Abstract: An audio processing system (100) accepts an audio bitstream having one of a plurality of predefined audio frame rates. The system comprises a front-end component (110), which receives a variable number of quantized spectral components, corresponding to one audio frame in any of the predefined audio frame rates, and performs an inverse quantization according to predetermined, frequency-dependent quantization levels. The front-end component may be agnostic of the audio frame rate. The audio processing system further comprises a frequency-domain processing stage (120) and a sample rate converter (130), which provide a reconstructed audio signal sampled at a target sampling frequency independent of the audio frame rate. By its frame-rate adaptability, the system can be configured to operate frame-synchronously in parallel with a video processing system that accepts plural video frame rates.

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 audio signal encoding and decoding method using a neural network model, and an encoder and decoder for performing the same are disclosed. A method of encoding an audio signal using a neural network model, the method may include identifying an input signal, generating a quantized latent vector by inputting the input signal into a neural network model encoding the input signal, and generating a bitstream corresponding to the quantized latent vector, wherein the neural network model may include i) a feature extraction layer generating a latent vector by extracting a feature of the input signal, ii) a plurality of downsampling blocks downsampling the latent vector, and iii) a plurality of quantization blocks performing quantization of a downsampled latent vector.

Abstract: An information processing apparatus includes a processor configured to: segment, into multiple voice segments, voice data and text data converted from the voice data; impart a security level to each of the voice segments in accordance with contents of the text data and the voice data in each of the voice segments; and perform control on an output of each of the voice segments in accordance with the security level.

Abstract: An improved gain-shape vector quantization is achieved by determining a number of bits to be allocated to a gain adjustment- and shape-quantizer for a plurality of combinations of a current bit rate and a first signal property. The bit allocation is derived by using an average of optimal bit allocations for a training data set. A number of bits to the gain adjustment and the shape quantizers for a plurality of combinations of the bit rate and a first signal are pre-calculated, and a table indicating the number of bits to be allocated to the gain adjustment- and the shape-quantizers for a plurality of combinations of the bit rate and a first signal property is created. In this way, the table can be used for achieving an improved bit allocation.

Abstract: In accordance with some embodiments of the disclosed subject matter, mechanisms for seamless audio melding between audio items in a playlist are provided.

Abstract: A method for encoding an audio signal includes obtaining an ITD value of a current frame of an audio signal; obtaining a characteristic parameter of the current frame, wherein the characteristic parameter comprises at least one of a signal-to-noise ratio of the current frame or a peak feature of cross-correlation coefficients of the current frame; determining, based on the characteristic parameter, whether to use the initial ITD value as the finalized ITD value of the current frame; and encoding the current frame based on the finalized ITD value of the current frame, wherein if the initial ITD value is determined as not used as the finalized ITD value of the current frame, the finalized ITD value of the current frame is the same as a finalized ITD value of a previous frame of the current frame.

Abstract: Provided is a technique according to which it is possible to obtain a decoded sound signal of high sound quality without significantly increasing the delay time compared to a configuration in which only a decoded sound signal of the minimum necessary sound quality is obtained. In a terminal apparatus connected to a first communication line and a second communication line with a lower priority level therethan, sound signals of multiple channels are obtained and output based on a monaural code included in a first code string input from the first communication line and an extended code included in a second code string with the closest frame number to that of the monaural code among extended codes included in the second code string input from the second communication line.

Abstract: An audio coding method includes obtaining a current frame of an audio signal, where the current frame includes a high frequency band signal; coding the high frequency band signal to obtain a coding parameter of the current frame, where coding includes tonal component screening, the coding parameter indicates information about a target tonal component of the high frequency band signal, the target tonal component is obtained after tonal component screening, and information about a tonal component includes location information, quantity information, and amplitude information or energy information of the tonal component; and performing bitstream multiplexing on the coding parameter to obtain a coded bitstream.

Abstract: In a method to decode signals, a computing device decodes spectral coefficients of a current frame are grouped into a plurality of sub-bands. The computing device classifies a sub-band as a bit allocation unsaturated sub-band based on an average quantity of allocated bits per spectral coefficient of a sub-band of the plurality of sub-bands and a threshold. The computing device obtains a noise filling gain based on an envelope of the sub-band, and obtains a reconstructed spectral coefficient of the sub-band by performing noise filling based on the noise filling gain. The computing device then obtains a frequency domain audio signal based on spectral coefficients in the sub-band obtained by decoding and the reconstructed spectral coefficient.

Abstract: Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.

Abstract: A method for encoding a high frequency signal includes determining a signal type of a high frequency signal of a current frame, smoothing and scaling time envelopes of the high frequency signal of the current frame and obtaining time envelopes of the high frequency signal of the current frame that require to be encoded when the high frequency signal of the current frame is a non-transient signal and a high frequency signal of the previous frame is a transient signal, and quantizing and encoding the time envelopes of the high frequency signal of the current frame that require to be encoded, and frequency information and signal type information of the high frequency signal of the current frame.

Abstract: The present invention discloses a waveform file processing method, storage medium, and device, wherein the method comprises a storage step, and such storage step comprises the following sub-steps: obtain a waveform file that comprises at least one waveform signal; assign a basic index value based on the waveform signal, and adopt the variable-length encoding method to, in a memory, encode the said waveform file as a resolvable serialized structure; when the memory consumed by the serialized structure reaches the threshold, trigger the compression and persistence for the current serialized structure, and obtain the waveform processing file. The present invention uses a unique organization mode to locally or remotely generate a waveform file of a specific format so that the efficiency of subsequent storage, reading, and debugging based on the waveform database file of the said format is significantly improved.

Abstract: System and for signal re-transmission including a channelizer, a signal-effect-processor and a controller. The signal-effect-processor includes a plurality of sub-band-processors and a summer. The channelizer receives a sampled Intermediate-Frequency signal exhibiting a first sampling-rate. The channelizer produces a plurality of sub-band-signals, each associated with a respective sub-band of the Intermediate-Frequency signal. Each sub-band-signal exhibit a second sampling-rate lower than the first sampling-rate. Each of at least one selected sub-band-processor receives a respective sub-band-signal, introduces at least one effect to the respective sub-band-signal, and increases the sampling-rate of the respective sub-band-signal to the first sampling-rate, thereby producing a respective affected sub-band re-transmit signal. Each selected sub-band-processor is further provides the respective affected sub-band re-transmit signal to a respective input of the summer.

Abstract: An encoder for encoding frequency transform coefficients of a harmonic audio signal include the following elements: A peak locator configured to locate spectral peaks having magnitudes exceeding a predetermined frequency dependent threshold. A peak region encoder configured to encode peak regions including and surrounding the located peaks. A low-frequency set encoder configured to encode at least one low-frequency set of coefficients outside the peak regions and below a crossover frequency that depends on the number of bits used to encode the peak regions. A noise-floor gain encoder configured to encode a noise-floor gain of at least one high-frequency set of not yet encoded coefficients outside the peak regions.

Abstract: A method for frequency measurement and related apparatuses are provided. The method includes the following. M first frequencies are measured on condition that a user equipment has a power-saving requirement. Electricity consumed by measuring the M first frequencies is less than that consumed by measuring N first frequencies. The N first frequencies are measured on condition that the user equipment has no power-saving requirement. N is a positive integer and larger than or equals M. A first priority of the first frequency is higher than a second priority of a second frequency, where a current serving cell which the UE is camped on is on the second frequency.

Abstract: A device for signal processing includes a memory and a processor. The memory is configured to store a parameter associated with a bandwidth-extended audio stream. The processor is configured to select a plurality of non-linear processing functions based at least in part on a value of the parameter. The processor is also configured to generate a high-band excitation signal based on the plurality of non-linear processing functions.

Abstract: In some embodiments, a pitch filter for filtering a preliminary audio signal generated from an audio bitstream is disclosed. The pitch filter has an operating mode selected from one of either: (i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal, and (ii) an inactive mode where the pitch filter is disabled. The preliminary audio signal is generated in an audio encoder or audio decoder having a coding mode selected from at least two distinct coding modes, and the pitch filter is capable of being selectively operated in either the active mode or the inactive mode while operating in the coding mode based on control information.

Abstract: Provided is a technique according to which it is possible to obtain a decoded sound signal of high sound quality without significantly increasing the delay time compared to a configuration in which only a decoded sound signal of the minimum necessary sound quality is obtained. In a terminal apparatus connected to a first communication line and a second communication line with a lower priority level therethan, sound signals of multiple channels are obtained and output based on a monaural code included in a first code string input from the first communication line and an extended code included in a second code string with the closest frame number to that of the monaural code among extended codes included in the second code string input from the second communication line.

Abstract: A method for spectrum recovery in spectral decoding of an audio signal, comprises obtaining of an initial set of spectral coefficients representing the audio signal, and determining a transition frequency. The transition frequency is adapted to a spectral content of the audio signal. Spectral holes in the initial set of spectral coefficients below the transition frequency are noise filled and the initial set of spectral coefficients are bandwidth extended above the transition frequency. Decoders and encoders being arranged for performing part of or the entire method are also illustrated.

Abstract: A method and apparatus for variable rate compression with a conditional autoencoder is herein provided. According to one embodiment, a method for compression includes receiving a first image and a first scheme as inputs for an autoencoder network; determining a first Lagrange multiplier based on the first scheme; and using the first image and the first Lagrange multiplier as inputs, computing a second image from the autoencoder network. The autoencoder network is trained using a plurality of Lagrange multipliers and a second image as training inputs.

Abstract: Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.

Abstract: Systems and methods are presented for efficient cross-fading of compressed domain information streams on a user/client device. Exemplary systems may provide cross-fade between AAC/Enhanced AAC Plus information streams, between MP3 information streams, or between information streams of unmatched formats. These systems are distinguished in that cross-fade is directly applied to compressed bitstreams so a single decode operation is performed on the resulting bitstream. Thus, a set of frames from each input stream associated with the time interval in which a cross fade is decoded, and combined and recoded with a cross fade or other effect now in the compressed bitstream. Once sent through the client device's decoder, the user hears the transitional effect. The only input data that is decoded and processed is that associated with the portion of each stream used the crossfade, blend or other interstitial, and thus the vast majority of input streams are left compressed.

Abstract: An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.

Abstract: An encoder and a method therein for Pyramid Vector Quantizer, PVQ, shape search, the PVQ taking a target vector x as input and deriving a vector y by iteratively adding unit pulses in an inner dimension search loop. The method comprises, before entering a next inner dimension search loop for unit pulse addition, determining, based on the maximum pulse amplitude, maxampy, of a current vector y, whether more than a current bit word length is needed to represent enloopy, in a lossless manner in the upcoming inner dimension loop. The variable enloopy is related to an accumulated energy of the vector y. The performing of this method enables the encoder to keep the complexity of the search at a reasonable level.

Abstract: The present disclosure provides a bit allocation method and apparatus for an audio signal. The bit allocation method for an audio signal includes: obtaining T audio signals in a current frame, where T is a positive integer; determining a first audio signal set based on the T audio signals, where the first audio signal set includes M audio signals, M is a positive integer, T?M; determining M priorities of the M audio signals in the first audio signal set; and performing bit allocation on the M audio signals based on the M priorities of the M audio signals.

Abstract: A streaming engine employed in a digital data processor specifies a fixed read only data stream defined by plural nested loops. An address generator produces address of data elements. A steam head register stores data elements next to be supplied to operational units for use as operands. A promotion unit optionally increases date element data size by an integral power of 2 either zero filing or sign filling the additional bits. A decimation unit optionally decimates data elements by an integral factor of 2. For ease of implementation the promotion factor must be greater than or equal to the decimation factor.

Abstract: An audio decoder for providing a decoded audio signal representation on the basis of an encoded audio signal representation is configured to adjust decoding parameters in dependence on a configuration information, to decode one or more audio frames using a current configuration information, to compare a configuration information in a configuration structure associated with one or more frames to be decoded by the current configuration information, and to make a transition to perform decoding using the configuration information in the configuration structure associated with the one or more frames to be decoded as a new configuration information if the configuration information in the configuration structure associated with the one or more frames to be decoded, or a relevant portion thereof, is different from the current configuration information, and to consider a stream identifier information included in the configuration structure when comparing the configuration information.

Abstract: A system for communication between BMSs of a battery pack including a plurality of BMSs connected to a parallel communication network, and a plurality of slave BMSs allocated with different communication identifiers, each having a variable field; and a master BMS for allocating a communication identifier to each of the plurality of slave BMSs through the parallel communication network, changing a priority determination value allocated to the variable field according to a predetermined condition, and performing communication with the plurality of slave BMSs based on the communication identifier.

Abstract: In a method to decode signals, a computing device decodes spectral coefficients of a current frame are grouped into a plurality of sub-bands. The computing device classifies a sub-band as a bit allocation unsaturated sub-band based on an average quantity of allocated bits per spectral coefficient of a sub-band of the plurality of sub-bands and a threshold. The computing device obtains a noise filling gain based on an envelope of the sub-band, and obtains a reconstructed spectral coefficient of the sub-band by performing noise filling based on the noise filling gain. The computing device then obtains a frequency domain audio signal based on spectral coefficients in the sub-band obtained by decoding and the reconstructed spectral coefficient.

Abstract: An apparatus for encoding directional audio coding parameters including diffuseness parameters and direction parameters includes: a parameter calculator for calculating the diffuseness parameters with a first time or frequency resolution and for calculating the direction parameters with a second time or frequency resolution; and a quantizer and encoder processor for generating a quantized and encoded representation of the diffuseness parameters and the direction parameters.

Abstract: A method includes identifying multiple apportionments, where each apportionment identifies numbers of bit copies to be stored in at least one memory for at least some bits of a data value. The method also includes, for each apportionment, estimating a numerical error associated with use of the apportionment with a specified function, where the numerical error is estimated by creating errors in bit copies of multiple data values processed using the specified function. The method further includes combining portions of different ones of the apportionments having lower estimated numerical errors to create multiple derived apportionments. The method also includes, for each derived apportionment, estimating a numerical error associated with use of the derived apportionment with the specified function. In addition, the method includes selecting a final apportionment for use with the specified function, where the final apportionment includes or is based on at least one of the derived apportionments.

Abstract: An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal.

Abstract: In accordance with some embodiments of the disclosed subject matter, mechanisms for seamless audio melding between audio items in a playlist are provided.

Abstract: An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal.

Abstract: Methods of encoding and decoding a speech signal using a neural network model that recognizes sound sources, and encoding and decoding apparatuses for performing the methods are provided. A method of encoding a speech signal includes identifying an input signal for a plurality of sound sources; generating a latent signal by encoding the input signal; obtaining a plurality of sound source signals by separating the latent signal for each of the plurality of sound sources; determining a number of bits used for quantization of each of the plurality of sound source signals according to a type of each of the plurality of sound sources; quantizing each of the plurality of sound source signals based on the determined number of bits; and generating a bitstream by combining the plurality of quantized sound source signals.

Abstract: An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal.

Abstract: An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal.

Abstract: An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal.

Abstract: An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal.

Abstract: Disclosed is an electronic device that receives a packet of a content image including a plurality of frames from an external device, transmits a signal indicating whether a packet is received, including a request for a bit rate of an image in a transmitted signal, receives a packet with a changed bit rate of the image based on the request, and displays the image on the display based on the received packet.

Abstract: To provide demapping at a receiving side, a trained model for a demapper is used to output log-likelihood ratios of received signals representing data in a multi-user transmission. Inputs for the trained model for the demapper comprise a resource grid of equalized received signals.

Abstract: A method for mixing, processing and enhancing signals using signal decomposition is presented. A method for improving sorting of decomposed signal parts using cross-component similarity is also provided.

Abstract: An audio signal decoding device includes a non-transitory memory storage stores audio data in a form of a bitstream; and an audio decoder, by which a first spectral coefficient of a first sub-band of a current frame of an audio signal by decoding the bitstream is obtained; a first average quantity of allocated bits per spectral coefficient of the first sub-band is obtained; a first noise filling gain for the first sub-band is obtained when the first average quantity is less than a threshold; a second spectral coefficient is reconstructed according to the first noise filling gain; a frequency domain audio signal is obtained according to the first spectral coefficient and the second spectral coefficient; and a time domain audio signal is generated according to the frequency domain signal.

Abstract: A method for improving speech signal intelligibility is performed at a device. A speech signal is obtained. A correspondence between the speech signal and a respective user group among different user groups having distinct voice characteristics is identified. Pre-encoding signal augmentation is performed on the speech signal with a respective pre-augmentation filtering coefficient that corresponds to the respective user group to obtain a group-specific pre-augmented speech signal. The device encodes the pre-augmented speech signal for subsequent transmission through the voice communication channel. An encoded version of the pre-augmented speech signal has reduced loss of signal quality as compared to an encoded version of the speech signal that is obtained without the pre-encoding signal augmentation.

Abstract: An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal.

Abstract: An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.

Abstract: An audio signal encoding method performed by an encoder includes identifying a time-domain audio signal in a unit of blocks, quantizing a linear prediction coefficient extracted from a combined block in which a current original block of the audio signal and a previous original block chronologically adjacent to the current original block using frequency-domain linear predictive coding (LPC), generating a temporal envelope by dequantizing the quantized linear prediction coefficient, extracting a residual signal from the combined block based on the temporal envelope, quantizing the residual signal by one of time-domain quantization and frequency-domain quantization, and transforming the quantized residual signal and the quantized linear prediction coefficient into a bitstream.

Abstract: An objective of the present invention is to correct a temporal envelope shape of a decoded signal with a small information volume and to reduce perceptible distortions.

Abstract: A radio frequency (RF) system may include an RF transceiver and a baseband engine, application specific integrated circuit (ASIC) coupled to the RF transceiver and configured to perform a given baseband engine operation from among different baseband engine operations. The baseband engine ASIC may include a memory and a state machine coupled thereto and configured to store a respective set of programming instructions for each of different baseband engine operations and to permit selection of the given set of programming instructions. The baseband engine may also include a programmable processing circuit coupled to the memory and the state machine and configured to perform block coding computations responsive to the given set of programming instructions.