE restricted to use a subband corresponding to a part of a system transmission bandwidth, wherein the subband is a preconfigured part of the system transmission bandwidth, performing RLM in the determined at least one subband, and determining a radio link quality of the at least one subband based on the RLM.

Abstract: Systems and methods include audio encoders having improved coding of harmonic signals. The audio encoders can be implemented as transform-based codecs with frequency coefficients quantized using spectral weights. The frequency coefficients can be quantized by use of the generated spectral weights applied to the frequency coefficients prior to the quantization or by use of the generated spectral weights in computation of error within a vector quantization that performs the quantization. Additional apparatus, systems, and methods are disclosed.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for audio equalization. Example apparatus disclosed herein include a volume adjuster to apply equalization adjustments to an audio signal to generate an equalized audio signal, the equalization adjustments output from a neural network in response to an input feature set; a thresholding controller to: detect an irregularity in a frequency representation of the audio signal after application of the equalization adjustments, the irregularity corresponding to a change in volume between adjacent frequency values exceeding a threshold; and adjust a volume at a first frequency value of the adjacent frequency values to reduce the irregularity; an equalization (EQ) curve generator to generate an EQ curve to apply to the audio signal when the irregularity has been reduced; and a frequency to time domain converter to output the equalized audio signal in a time domain based on the EQ curve.

Abstract: Embodiments relate to an audio processing unit that includes a bitstream payload deformatter and a decoding subsystem. The decoding subsystem is coupled to the bitstream payload deformatter and configured to decode at least a portion of a block of an encoded audio bitstream. The block includes a fill element with an identifier indicating a start of the fill element and fill data after the identifier. The fill data includes at least one flag identifying whether a base form of spectral band replication or an enhanced form of spectral band replication is to be performed on audio content of the block. The identifier is a three bit unsigned integer transmitted most significant bit first and having a value of 0x6.

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

Abstract: The invention provides a concept for combined dynamic range compression and guided clipping prevention for audio devices. An audio decoder for decoding an audio bitstream and a metadata bitstream related to the audio bitstream according to the concept includes an audio processing chain including a plurality of adjustment stages including a dynamic range control stage for adjusting a dynamic range of the audio output signal and a guided clipping prevention stage for preventing clipping of the audio output signal; and a metadata decoder configured to receive the metadata bitstream and to extract dynamic range control gain sequences and guided clipping prevention gain sequences from the metadata bitstream, at least a part of the dynamic range control gain sequences being supplied to the dynamic range control stage, and at least a part of the guided clipping prevention gain sequences being supplied to the guided clipping prevention stage.

Abstract: A method for detecting a hotword includes receiving a sequence of input frames that characterize streaming audio captured by a user device and generating a probability score indicating a presence of a hotword in the streaming audio using a memorized neural network. The network includes sequentially-stacked single value decomposition filter (SVDF) layers and each SVDF layer includes at least one neuron. Each neuron includes a respective memory component, a first stage configured to perform filtering on audio features of each input frame individually and output to the memory component, and a second stage configured to perform filtering on all the filtered audio features residing in the respective memory component. The method also includes determining whether the probability score satisfies a hotword detection threshold and initiating a wake-up process on the user device for processing additional terms.

Abstract: An audio processor for processing an audio signal includes an audio signal phase measure calculator configured for calculating a phase measure of an audio signal for a time frame, a target phase measure determiner for determining a target phase measure for the time frame, and a phase corrector configured for correcting phases of the audio signal for the time frame using the calculated phase measure and the target phase measure to obtain a processed audio signal.

Abstract: In general, the present invention relates to a method and apparatus for estimating spatial content of a soundfield at a desired location, including a location that has actual sound content obstructed or distorted. According to certain aspects, the present invention aims at presenting a more natural, spatially accurate sound, for example to a user at the desired location who is wearing a helmet, mimicking the sound a user would experience if they were not wearing any headgear. Modes for enhanced spatial hearing may be applied which would include situation-dependent processing for augmented hearing. According to other aspects, the present invention aims at remotely reproducing the soundfield at a desired location with faithful reproduction of the spatial content of the soundfield.

Abstract: The present invention relates to a method and an apparatus for encoding and decoding spectrum coefficients in the frequency domain. The spectrum encoding method may comprise the steps of: selecting an encoding type on the basis of bit allocation information of respective bands; performing zero encoding with respect to a zero band; and encoding information of selected significant frequency components with respect to respective non-zero bands. The spectrum encoding method enables encoding and decoding of spectrum coefficients which is adaptive to various bit-rates and various sub-band sizes. In addition, a spectrum can be encoded using a TCQ method at a fixed bit rate using a bit-rate control module in a codec that supports multiple rates. Encoding performance of the codec can be maximised by encoding high performance TCQ at a precise target bit rate.

Abstract: An apparatus includes a decoder configured to receive, from an encoder, a frame associated with an audio bitstream and with a first sampling rate. The decoder is configured to perform a frequency-domain upmix on data associated with the frame to generate left and right frequency-domain signals and is further configured to generate, based on the left and right frequency-domain signals, left and right time-domain signals that each have a second sampling rate. The second sampling rate is determined by the decoder, based on one or both of the first sampling rate and an output sampling rate, and is adjustable by the decoder to enable different frames to be decoded at different second sampling rates. The decoder is further configured to generate, based on the left and right time-domain signals, left and right resampled signals that each have the output sampling rate.

Abstract: Apparatus, computer readable media and methods for processing a multi-channel, spatial audio format input signal. For example, one such method comprises determining object location metadata based on the received spatial audio format input signal; and extracting object audio signals based on the received spatial audio format input signal, wherein the extracting object audio signals based on the received spatial audio format input signal includes determining object audio signals and residual audio signals.

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: Audio devices and methods are provided for detecting instability in an associated feedforward audio processing system. A microphone provides a feedforward signal for processing by a feedforward filter. The processed signal may provide noise reduction and/or sound enhancement associated with the surrounding environment. The processed signal contributes to a driver signal provided to an acoustic transducer, e.g., a driver, to produce acoustic signals for a user. A processor is configured to detect an indication of instability in one or more of the signals, and to adjust a phase response of the feedforward signal path in response to detecting the indication of instability.

Abstract: Example embodiments disclosed herein relate to signal processing. A method for decomposing a plurality of audio signals from at least two different channels is disclosed. The method comprises obtaining a set of components that are weakly correlated, the set of components generated based on the plurality of audio signals. The method comprises extracting a feature from the set of components, and determining a set of gains associated with the set of components at least in part based on the extracted feature, each of the gains indicating a proportion of a diffuse part in the associated component. The method further comprises decomposing the plurality of audio signals by applying the set of gains to the set of components. Corresponding system and computer program product are also disclosed.

Abstract: A lossless encoding method is provided that includes determining a lossless encoding mode of a quantization coefficient as one of an infinite-range lossless encoding mode and a finite-range lossless encoding mode; encoding the quantization coefficient in the infinite-range lossless encoding mode in correspondence with a result of the lossless encoding mode determination; and encoding the quantization coefficient in the finite-range lossless encoding mode in correspondence with a result of the lossless encoding mode determination.

Abstract: In accordance with some embodiments, an apparatus for privacy protection is provided. The apparatus includes a first housing portion and a second housing portion arranged to receive and enclose one or more personal communication devices. The apparatus further includes at least one sound attenuation layer disposed in the second housing portion, the at least one sound attenuation layer absorbs sound. The apparatus also includes a noise generator to provide one or more noise signal streams and audio output device(s), which are at least partially supported by the first housing portion and coupled to the noise generator to receive the one or more noise signal streams. The audio output device(s) are operable to output noise signal based on the one or more noise signal streams and direct the noise signal at the one or more personal communication devices placed adjacent the at least one sound attenuation layer.

Abstract: The present technology reduces a process load in a reception side when a plurality of types of audi data is transmitted. A metafile having meta information used to acquire, in a reception device, a predetermined number of audio streams including a plurality of groups of encoded data is transmitted. To the metafile, attribute information indicating each attribute of the encoded data of the plurality of groups is inserted. For example, to the metafile, stream correspondence relation information indicating in which audio stream the encoded data of the plurality of groups is included respectively is further inserted.

Abstract: Methods and apparatuses for audio pass-through are disclosed. In one aspect, a method of operating a wearable audio device may involve receiving an ambient audio signal, detecting a first audio signal from within the received ambient audio signal, and determining that the first audio signal meets a first metric for pass-through to a user of the wearable audio device. The first metric may be indicative of a first priority level of the first audio signal to the user. The method may further involve isolating the first audio signal based on filtering the ambient audio signal in response to determining that the first audio signal meets the first metric and adjusting playback of the isolated first audio signal for the user via a speaker.

Abstract: An example sound output apparatus may include an input unit, an output unit, and a processor configured to perform first-filtering of a frequency band of an audio signal received through the input unit based on a sound pressure of the sound output apparatus, second-filtering of a frequency band of the first-filtered audio signal based on a frequency response characteristic of the sound output apparatus, adjust a loudness of the received audio signal based on a perception volume level of the second-filtered audio signal, and output the adjusted audio signal through the output unit.

Abstract: A method includes performing modifying, at a decoder, at least a portion of inter-channel phase difference (IPD) parameter values based on a mismatch value to generate modified IPD parameter values. The mismatch value is indicative of an amount of temporal misalignment between an encoder-side reference channel and an encoder-side target channel. The modified IPD parameter values are applied to a decoded frequency-domain mid channel during an up-mix operation.

Abstract: An apparatus for encoding a multi-channel signal including at least two channels includes a time-spectral converter for converting sequences of blocks of sampling values of the at least two channels into a frequency domain representation having sequences of blocks of spectral values for the at least two channels; a multi-channel processor for applying a joint multi-channel processing to the sequences of blocks of spectral values to obtain at least one result sequence of blocks of spectral values including information related to the at least two channels; a spectral-time converter for converting the result sequence of blocks of spectral values into a time domain representation including an output sequence of blocks of sampling values; and a core encoder for encoding the output sequence of blocks of sampling values to obtain an encoded multi-channel signal.

Abstract: A method of sensing a touch in a display apparatus includes: determining a display frame rate of an input image applied to the display apparatus; determining a number of touch driving blocks according to the display frame rate; and sensing a touch based on driving one touch driving block of the determined touch driving blocks at each display frame.

Abstract: The present disclosure relates to a method for playing audio, an apparatus for playing audio, a playing device, and storage medium, and belongs to the field of electronic technology. The method includes detecting an amount of audio sampling points in a buffer area that is configured to store the audio sampling points received from an electronic device based on a first sampling frequency, wherein the audio sampling points are obtained by decoding audio data from the electronic device; when a first quantity of the sampling points based on the first sampling frequency is out of a default quantity range, adjusting a preset first frequency-division parameter to obtain a second frequency-division parameter; determining a second sampling frequency based on the second frequency-division parameter; and extracting the audio sampling points from the buffer area based on the second sampling frequency for playing audio data.

Abstract: A sound file sound quality identification method is provided. The method includes converting a format of a to-be-identified sound file into a preset reference audio format; performing framing on the sound file to obtain a plurality of frames; and performing Fourier transformation processing on the to-be-identified sound file to obtain a spectrum of each frame. The method also includes performing model matching according to the spectrum of each frame of the to-be-identified sound file to obtain a preliminary classification result of the to-be-identified sound file; determining an energy change point of the to-be-identified sound file according to the spectrum of each frame; and determining a sound quality of the to-be-identified sound file according to the preliminary classification result of the to-be-identified sound file and the energy change point of the to-be-identified sound file.

Abstract: A multi-channel signal encoding method includes determining a downmixed signal of a first channel signal and a second channel signal in a multi-channel signal, and reverberation gain parameters corresponding to different subbands of the first channel signal and the second channel signal, determining a target reverberation gain parameter that needs to be encoded in the reverberation gain parameters corresponding to the different subbands of the first channel signal and the second channel signal, generating parameter indication information, where the parameter indication information is used to indicate a subband corresponding to the target reverberation gain parameter, and encoding the target reverberation gain parameter, the parameter indication information, and the downmixed signal to generate a bitstream.

Abstract: Apparatus, systems, articles of manufacture, and methods are disclosed for multiple scrambled layers for audio watermarking. An example system includes a scrambler executing instructions to: divide a watermark into a plurality of watermark symbols; map the watermark symbols to a plurality of frequency bins in a plurality of frequency clumps according to a first distribution scheme to create a first watermark layer having a first combination of the frequency bins and a second watermark layer having a second combination of the frequency bins, the first combination of the frequency bins and the second combination of the frequency bins partially overlap. The scrambler also is to determine a sequence for shifting watermark symbols among the frequency bins, and generate a second distribution scheme to map the watermark symbols in accordance with the sequence. The example system also includes a transceiver to communicate the second distribution scheme to a device.

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

Abstract: The present document relates to time-alignment of encoded data of an audio encoder with associated metadata, such as spectral band replication (SBR) metadata. An audio decoder configured to determine a reconstructed frame of an audio signal from an access unit of a received data stream is described. The access unit comprises waveform data and metadata, wherein the waveform data and the metadata are associated with the same reconstructed frame of the audio signal. The audio decoder comprises a waveform processing path configured to generate a plurality of waveform subband signals from the waveform data, and a metadata processing path configured to generate decoded metadata from the metadata.

Abstract: Provided are a signal processing method and apparatus for enhancing sound quality. The signal processing method performed by a signal transmitting apparatus includes determining, based on a plurality of parameters, a valid bandwidth so as to encode an input signal; performing pre-processing on the input signal, based on the valid bandwidth; and encoding the pre-processed input signal, based on the valid bandwidth, and the signal processing method performed by a signal receiving apparatus includes decoding a bitstream or a packet received via a transmission channel; determining a valid bandwidth, based on a plurality of parameters used in the decoding; and performing post-processing on a decoded signal, based on the valid bandwidth.

Abstract: A coder and decoder, and methods therein, are provided for coding and decoding of spectral peak positions in audio coding. According to a first aspect, an audio signal segment coding method is provided for coding of spectral peak positions. The method comprises determining which one out of two lossless spectral peak position coding schemes that requires the least number of bits to code the spectral peak positions of an audio signal segment; and selecting the spectral peak position coding scheme that requires the least number of bits to code the spectral peak positions of the audio signal segment. A first one of the two lossless spectral peak position coding schemes is suitable for periodic or semi-periodic spectral peak position distributions; and a second one of two lossless spectral peak position coding schemes is suitable for sparse spectral peak position distributions.

Abstract: An apparatus includes a receiver and a decoder. The receiver is configured to receive a bitstream that includes an encoded mid channel and a quantized value representing a shift between a reference channel associated with an encoder and a target channel associated with the encoder. The quantized value is based on a value of the shift. The value of the shift is associated with the encoder and has a greater precision than the quantized value. The decoder is configured to decode the encoded mid channel to generate a decoded mid channel and to generate a first channel based on the decoded mid channel. The decoder is further configured to generate a second channel based on the decoded mid channel and the quantized value. The first channel corresponds to the reference channel and the second channel corresponds to the target channel.

Abstract: Systems and methods for generating a haptic output from an audio signal having a continuous stream of sampled digital audio data are provided. A haptic processing system receives the digital audio data, analyses the digital audio data for processing and extracts haptic signals for generating a haptic effect through an actuator. The method includes passing the digital audio signal on through dynamic processor(s), adjusting the dynamic range of the digital audio signal, extracting the signal envelope of the audio data, synthesising low-frequency signals from the extracted signal envelope, and enhancing the low-frequency content using a resonator. The haptic output is generated by mixing the digital audio signal with outputs from the different modules of the haptic processing system. An analytics module monitors, controls and adjusts the processing of the digital audio signal at the noise gate module, the compressor module and the envelope module to enhance the haptic output.

Abstract: A system and method can determine a time-varying position of a sound in a multi-channel audio signal. At least one processor can: receive a multi-channel audio signal representing a sound, each channel of the multi-channel audio signal providing audio associated with a corresponding channel position around a perimeter of a soundstage; determine a time-varying volume level for each channel of the multi-channel audio signal; determine, from the time-varying volume levels and the channel positions, a time-varying position in the soundstage of the sound; and generate a location data signal representing the time-varying position of the sound. The channel positions can be time-invariant. The position magnitude can be scaled to provide a unit magnitude as a sound pans from a channel to an adjacent channel. The position azimuth angle can be scaled to account for center location bias.

Abstract: A decoder for decoding a plurality of spectral-domain audio samples is provided. The decoder includes a first decoding module for generating a first group and a second group of time-domain intermediate audio samples from the spectral-domain audio samples. Moreover, the decoder includes an overlap-adder for overlap-adding the first group of time-domain intermediate audio samples with an overlap of more than 5% and at most 50% with the second group of time-domain intermediate audio samples. Furthermore, the decoder includes a second decoding module for generating a third group and a fourth group of time-domain intermediate audio samples from the spectral-domain audio samples. Moreover, the decoder includes an output interface.