Abstract: An apparatus, method, and computer-readable storage medium that generate at least a cue point in a musical piece. The method includes generating a beat grid representing the musical piece, determining values for the beat grid, the values corresponding to an audio feature of the musical piece, and each value representing an entire duration of each beat in the beat grid of the musical piece, calculating a score for the audio feature at each of a plurality of positions in the beat grid of the musical piece, using some or all of the determined values, and generating the cue point at a particular position of the plurality of positions, based on the calculated scores.

Abstract: Systems, methods, and devices for monitoring industrial equipment using audio are described herein. One system includes two computing devices. The first computing device can receive, from an audio sensor, audio sensed during operation of industrial equipment, extract a plurality of features from the audio, determine whether any portion of the audio is anomalous, and send, upon determining a portion of the audio is anomalous, the anomalous portion of the audio to the second, remotely located, computing device. The second computing device can provide the anomalous portion of the audio to a user to determine whether the anomalous portion of the audio corresponds to a fault occurring in the equipment, and receive, from the user upon determining the anomalous portion of the audio corresponds to a fault occurring in the equipment, input indicating the anomalous portion of the audio corresponds to the fault to learn fault patterns in the equipment.

Abstract: An encoding method for a multi-channel audio signal, an encoding apparatus for performing the encoding method, and a decoding method for a multi-channel audio signal and a decoding apparatus for performing the decoding method are disclosed. A method and apparatus of bypassing an MPEG Surround (MPS) standard operation and using an arbitrary tree when a number of audio signals of N channels exceeds a channel number defined in an MPS standard, is disclosed.

Abstract: A method is provided for extending the frequency band of an audio signal during a decoding or improvement process. The method includes obtaining the decoded signal in a first frequency band, referred to as a low band. Tonal components and a surround signal are extracted from the signal from the low-band signal, and the tonal components and the surround signal are combined by adaptive mixing using energy-level control factors to obtain an audio signal, referred to as a combined signal. The low-band decoded signal before the extraction step or the combined signal after the combination step are extended over at least one second frequency band which is higher than the first frequency band. Also proved are a frequency-band extension device which implements the described method and a decoder including a device of this type.

Abstract: The present document describes a method (400) for encoding a soundfield representation (SR) input signal (101, 301) describing a soundfield at a reference position, wherein the SR input signal (101, 301) comprises a plurality of channels for a plurality of different directivity patterns of the soundfield at the reference position. The method (400) comprises extracting (401) one or more audio objects (103, 303) from the SR input signal (101, 301). Furthermore, the method (400) comprises determining (402) a residual signal (102, 302) based on the SR input signal (101, 301) and based on the one or more audio objects (103, 303). The method (400) also comprises performing joint coding of the one or more audio objects (103, 303) and/or the residual signal (102, 302). In addition, the method (400) comprises generating (403) a bitstream (701) based on data generated in the context of joint coding of the one or more audio objects (103, 303) and/or the residual signal (102, 302).

Abstract: A method for recovering audio signals, a terminal and a storage medium are provided. The method includes: buffering an audio signal sampled at a preset number of sampling points each time, and then performing frequency spectrum analysis on the sampled audio signal by FFT; when it is determined that the audio signal is compressed, filtering a frequency point; recovering high-frequency signals based on audio signals before the frequency point; and performing phase recovery on the high-frequency signals. Thus, compressed high-frequency signals in the audio signals may be recovered.

Abstract: An audio signal processing device comprises: a receiver for receiving an input audio signal; a processor for generating loudness metadata corresponding to the input audio signal; and an outputter for transmitting the loudness metadata generated by the processor. The processor is configured to acquire loudness information analyzed from input content, acquires loudness information about the input audio signal by measuring the loudness of the input audio signal, generates the loudness metadata by converting the loudness information, and transmits, through the outputter, the generated loudness metadata to an output device for outputting the input audio signal.

Abstract: According to an embodiment, the above-described specification discloses an electronic device comprises at least one processor configured to: receive a first audio signal and a second audio signal; detect a spectral envelope signal from the first audio signal and extract a feature point from the second audio signal; extend a high-band of the second audio signal based on the spectral envelope signal from the first audio signal and the feature point from the second audio signal to generate a high-band extension signal; and mix the high-band extension signal and the first audio signal, thereby resulting in a synthesized signal.

Abstract: The disclosed embodiments relate to media devices implementing new television standards, such as ATSC 3.0, which includes audio/video essence and metadata/signaling. The disclosed embodiments include techniques for creating a signaling file that contains metadata and signaling data about the broadcast service being presented. The signaling file may include URLs that can be used to access signaling information for supplementary signaling and content. The signaling also contains a mapping between a first timing system and a second timing system. The first timing system may comprise the watermark timeline represented in interval s and the second timing system may comprise a DASH presentation time.

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

Abstract: An apparatus for decoding an encoded audio signal, includes a spectral domain audio decoder for generating a first decoded representation of a first set of first spectral portions, the decoded representation having a first spectral resolution; a parametric decoder for generating a second decoded representation of a second set of second spectral portions having a second spectral resolution being lower than the first spectral resolution; a frequency regenerator for regenerating every constructed second spectral portion having the first spectral resolution using a first spectral portion and spectral envelope information for the second spectral portion; and a spectrum time converter for converting the first decoded representation and the reconstructed second spectral portion into a time representation.

Abstract: There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary encoding method includes inter alia calculating M downmix signals by forming combinations of N audio objects, wherein M?N, and calculating parameters which allow reconstruction of a set of audio objects formed on basis of the N audio objects from the M downmix signals. The calculation of the M downmix signals is made according to a criterion which is independent of any loudspeaker configuration.

Abstract: In general, techniques are described by which to provide priority information for higher order ambisonic (HOA) audio data. A device comprising a memory and a processor may perform the techniques. The memory stores HOA coefficients of the HOA audio data, the HOA coefficients representative of a soundfield. The processor may decompose the HOA coefficients into a sound component and a corresponding spatial component, the corresponding spatial component defining shape, width, and directions of the sound component, and the corresponding spatial component defined in a spherical harmonic domain. The processor may also determine, based on one or more of the sound component and the corresponding spatial component, priority information indicative of a priority of the sound component relative to other sound components of the soundfield, and specify, in a data object representative of a compressed version of the HOA audio data, the sound component and the priority information.

Abstract: An encoding method for a wireless local area network includes encoding a packet via a cyclic coding scheme into a first amount of multiple first blocks; processing each of the first amount of the first blocks into a second amount of multiple second blocks; and encoding the second amount of the second blocks via a block coding scheme.

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

Abstract: Systems and methods include a plurality of audio input components configured to generate a plurality of audio input signals, and a logic device configured to receive the plurality of audio input signals, determine whether the plurality of audio signals comprise target audio associated with an audio source, estimate a relative location of the audio source with respect to the plurality of audio input components based on the plurality of audio signals and a determination of whether the plurality of audio signals comprise the target audio, and process the plurality of audio signals to generate an audio output signal by enhancing the target audio based on the estimated relative location. The logic device is further configured to use relative transfer-based covariance to construct directional covariance matrix aligned across frequency bands and find a direction that minimizes beam power subject to distortionless criteria.

Abstract: The techniques disclosed herein provide apparatuses and related methods for the communication of spatial audio and related metadata. In some implementations, a source provides prerecorded spatial audio that has embedded metadata. A computing device processes the prerecorded spatial audio to generate an audio codec that is segmented to include a first section of audio data and a second section that includes metadata extracted from the prerecorded spatial audio. The generated audio codec may be received by a device that includes an encoder. The encoder may process the generated audio codec to generate audio data that includes the metadata.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for suppressing hotword triggers when detecting a hotword in recorded media are disclosed. In one aspect, a method includes the actions of receiving, by a computing device, audio corresponding to playback of an item of media content. The actions further include determining, by the computing device, that the audio includes an utterance of a predefined hotword and that the audio includes an audio watermark. The actions further include analyzing, by the computing device, the audio watermark. The actions further include based on analyzing the audio watermark, determining, by the computing device, whether to perform speech recognition on a portion of the audio following the predefined hotword.

Abstract: This disclosure provides an encoding method, a decoding method, an encoding apparatus, and a decoding apparatus for a stereo signal. The encoding method includes: performing interpolation processing based on the inter-channel time difference in the current frame and an inter-channel time difference in a previous frame of the current frame; performing time-domain downmixing processing on the stereo signal after the delay alignment in the current frame, to obtain a primary-channel signal and a secondary-channel signal in the current frame; and quantizing the inter-channel time difference after the interpolation processing in the current frame, the primary channel signal and the secondary channel signal.

Abstract: An encoding device according to the disclosure includes a first encoding unit that generates a first encoded signal in which a low-band signal having a frequency lower than or equal to a predetermined frequency from a voice or audio input signal is encoded, and a low-band decoded signal; a second encoding unit that encodes, on the basis of the low-band decoded signal, a high-band signal having a band higher than that of the low-band signal to generate a high-band encoded signal; and a first multiplexing unit that multiplexes the first encoded signal and the high-band encoded signal to generate and output an encoded signal. The second encoding unit calculates an energy ratio between a high-band noise component, which is a noise component of the high-band signal, and a high-band non-tonal component of a high-band decoded signal generated from the low-band decoded signal and outputs the ratio as the high-band encoded signal.

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: Methods for converting, encoding, decoding and transcoding an acoustic field, more particularly a first-order Ambisonics three-dimensional acoustic field.

Abstract: A sound processing apparatus includes: a conversion unit that samples predetermined audio data with a first quantization bit rate and a second quantization bit rate greater than the first quantization bit rate and outputs the respective results as first digital audio data and second digital audio data; a frequency analysis unit that performs frequency analysis on each output from the conversion unit; and a determination unit that performs predetermined determination processing on the basis of analysis results by the frequency analysis unit.

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

Abstract: A method for generating loudspeaker signals associated with a target screen size is disclosed. The method includes receiving a bit stream containing encoded higher order ambisonics signals, the encoded higher order ambisonics signals describing a sound field associated with a production screen size. The method further includes decoding the encoded higher order ambisonics signals to obtain a first set of decoded higher order ambisonics signals representing dominant components of the sound field and a second set of decoded higher order ambisonics signals representing ambient components of the sound field. The method also includes combining the first set of decoded higher order ambisonics signals and the second set of decoded higher order ambisonics signals to produce a combined set of decoded higher order ambisonics signals.

Abstract: An apparatus configured to: based on (i) captured spatial audio content of a scene comprising audio that is associated with information indicative of at least a direction in the scene from which said audio was captured; and (ii) visual focus information comprising information indicative of at least a first part of the scene on which corresponding captured visual imagery of the scene is focused for presentation to a user; provide for presentation of the captured spatial audio content to accompany the captured visual imagery, the captured spatial audio content presented as spatial audio, the spatial audio content provided for presentation with a spatial audio focus selectively applied to audio captured from a second part of the scene different to the first part, the spatial audio focus comprising an audio-modifying effect to increase the audibility of the audio having a direction corresponding to the second part.

Abstract: In methods and apparatus and non-transitory memory units for encoding/decoding audio signal information, the encoder side may determine if a signal frame is useful for long term post filtering and/or packet lost concealment and may encode information in accordance to the results of the determination, and the decoder side may apply the LTPF and/or PLC in accordance to the information obtained from the encoder.

Abstract: Audio video synchronization and alignment or alignment of audio to some other external clock are rendered more effective or easier by treating fragment grid and frame grid as independent values, but, nevertheless, for each fragment the frame grid is aligned to the respective fragment's beginning. A compression effectiveness lost may be kept low when appropriately selecting the fragment size. On the other hand, the alignment of the frame grid with respect to the fragments' beginnings allows for an easy and fragment-synchronized way of handling the fragments in connection with, for example, parallel audio video streaming, bitrate adaptive streaming or the like.

Abstract: A method for processing a monophonic signal in a 3D audio decoder, including processing binauralizing decoded signals intended to be delivered spatially by a headset. The method is such that, on detection, in a datastream representative of the monophonic signal, of an indication of non-binauralization processing, which indication is associated with spatial delivery position information, the decoded monophonic signal is directed to a stereophonic rendering engine, which takes into account the position information to construct two delivery channels that are directly processed via a direct mixing that sums these two channels with a binauralized signal output from the binauralization processing, in order to be delivered via the headset. A decoder device that implements the processing method is also provided.

Abstract: To suitably regulate sound pressure of object content on a receiving side. An audio stream including coded data of a predetermined number of pieces of object content is generated. A container of a predetermined format including the audio stream is transmitted. Information indicating a range within which sound pressure is allowed to increase and decrease for each piece of object content is inserted into a layer of the audio stream and/or a layer of the container. On a receiving side, sound pressure of each piece of object content increases and decreases within the allowable range based on the information.

Abstract: A low power voice processing system that includes a plurality of non-audio sensors, at least one microphone system, and a plurality of audio modules, at least some of which can be configured in selected modes. A context determination module is connected to the plurality of audio modules, and further connected to receive input from the plurality of non-audio sensors and the at least one microphone system. The context determination module acts to determine use context for the voice processing system and at least in part selects mode operation of at least some of the plurality of audio modules.

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 processing text information is provided, the method including: obtaining text information and extracting text characters contained in the text information; extracting, from the text characters, target characters satisfying a predetermined rule; calculating a filtering index of the text information according to the target characters; and when the filtering index meets a predetermined condition, executing an operation corresponding to the predetermined condition on the text information. In addition, an apparatus for processing text information is further provided. The method and apparatus for processing text information can improve the accuracy and efficiency of filtering out junk text information.

Abstract: System and method that handles the inconsistent processing of interactive media (e.g. digital games, music visualization software, video playback software, animation software, etc.) and the generally consistent timeline delay to enable synchronization of messages and events with the playback of potentially variable-rate timelines.

Abstract: A method of predicting a channel parameter of an original signal from a downmix signal is disclosed. The method may include generating an input feature map to be used to predict a channel parameter of the original signal based on a downmix signal of an original signal, determining an output feature map including a predicted parameter to be used to predict the channel parameter by applying the input feature map to a neural network, generating a label map including information associated with the channel parameter of the original signal, and predicting the channel parameter of the original signal by comparing the output feature map and the label map.

Abstract: A system may include at least one processor. The at least one processor may receive data from a plurality of independent data sources. The data from each respective data source is received at a rate determined by the respective data source. The at least one processor may further write the received data to at least one data store at a rate independent of the respective rates at which data from the plurality of independent data sources is received. A method and computer-readable medium are also disclosed.

Abstract: The present disclosure relates to sampling wireless signals received at a receiver. Subsets of the digital samples are multiplied by a Pseudo-Noise (PN) code to generate tone signals. A pattern of tone signals may indicate the presence of a repeating preamble. This may be used to locate a payload that follows the preamble. A tone signal of a payload may be decoded by multiplying a subset of the digital samples with the PN code to generate a tone signal. The tone signal may be transformed into the frequency domain to identify a frequency component having an energy level above a threshold. Using the frequency component, the payload is decoded. By structuring a packet as a series of PN-code modulated tone signals, packets received from different sources may be differentiated even when they arrive at overlapping points of time. This allows for a larger transmitting capacity in a network.

Abstract: The present invention pertains to an audio playback method and an audio playback apparatus in a 6DoF environment. The audio playback method of the present invention is characterised by comprising: a decoding step of decoding a received audio signal, and outputting the decoded audio signal and metadata; a modelling step of receiving input of position information of a user, checking whether the position of the user has changed from a previous position, and if the position of the user has changed, modelling binaural rendering data so as to correspond to the changed position of the user; and a rendering step of binaural-rendering the decoded audio signal using the modelled rendering data, and outputting the same as a two-channel audio signal.

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: The present invention provides an IP telephone and the like capable of preventing a loss of the beginning part of a talk. A communication apparatus according to the present invention includes: a memory (110) configured to store communication data with a communication destination; and a control unit (120) configured to transmit and receive the communication data to and from the communication destination in each first duration and store the transmitted and received communication data in the memory (110), then output, after a communication session with an external recording apparatus that records the communication data is established, the communication data stored in the memory (110) to the external recording apparatus in each second duration that is shorter than the first duration.

Abstract: A method for training hotword detection includes receiving a training input audio sequence including a sequence of input frames that define a hotword that initiates a wake-up process on a device. The method also includes feeding the training input audio sequence into an encoder and a decoder of a memorized neural network. Each of the encoder and the decoder of the memorized neural network include sequentially-stacked single value decomposition filter (SVDF) layers. The method further includes generating a logit at each of the encoder and the decoder based on the training input audio sequence. For each of the encoder and the decoder, the method includes smoothing each respective logit generated from the training input audio sequence, determining a max pooling loss from a probability distribution based on each respective logit, and optimizing the encoder and the decoder based on all max pooling losses associated with the training input audio sequence.

Abstract: A gain adjustment apparatus for use in decoding of audio that has been encoded with separate gain and shape representations includes an accuracy meter configured to estimate an accuracy measure of the shape representation, and to determine a gain correction based on the estimated accuracy measure. An envelope adjuster further included in the apparatus is configured to adjust the gain representation based on the determined gain correction.

Abstract: Methods, apparatus and articles of manufacture to identify sources of network streaming services are disclosed. An example method includes receiving a first audio signal that represents a decompressed second audio signal, identifying, from the first audio signal, a parameter of an audio compression configuration used to form the decompressed second audio signal, and identifying a source of the decompressed second audio signal based on the identified audio compression configuration.

Abstract: An audio encoder for encoding an audio signal has: a first encoding processor for encoding a first audio signal portion in a frequency domain, having: a time frequency converter for converting the first audio signal portion into a frequency domain representation; an analyzer for analyzing the frequency domain representation to determine first spectral portions to be encoded with a first spectral resolution and second regions to be encoded with a second resolution; and a spectral encoder for encoding the first spectral portions with the first spectral resolution and encoding the second portions with the second resolution; a second encoding processor for encoding a second different audio signal portion in the time domain; a controller for analyzing and determining, which portion of the audio signal is the first audio signal portion encoded in the frequency domain and which portion is the second audio signal portion encoded in the time domain; and an encoded signal former for forming an encoded audio signal havi

Abstract: An apparatus for decoding an encoded audio signal, includes: a spectral domain audio decoder for generating a first decoded representation of a first set of first spectral portions being spectral prediction residual values; a frequency regenerator for generating a reconstructed second spectral portion using a first spectral portion of the first set of first spectral portions, wherein the reconstructed second spectral portion additionally includes spectral prediction residual values; and an inverse prediction filter for performing an inverse prediction over frequency using the spectral residual values for the first set of first spectral portions and the reconstructed second spectral portion using prediction filter information included in the encoded audio signal.

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: 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: 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.