Abstract: A method may include obtaining audio to be projected in an environment in which the audio includes a plurality of audio channels. The method may include mapping a first audio channel of the plurality of audio channels to a first channel object, the first channel object including first audio of the first audio channel. The method may include obtaining environmental parameters associated with a speaker system including a plurality of speakers, the environmental parameters including one or more of: speaker locations, sensor information, speaker acoustic properties, environmental acoustic properties, environment geometry, or listener location. The method may include obtaining a first target sound effect associated with the first audio channel. The method may include directing projection of the first channel object by a speaker of the plurality of speakers according to the first target sound effect and based on the environmental parameters to simulate the first target sound effect.

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: Apparatus, systems, articles of manufacture, and methods are disclosed for multiple scrambled layers for audio watermarking.

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 indication may be received, by a storage service, from an account, of at least a first object and a first threshold time duration, wherein the first object is to be automatically transferred from a lower-latency storage tier to a higher-latency storage tier based on object access history data and the first threshold time duration. A first determined time duration since a most recent access of the first object may be determined based on the object access history data. The first determined time duration may be compared to the first threshold time duration. It may be determined that the first determined time duration meets or exceeds the first threshold time duration. The first object may be transferred from the lower-latency storage tier to the higher-latency storage tier based on the first determined time duration meeting or exceeding the first threshold time duration.

Abstract: The present invention relates to a generation of a signal by executing a waveform dataset comprising waveform descriptive parameters. The execution of the waveform description parameters is limited by target device information specifying one or more specific target devices and time information specifying an execution period of the waveform descriptive parameters. By providing a waveform dataset comprising not only the waveform descriptive parameters, but also further information, in particular time information for limiting the execution period of the waveform descriptive parameters, the generation of the respective waveform signal is controlled.

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

Abstract: An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, includes a signal analyzer for estimating a signal characteristic of the first channel signal or the second channel signal or both signals or a signal derived from the first channel signal or the second channel signal; a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block; a weighter for weighting a smoothed or non-smoothed cross-correlation spectrum to obtain a weighted cross correlation spectrum using a first weighting procedure or using a second weighting procedure depending on a signal characteristic estimated by the signal analyzer, wherein the first weighting procedure is different from the second weighting procedure; and a processor for processing the weighted cross-correlation spectrum to obtain the inter-channel time difference.

Abstract: A stereo signal encoding method includes obtaining indication information of an encoding mode of a residual signal of a current frame, where the indication information includes at least one of an encoding status of a residual signal of a previous frame, a value of an updating manner flag for a long-term smooth parameter of the current frame, or a value of a status change parameter of the current frame relative to a stereo signal of the previous frame, and determining the encoding mode of the residual signal of the current frame based on the indication information of the encoding mode of the residual signal of the current frame, where the encoding mode indicates whether to encode the residual signal of the current frame.

Abstract: A data transcoding device includes a memory device for storing clear data containing private information and a processor configured as a data transcoder. The processor is configured to create packets of the clear data, prepare the packets for transcoding the clear data into an indecipherable multimedia data file appearing as noise, by determining properties of the indecipherable multimedia file based on parameters of the clear data. The processor is configured to generate the indecipherable multimedia file by transcoding the clear data based on the determined properties.

Abstract: An audio system has an amplifier for driving an audio actuator and includes a switched-mode power supply that draws power from a power source (e.g., battery) to supply power to the amplifier, a capacitor charged by the switched-mode power supply to supply power to the amplifier, and a feed-forward compressor that performs dynamic range compression of an audio input to provide an audio output for amplification by the amplifier. The compressor includes a sidechain frequency-biasing filter that generates a frequency-biased version of the audio input that is attenuated as frequency increases which causes the compressor to decrease the compression as frequency increases. A control block limits current drawn from the battery by the switched-mode power supply independent of audio input frequency, but the frequency-biasing filter enables the amplifier to service audio power transients greater than the current-limited power supply can supply by advantageously concurrently sourcing extra power from the capacitor.

Abstract: Methods and audio processing units for generating an object based audio program including conditional rendering metadata corresponding to at least one object channel of the program, where the conditional rendering metadata is indicative of at least one rendering constraint, based on playback speaker array configuration, which applies to each corresponding object channel, and methods for rendering audio content determined by such a program, including by rendering content of at least one audio channel of the program in a manner compliant with each applicable rendering constraint in response to at least some of the conditional rendering metadata. Rendering of a selected mix of content of the program may provide an immersive experience.

Abstract: In methods and systems for filtering an information input signal, a system may have: a first filter unit filtering an input signal at an initial subinterval in a current update interval according to parameters associated to the preceding update interval, the parameters being scaled by a first scaling factor changing towards 0; and a second filter unit filtering a second filter input signal, based on the output of the first filter unit, at the initial subinterval, according to parameters associated to the current update interval, the parameters being scaled by a second scaling factor changing from 0, or a value close to 0, toward a value more distant from 0.

Abstract: A speaker recognition system for authenticating a mobile device user includes an enrollment and learning software module, a voice biometric authentication software module, and a secure software application. Upon request by a user of the mobile device, the enrollment and learning software module displays text prompts to the user, receives speech utterances from the user, and produces a voice biometric print. The enrollment and training software module determines when a voice biometric print has met at least a quality threshold before storing it on the mobile device. The secure software application prompts a user requiring authentication to repeat an utterance based at least on an attribute of a selected voice biometric print, receives a corresponding utterance, requests the voice biometric authentication software module to verify the identity of the second user using the utterance, and, if the user is authenticated, imports the voice biometric print.

Abstract: In general, techniques are described by which to correlate scene-based audio data for psychoacoustic audio coding. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may store a bitstream including a plurality of encoded correlated components of a soundfield represented by scene-based audio data. The one or more processors may perform psychoacoustic audio decoding with respect to one or more of the plurality of encoded correlated components to obtain a plurality of correlated components, and obtain, from the bitstream, an indication representative of how the one or more of the plurality of correlated components were reordered in the bitstream. The one or more processors may reorder, based on the indication, the plurality of correlated components to obtain a plurality of reordered components, and reconstruct, based on the plurality of reordered components, the scene-based audio data.

Abstract: The present disclosure relates to an apparatus for decoding an encoded Unified Audio and Speech stream. The apparatus comprises a core decoder for decoding the encoded Unified Audio and Speech stream. The core decoder includes a fast Fourier transform, FFT, module implementation based on a Cooley-Tuckey algorithm. The FFT module is configured to determine a discrete Fourier transform, DFT. Determining the DFT involves recursively breaking down the DFT into small FFTs based on the Cooley-Tucker algorithm and using radix-4 if a number of points of the FFT is a power of 4 and using mixed radix if the number is not a power of 4. Performing the small FFTs involves applying twiddle factors. Applying the twiddle factors involves referring to pre-computed values for the twiddle factors.

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. The high frequency regeneration is performed as a post-processing operation with a delay of 3010 samples per audio channel.

Abstract: Systems and methods for packet payload mapping for robust transmission of data are described. For example, methods may include receiving, using a network interface, packets that each respectively include a primary frame and one or more preceding frames from the sequence of frames of data that are separated from the primary frame in the sequence of frames by a respective multiple of a stride parameter; storing the frames of the packets in a buffer with entries that each hold the primary frame and the one or more preceding frames of a packet; reading a first frame from the buffer as the primary frame from one of the entries; determining that a packet with a primary frame that is a next frame in the sequence has been lost; and, responsive to the determination, reading the next frame from the buffer as a preceding frame from one of the entries.

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 method for a communication protocol that includes wideband access for low latency audio. A source device and a sink device establish a wireless communication link. The protocol for communicating via the wireless communication link includes a frame having a first deterministic portion and a second opportunistic portion. The source device transmits during the first deterministic portion of the frame, first data to the sink device. The source device determines whether a predetermined condition associated with the wireless communication link is satisfied. When the predetermined condition is satisfied, the source device transmits, during the second opportunistic portion of the frame, second data to the sink device.

Abstract: Systems, methods, and computer program products for determining an attack on a neural network. A data sample is received at a first classifier neural network and at a watermark classifier neural network, wherein the first classifier neural network is trained using a first dataset and a watermark dataset. The first classifier neural network determines a classification label for the data sample. A watermark classifier neural network determines a watermark classification label for the data sample. A data sample is determined as an adversarial data sample based on the classification label for the data sample and the watermark classification label for the data sample.

Abstract: A decoding device comprising a decoding unit configured to decode a tactile signal encoded for each of frequency bands. A decoding method comprising decoding a tactile signal encoded for each of frequency bands. A non-transitory storage medium encoded with instructions that, when executed by a computer, execute processing comprising decoding a tactile signal encoded for each of frequency bands.

Abstract: Embodiments provide a transfer method for wirelessly transferring data in a communication system (e.g. a sensor network or telemetry system). The data includes core data and extension data, wherein the core data is encoded and distributed in an interleaved manner to a plurality of core sub-data packets, wherein the extension data is encoded and distributed in an interleaved manner to a plurality of extension sub-data packets, wherein at least a part of the core data contained in the core sub-data packets is needed for receiving the extension data or extension data packets.

Abstract: This application provides a data transmission method and apparatus. The method includes: obtaining significant bits in an Nth data frame, where N is an integer greater than or equal to 1, a quantity of the significant bits in the Nth data frame is smaller than a quantity of bits in the Nth data frame, and remaining bits other than the significant bits in the Nth data frame are insignificant bits in the Nth data frame; and sending the significant bits in the Nth data frame, and skipping sending the insignificant bits in the Nth data frame. According to the application, on a premise of ensuring voice intelligibility, a transmit end apparatus transmits only the significant bits, to minimize a data transmission amount, thereby improving transmission reliability and coverage.

Abstract: Apparatus for generating an enhanced sound field description, including: a sound field generator for generating at least one sound field description indicating a sound field with respect to at least one reference location; and a meta data generator for generating meta data relating to spatial information of the sound field, wherein the at least one sound field description and the meta data constitute the enhanced sound field description. The at least one sound field description includes in a certain case a first sound field description related to the at least one reference location and a second sound field description related to a further reference location being different from the reference location, and the meta data relating to the spatial information indicates the reference location and the further reference location or one of both reference locations and a vector extending between both reference locations.

Abstract: Method and apparatus are provided for reconstructing a noise component of a speech/audio signal. A bitstream is received and decoded to obtain a speech/audio signal. A first speech/audio signal is determined according to the speech/audio signal. A symbol of each sample value in the first speech/audio signal and an amplitude value of each sample value in the first speech/audio signal is determined. An adaptive normalization length and an adjusted amplitude value of each sample value are determined according to the adaptive normalization length and the amplitude value of each sample value. A second speech/audio signal is determined according to the symbol of each sample value and the adjusted amplitude value of each sample value.

Abstract: A method, apparatus and system for implementing pre-equalization equalizer tap analysis and correlation in a DOCSIS 3.1 network environment. The disclosed principles improve the pre-equalization analysis in the DOCSIS 3.1 environment by filtering out short distance reflections, which is required for the proper grouping and correlation of modems.

Abstract: In one aspect, an example method to be performed by a vehicle-based media system includes (a) receiving audio content; (b) causing one or more speakers to output the received audio content; (c) using a microphone of the vehicle-based media system to capture the output audio content; (d) identifying reference audio content that has at least a threshold extent of similarity with the captured audio content; (e) identifying a geographic location associated with the identified reference audio content; and (f) based at least on the identified geographic location associated with the identified reference audio content, outputting, via the user interface of the vehicle-based media system, a prompt to navigate to the identified geographic location.

Abstract: A method for synchronizing an electronic interactive device on basis of a first sound track is provided. The method may include identifying a first peak point and a valley point of the first sound track by calculating a first energy of the first peak point and a first energy of the valley point of the first peak point and comparing the first energy of the first peak point with first energy of neighboring points of the first peak point. The method may also include identifying a first peak point of a second soundtrack, and determining a similarity between the first soundtrack and a second sound track on basis of the first peak point of the first soundtrack and the first peak point of the second sound track.

Abstract: A video encoding and decoding system that implements an adaptive transfer function method internally within the codec for signal representation. A focus dynamic range representing an effective dynamic range of the human visual system may be dynamically determined for each scene, sequence, frame, or region of input video. The video data may be cropped and quantized into the bit depth of the codec according to a transfer function for encoding within the codec. The transfer function may be the same as the transfer function of the input video data or may be a transfer function internal to the codec. The encoded video data may be decoded and expanded into the dynamic range of display(s). The adaptive transfer function method enables the codec to use fewer bits for the internal representation of the signal while still representing the entire dynamic range of the signal in output.

Abstract: There is disclosed inter alia an apparatus comprising means for receiving an audio format comprising a plurality of individual audio signal streams and metadata, wherein the metadata comprises a dependency field associated with each of the plurality of individual audio signal streams; means for determining that a dependency field associated with a first individual audio signal stream of the plurality of individual audio signal streams indicates that the first individual audio signal stream is related to a second individual audio signal stream of the plurality of individual audio signal streams; and means for encoding the first and second individual audio signal streams as a combined multichannel audio signal by an audio encoder.

Abstract: A device includes a receiver and a decoder. The receiver is configured to receive bitstream parameters corresponding to at least an encoded mid signal. The decoder is configured to generate a synthesized mid signal based on the bitstream parameters. The decoder is also configured to generate a synthesized side signal selectively based on the bitstream parameters in response to determining whether the bitstream parameters correspond to an encoded side signal.

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: It is an object of the present invention to play back other content on any timing without being restricted to the embedded timing of watermark data embedded in the content even when the progress time of the content fluctuates.

Abstract: Methods and systems are disclosed for detecting threats in voice communications such as telephone calls. Various voice phishing (vishing) detectors detect respective type of threats and can be used or activated individually or in various combinations. A tampering detector utilizes deep scattering spectra and shifted delta cepstra features to detect tampering in the form of voice conversion, speech synthesis, or splicing. A content detector predicts a likelihood that word patterns on an incoming voice signal are indicative of a vishing threat. A spoofing detector authenticates or repudiates a purported speaker based on comparison of voice profiles. The vishing detectors can be provided as an authentication service or embedded in communication equipment. Machine learning and signal processing aspects are disclosed, along with applications to mobile telephony and call centers.

Abstract: Systems and methods for detecting and acknowledging audio transmissions containing data. In one embodiment, a method is presented that includes receiving multiple audio signals that are detected by multiple receivers from within a service area. A first audio transmission may be detected in a first subset of the audio signal that are received by a first subset of the receivers. The first subset of the receivers may be positioned to receive audio transmissions from computing devices located within a first portion of the service area. At least one transmitter may be identified that is positioned to transmit audio transmissions to computing devices located within at least a subset of the first portion of the service area. A second audio transmission may be transmitted using the at least one first transmitter.

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: 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: Methods, apparatus, systems and articles of manufacture are disclosed to identify media that has been pitch shifted, time shifted, and/or resampled. An example method includes: generating, by executing an instruction with a processor, a fingerprint from an audio signal; transmitting the fingerprint and adjusting instructions to a central facility to facilitate a query, the adjusting instructions identifying at least one of a pitch shift or a time shift; and receiving a response including an identifier for the audio signal and information corresponding to how the audio signal was adjusted; storing information indicative of the identifier and the information into a database.

Abstract: A system and a method for rendering an image including an image gateway arranged to receive content data representative of one or more objects and style data representative of one or more desired rendering styles; and an image manipulation engine arranged to process the content data and the style data to render a target image, wherein the target image includes a representation of the one or more objects and includes the one or more desired rendering styles.

Abstract: An inter-channel phase difference (IPD) parameter extraction method and apparatus, where the extraction method includes obtaining a parameter 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 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: A multi-channel signal encoding method includes determining a downmixed signal of a first channel signal and a second channel signal, 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, and 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: 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 method includes receiving a set of time domain samples representing audio captured using one or more microphones, and generating, from the time domain samples, a spectrum comprising a set of frequency domain coefficients, each coefficient representing a frequency bin corresponding to a range of frequencies. The method also includes adjusting the spectrum to generate a smoothed spectrum, wherein generating the smoothed spectrum includes determining that a magnitude of a first frequency domain coefficient is less than a threshold, and in response, replacing the first frequency domain coefficient by a value computed as a function of a plurality of frequency domain coefficients that include the first frequency domain coefficient. The method further includes generating an audio signal based on time domain samples computed using the smoothed spectrum.

Abstract: An apparatus for generating an error concealment signal, includes: an LPC representation generator for generating a replacement LPC representation; a gain calculator for calculating a gain information from the LPC representations; a compensator for compensating a gain influence of the replacement LPC representation using the gain information; and an LPC synthesizer for filtering codebook information using the replacement LPC representation to obtain the error concealment signal, wherein the compensator is configured for weighting the codebook information or an LPC synthesis output signal.

Abstract: An apparatus for encoding directional audio coding parameters having diffuseness parameters and direction parameters, has: a parameter quantizer for quantizing the diffuseness parameters and the direction parameters; a parameter encoder for encoding quantized diffuseness parameters and quantized direction parameters; and an output interface for generating an encoded parameter representation having information on encoded diffuseness parameters and encoded direction parameters.

Abstract: Example embodiments disclosed herein relate to audio object clustering. A method for metadata-preserved audio object clustering is disclosed. The method comprises classifying a plurality of audio objects into a number of categories based on information to be preserved in metadata associated with the plurality of audio objects. The method further comprises assigning a predetermined number of clusters to the categories and allocating an audio object in each of the categories to at least one of the clusters according to the assigning. Corresponding system and computer program product are also disclosed.

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.