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: 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: 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 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: 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: 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: 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 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: 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.

Abstract: A Unified Speech and Audio Codec (USAC) that may process a window sequence based on mode switching is provided. The USAC may perform encoding or decoding by overlapping between frames based on a folding point when mode switching occurs. The USAC may process different window sequences for each situation to perform encoding or decoding, and thereby may improve a coding efficiency.

Abstract: Dynamic device speaker tuning for echo control includes detecting audio rendering from a speaker on a device; based at least on detecting the audio rendering, capturing, with a microphone on the device, an echo of the rendered audio; performing a Fourier Transform on the echo and the rendered audio; determining a real-time transfer function for at least one signature band; determining a difference between the real-time transfer function and a reference transfer function; and tuning the speaker for audio rendering, based at least on the difference between the real-time transfer function and the reference transfer function, by adjusting an audio amplifier equalization. For some examples, the signature band represents a wall echo or an alternative mounting option. For some examples, the echo is collected during intervals while the audio rendering is ongoing.

Abstract: A method of encoding samples in a digital signal is provided that includes receiving a frame of N samples of the digital signal, determining L possible distinct data values in the N samples, determining a reference data value in the L possible distinct data values and a coding order of L?1 remaining possible distinct data values, wherein each of the L?1 remaining possible distinct data values is mapped to a position in the coding order, decomposing the N samples into L?1 coding vectors based on the coding order, wherein each coding vector identifies the locations of one of the L?1 remaining possible distinct data values in the N samples, and encoding the L?1 coding vectors.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information is configured to obtain decoded spectral values on the basis of an encoded information representing the spectral values. The audio decoder is configured to jointly decode two or more most significant bits per spectral value on the basis of respective symbol codes for a set of spectral values using an arithmetic decoding, wherein a respective symbol code represents two or more most significant bits per spectral value for one or more spectral values. The audio decoder is configured to decode one or more least significant bits associated with one or more of the spectral values in dependence on how much least significant bit information is available, such that one or more least significant bits associated with one or more of the spectral values are decoded.

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

Abstract: A computer-implemented method and corresponding system for processing sensor data associated with a vehicle is provided. The sensor data may be compressed or encoded with a dictionary according to sparse approximation theory, resulting in a sparse representation of the sensor data. Processing may further comprise detecting an event associated with the vehicle, wherein an event may be an accident recorded by sensors of the vehicle providing the sensor data. The detection of the event may be based on processing of the sparse representation of the sensor data alone without decoding the sparse representation. The detection of the event may further employ machine learning methods trained to the detection of an event from the sparse representation of the sensor data, or a combination of sparse representations of sensor data originating from a plurality of vehicles or sensors.

Abstract: A device comprising: at least two interfaces for interconnecting between at least one audio source and at least one audio sink device; and unidirectional circuitry configured to allow transmission of an audio signal only from said at least one audio source to said at least one audio sink device.

Abstract: A technique for managing real-time communications includes generating, during a communication session between at least a first computing device and a second computing device over a computer network, multiple audio factors of the communication session, each of the audio factors being susceptible to degradation in a way that affects audio quality of the communication session. The technique further includes combining the audio factors to produce an overall measure of audio quality and taking remedial action to improve the overall measure of audio quality by adjusting a setting on the first computing device.

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

Abstract: Systems and methods are presented for efficient cross-fading (or other multiple clip processing) of compressed domain information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Exemplary implementation systems may provide cross-fade between AAC/Enhanced AAC Plus (EAACPlus) information streams or between MP3 information streams or even between information streams of unmatched formats (e.g. AAC to MP3 or MP3 to AAC). Furthermore, these systems are distinguished by the fact that cross-fade is directly applied to the compressed bitstreams so that a single decode operation may be performed on the resulting bitstream. Moreover, using the described methods, similar cross fade in the compressed domain between information streams utilizing other formats of compression, such as, for example, MP2, AC-3, PAC, etc. can also be advantageously implemented.

Abstract: The present disclosure provides a method and a device in a UE and a base station used for wireless communication. A first node operates first information; and transmits a first radio signal in M1 out of M frequency sub-band(s). Herein, the first information is used for determining S candidate integers; K1 of K bit block group(s) is(are) used for generating the first radio signal, any of the K bit block group(s) comprises a positive integer number of code block(s); K is a candidate integer of the S1 candidate integer(s), M is used for determining the S1 candidate integer(s) out of the S candidate integers; the first node is a base station, the operating action is transmitting; or the first node is a UE, the operating action is receiving. The above method helps reduce the amount of data retransmitted due to LBT failure of partial frequency sub-bands.

Abstract: Some disclosed systems and methods include assigning channel streams to satellite playback devices, delaying initial transmission of surround sound content to satellite playback devices to accumulate audio samples for transmission in individual frames of surround sound content, and taking advantage of configured playback delays for different channels of surround sound content to prioritize transmission of channels of surround sound content having shorter configured playback delays over transmission of channels of surround sound content having longer configured playback delays.

Abstract: An audio post-processor for post-processing an audio signal having a time-variable high frequency gain information as side information includes: a band extractor for extracting a high frequency band of the audio signal and a low frequency band of the audio signal; a high band processor for performing a time-variable modification of the high frequency band in accordance with the time-variable high frequency gain information to obtain a processed high frequency band; and a combiner for combining the processed high frequency band and the low frequency band. Furthermore, a pre-processor is illustrated.

Abstract: A video encoder may be configured to apply a multi-stage quantization process, where residuals are first quantized using an effective quantization parameter derived from the statistics of the samples of the block. The residual is then further quantized using a base quantization parameter that is uniform across a picture. A video decoder may be configured to decode the video data using the base quantization parameter. The video decoder may further be configured to estimate the effective quantization parameter from the statistics of the decoded samples of the block. The video decoder may then use the estimated effective quantization parameter for use in determining parameters for other coding tools, including filters.

Abstract: The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing a multimedia signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing filtering for multimedia signal having a plurality of subbands with a low calculation amount.

Abstract: In an example, an apparatus comprises logic, at least partially including hardware logic, to implement a lossy compression algorithm which utilizes a data transform and quantization process to compress data in a convolutional neural network (CNN) layer. Other embodiments are also disclosed and claimed.

Abstract: Lifting is a transform designed for color compression of point clouds which is adopted in one of the MPEG test models. The performance of lifting is improved herein. All the lifting coefficients are first divided into several subbands based on their assigned weights, which indicate the level of importance of each coefficient. Then, for each subband, a set of three dead-zones are derived for the three color components. The dead-zones of Cb and Cr channels are typically larger than that of Luma channel. In the original lifting scheme, Chroma is not suppressed at all. In contrast, as described herein, the size of the dead-zone is increased for different color components, which means that more quality (and bandwidth) is able to be adaptively provided for luminance coefficients than chrominance coefficients.

Abstract: A method, system, and computer program product for processing an encoded audio signal is described. In one exemplary embodiment, the system receives an encoded low-frequency range signal and encoded energy information used to frequency shift the encoded low-frequency range signal. The low-frequency range signal is decoded and an energy depression of the decoded signal is smoothed. The smoothed low-frequency range signal is frequency shifted to generate a high-frequency range signal. The low-frequency range signal and high-frequency range signal are then combined and outputted.

Abstract: The invention relates to a codec and a discriminator and methods therein for audio signal discrimination and coding. Embodiments of a method performed by an encoder comprises, for a segment of the audio signal: identifying a set of spectral peaks; determining a mean distance S between peaks in the set; and determining a ratio, PNR, between a peak envelope and a noise floor envelope. The method further comprises selecting a coding mode, out of a plurality of coding modes, based at least on the mean distance S and the ratio PNR; and applying the selected coding mode for coding of the segment of the audio signal.

Abstract: A method and device for decoding a signal, where the method includes: obtaining an average quantity of allocated bits per spectral coefficient of a sub-band of a current frame of the audio signal, wherein the sub-band includes a plurality of spectral coefficients; obtaining a noise filling gain for the sub-band when the average quantity of allocated bits per spectral coefficient is less than a classification threshold; reconstructing, according to the noise filling gain, at least some of the spectral coefficients to generate reconstructed spectral coefficients when the average quantity of allocated bits per spectral coefficient is less than a classification threshold; obtaining a frequency domain signal according to the reconstructed spectral coefficients; and generating a time domain signal based on the frequency domain signal. Therefore, a sub-band with unsaturated bit allocation in a frequency domain signal may be obtained by classification, thereby improving signal decoding quality.