Abstract: A system for enabling spatializing audio data is provided. The system analyzes audio data to identify when to generate spatialized audio data. The system can receive incoming audio data including a plurality of channel-based audio signals as well as object-based audio. The system performs an analysis of the audio data and/or metadata associated with the audio data to determine when to generate the spatialized audio data. The system can identify one or more categories associated with the audio data (e.g., stereo, mono, game effect, . . . ) and use the category to determine whether to spatialize the audio data or not spatialize the audio data.

Abstract: Audio objects are associated with positional metadata. A received downmix signal comprises downmix channels that are linear combinations of one or more audio objects and are associated with respective positional locators. In a first aspect, the downmix signal, the positional metadata and frequency-dependent object gains are received. An audio object is reconstructed by applying the object gain to an upmix of the downmix signal in accordance with coefficients based on the positional metadata and the positional locators. In a second aspect, audio objects have been encoded together with at least one bed channel positioned at a positional locator of a corresponding downmix channel. The decoding system receives the downmix signal and the positional metadata of the audio objects. A bed channel is reconstructed by suppressing the content representing audio objects from the corresponding downmix channel on the basis of the positional locator of the corresponding downmix channel.

Abstract: An audio encoding apparatus and method that encodes hybrid contents including an object sound, a background sound, and metadata, and an audio decoding apparatus and method that decodes the encoded hybrid contents are provided. The audio encoding apparatus may include a mixing unit to generate an intermediate channel signal by mixing a background sound and an object sound, a matrix information encoding unit to encode matrix information used for the mixing, an audio encoding unit to encode the intermediate channel signal, and a metadata encoding unit to encode metadata including control information of the object sound.

Abstract: A decoder for generating an un-mixed audio signal including a plurality of un-mixed audio channels is provided. Moreover, an encoder and an encoded audio signal is provided. The decoder includes an un-mixing-information determiner for determining un-mixing information by receiving first parametric side information and second parametric side information on the at least one audio object signal, wherein the frequency resolution of the second parametric side information is higher than that of the first parametric side information. Moreover, the decoder includes an un-mix module for applying the un-mixing information on a downmix signal, to obtain an un-mixed audio signal including the plurality of un-mixed audio channels.

Abstract: A sub-band processing system for reducing computational complexity and memory requirements is disclosed. The sub-band processing system includes: a first logic that partitions and stores a frequency spectrum of bins of real and imaginary data into a smaller number of sub-bands; a second logic that executes a first lossy compression for a first set of the sub-bands, wherein the first set includes those sub-bands having indices that are greater than or equal to a first index; and a third logic that executes, subsequent to a frequency spectrum processing of the lossy compressed data rendered by the second logic, a second lossy compression for a second set of the sub-bands, wherein the second set includes those sub-bands having indices that are less than the first index and greater than or equal to a second index.

Abstract: The technical disclosure relates to artificial neural network. In particular, the technical disclosure relates to how to implement efficient data access control in the neural network hardware acceleration system. Specifically, it proposes an overall design of a device that can process data receiving, bit-width transformation and data storing. By employing the technical disclosure, neural network hardware acceleration system can avoid the data access process becomes the bottleneck in neural network computation.

Abstract: A decoder is provided. The decoder includes a parametric decoding unit for generating a plurality of first estimated audio object signals by upmixing three or more downmix signals, wherein the three or more downmix signals encode a plurality of original audio object signals, wherein the parametric decoding unit is configured to upmix the three or more downmix signals depending on parametric side information indicating information on the plurality of original audio object signals. Moreover, the decoder includes a residual processing unit for generating a plurality of second estimated audio object signals by modifying one or more of the first estimated audio object signals, wherein the residual processing unit is configured to modify the one or more of the first estimated audio object signals depending on one or more residual signals.

Abstract: Disclosed are a multi-channel audio signal processing method and a multi-channel audio signal processing apparatus. The multi-channel audio signal processing method may generate N channel output signals from N/2 channel downmix signals based on an N-N/2-N structure.

Abstract: A method comprises at least the following steps: generating pseudo random noise in the digital domain coded on a number N of bits, sampled at a given frequency FH/N; multiplexing in the digital domain the binary signals produced by each of the N bits at a sampling frequency FH so as to obtain noise coded on one bit at said frequency FH; transferring the noise thus coded into the analog domain via a low-voltage differential transmission interface; filtering the analog signal in a passband which can be centered on half the sampling frequency of an analog-digital converter.

Abstract: A base station system according to an aspect of the invention includes a baseband unit (BBU) and a radio frequency (RF) unit (RFU) connected to the BBU via a communication line. The RFU (BBU) measures the frequencies of occurrence of sampled values indicated by sampled data that is a digital signal corresponding to a baseband signal to be transmitted to the BBU (RFU), and generates a frequency distribution representing a relationship between the sampled value having occurred and the frequency of occurrence. Furthermore, the RFU (BBU) determines a plurality of thresholds for compressing the sampled data, which are used for quantization processing of the sampled value, in accordance with the generated frequency distribution, and compresses the sampled data by the quantization processing using the plurality of thresholds.

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

Abstract: Disclosed is a transmission device capable of reducing quantization errors in nonlinear quantization of IQ signals. In the device, a standard deviation measuring unit measures a standard deviation ? of the amplitude distribution of input signals. A kurtosis measuring unit measures the kurtosis of the amplitude distribution of the input signals. A correction coefficient determining unit determines a correction coefficient ? corresponding to the kurtosis of the amplitude distribution of the input signals. A quantization unit quantizes the input signals using the correction coefficient ? and calculates quantization data. A multiplexing unit multiplexes the quantization data and quantization control information with each other.

Abstract: Techniques related to content adaptive entropy coding of partitions data are described.

Abstract: A system and method of providing a modulated signal useable in a signal transmission system. The method includes transforming, perhaps with respect to both time and frequency, a data frame including a plurality of data elements into a transformed data matrix. The transformed data matrix includes a plurality of transformed data elements where each of the plurality of transformed data elements is based upon each of the plurality of data elements. The method further includes generating the modulated signal in accordance with the transformed data elements of the transformed data matrix.

Abstract: A base station system according to an aspect of the invention includes a baseband unit (BBU) and a radio frequency (RF) unit (RFU) connected to the BBU via a communication line. The RFU (BBU) measures the frequencies of occurrence of sampled values indicated by sampled data that is a digital signal corresponding to a baseband signal to be transmitted to the BBU (RFU), and generates a frequency distribution representing a relationship between the sampled value having occurred and the frequency of occurrence. Furthermore, the RFU (BBU) determines a plurality of thresholds for compressing the sampled data, which are used for quantization processing of the sampled value, in accordance with the generated frequency distribution, and compresses the sampled data by the quantization processing using the plurality of thresholds.

Abstract: Methods, systems, and devices for wireless communication are provided for mobility management for wireless communications systems that utilize a flexible bandwidth carrier. Some embodiments include approaches for determining bandwidth information, such as one or more bandwidth scaling factors N and/or flexible bandwidths, at a user equipment (UE), where the bandwidth information may not be signaled to the UE. Embodiments for determining bandwidth information include: random ordered bandwidth scaling factor approaches, delay ordered bandwidth scaling factor approaches, storing bandwidth scaling factor value in UE Neighbor Record approaches, spectrum measurement approaches, spectrum calculation approaches, and/or a priori approaches. Flexible bandwidth carrier systems may utilize spectrum portions that may not be big enough to fit a normal waveform.

Abstract: A digital audio pass-through device capable of connecting multiple host computing devices is described. The digital audio pass-through device allows computing devices such as personal computers (Mac or PC), tablets, and smart phones to share high quality digital audio data streams with one another via USB connections. The digital audio pass-through device also allows for the aggregation of high quality audio data from multiple computing devices with limited to no latency.

Abstract: A method and system for delivering encoded content are provided. A holdback representing a portion of the encoded content is extracted, thereby damaging the encoded content. The damaged encoded content is distributed. The holdback is transmitted to enable reintegration of the holdback with the damaged encoded content to restore the encoded content.

Abstract: The invention relates to an apparatus (1) for improving a stereo audio signal of an FM stereo radio receiver (10). It is suggested to employ components of a standard HE-ACC v2 decoder (40) in such apparatus (1) to efficiently recover the stereo audio signal (L\R?) while also exhibiting good signal quality. To that end, parametric stereo parameters (15) shall be quantized and submitted to such decoder (40) together with a mono-downmix signal (25). The decoder (25) can then easily recover the parametric stereo parameters (15) by de-quantizing them via a simple lookup table, and finally creating the stereo output signal (L?R?) from the de-quantized parametric stereo parameters and the mono-downmix signal (25). The apparatus (1) further includes a parameter estimation stage (5) which may again be embodied on the basis of components of a standard HE-ACC v2 encoder (65).

Abstract: Multibitrate (MBR) live video broadcasting is disclosed in which live video input is copied into a plurality of streams each designated for encoding into a different bitrate. The MBR broadcasting operation is made efficient by performing pre-quantization calculations only the first of the plurality of streams. The results of those calculations are then merely copied to the other streams. Quantization and encoding processes may then be applied to each of the streams to process the streams into their respective, predetermined bitrates.

Abstract: A system comprises: first and second base stations; and a plurality of relay nodes, each of said relay nodes connected to the first base station, each of said relay nodes being connected to at least one other relay node, whereby at least one relay node is configured to at least one of receive and send information for another of said relays nodes; wherein when at least one of the plurality of relay nodes is handed over to a second base station the at least one relay node is configured to receive and/or send information via another of the relay nodes connected to the first base station.

Abstract: Provided is an apparatus and method for coding and decoding multi-object audio signals with various channels and providing backward compatibility with a conventional spatial audio coding (SAC) bitstream. The apparatus includes: an audio object coding unit for coding audio-object signals inputted to the coding apparatus based on a spatial cue and creating rendering information for the coded audio-object signals, where the rendering information provides a coding apparatus including spatial cue information for audio-object signals; channel information of the audio-object signals; and identification information of the audio-object signals, and used in coding and decoding of the audio signals.

Abstract: Provided is a signal processing apparatus including a first periodicity detecting section detecting periodicity information of an acoustic signal included in a first content, as first periodicity information, a second periodicity detecting section detecting the periodicity information of an acoustic signal included in a second content, as second periodicity information, a similarity calculating section calculating a similarity between the first periodicity information detected by the first periodicity detecting section and the second periodicity information detected by the second periodicity detecting section, and a synchronization information generating section generating synchronization information used at a time of synchronizing the first content and the second content, based on the similarity calculated by the similarity calculating section.

Abstract: An apparatus for and a method of processing a multi-channel audio signal using space information. The apparatus includes: a main coding unit down mixing a multi-channel audio signal by applying space information to surround components included in the multi-channel audio signal, generating side information using the multi-channel audio signal or a stereo signal of a down-mixed result, coding the stereo signal and the side information, and transmitting the coded result as a coding signal; and a main decoding unit receiving the coding signal, decoding the stereo signal and the side information using the received coding signal, up mixing the decoded stereo signal using the decoded side information, and restoring the multi-channel audio signal.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: A data transmitting apparatus is provided. The data transmitting apparatus includes a packet generator configured to generate a packet including a plurality of sub packets and a transmitter configured to transmit the generated packet to a data receiving apparatus. Each of the plurality of sub packets includes audio data corresponding to content among a plurality of contents.

Abstract: The present document relates to audio forensics, notably the blind detection of traces of parametric audio encoding/decoding. In particular, the present document relates to the detection of parametric frequency extension audio coding, such as spectral band replication (SBR) or spectral extension (SPX), from uncompressed waveforms such as PCM (pulse code modulation) encoded waveforms. A method for detecting frequency extension coding history in a time domain audio signal is described.

Abstract: The application relates to audio encoder and decoder systems. An embodiment of the encoder system comprises a downmix stage for generating a downmix signal and a residual signal based on a stereo signal. In addition, the encoder system comprises a parameter determining stage for determining parametric stereo parameters such as an inter-channel intensity difference and an inter-channel cross-correlation. Preferably, the parametric stereo parameters are time- and frequency-variant. Moreover, the encoder system comprises a transform stage. The transform stage generates a pseudo left/right stereo signal by performing a transform based on the downmix signal and the residual signal. The pseudo stereo signal is processed by a perceptual stereo encoder. For stereo encoding, left/right encoding or mid/side encoding is selectable. Preferably, the selection between left/right stereo encoding and mid/side stereo encoding is time- and frequency-variant.

Abstract: A system and method of providing a modulated signal useable in a signal transmission system. The method includes transforming, perhaps with respect to both time and frequency, a data frame including a plurality of data elements into a transformed data matrix. The transformed data matrix includes a plurality of transformed data elements where each of the plurality of transformed data elements is based upon each of the plurality of data elements. The method further includes generating the modulated signal in accordance with the transformed data elements of the transformed data matrix.

Abstract: A method for modulating data for transmission within a communication system. The method includes establishing a time-frequency shifting matrix of dimension N×N, wherein N is greater than one. The method further includes combining the time-frequency shifting matrix with a data frame to provide an intermediate data frame. A transformed data matrix is provided by permuting elements of the intermediate data frame. A modulated signal is generated in accordance with elements of the transformed data matrix.

Abstract: Compression of audio signal data is described herein. In various embodiments, the compression of each unit of the audio signal data includes the employment of a distribution substantially representative of a subblock of residual data of the unit of audio signal data, to reduce the amount of data having to be transmitted to transmit the unit of audio signal data to a recipient.

Abstract: A decoding apparatus (10) is disclosed which includes: a storing means (11) for storing encoded audio signals including multi-channel audio signals; a transforming means (40) for transforming the encoded audio signals to generate transform block-based audio signals in a time domain; a window processing means (41) for multiplying the transform block-based audio signals by a product of a mixture ratio of the audio signals and a first window function, the product being a second window function; a synthesizing means (43) for overlapping the multiplied transform block-based audio signals to synthesize audio signals of respective channels; and a mixing means (14) for mixing audio signals of the respective channels between the channels to generate a downmixed audio signal. Furthermore, an encoding apparatus is also disclosed which downmixes the multi-channel audio signals, encodes the downmixed audio signals, and generates the encoded, downmixed audio signals.

Abstract: A method (700, 800) and apparatus (100, 200) processes audio frames to transition between different codecs. The method can include producing (720), using a first coding method, a first frame of coded output audio samples by coding a first audio frame in a sequence of frames. The method can include forming (730) an overlap-add portion of the first frame using the first coding method. The method can include generating (740) a combination first frame of coded audio samples based on combining the first frame of coded output audio samples with the overlap-add portion of the first frame. The method can include initializing (760) a state of a second coding method based on the combination first frame of coded audio samples. The method can include constructing (770) an output signal based on the initialized state of the second coding method.

Abstract: A method comprising: sampling received audio at a first rate to produce a first audio signal; transforming the first audio signal into a sparse domain to produce a sparse audio signal; re-sampling of the sparse audio signal to produce a re-sampled sparse audio signal; and providing the re-sampled sparse audio signal, wherein bandwidth required for accurate audio reproduction is removed but bandwidth required for spatial audio encoding is retained AND/OR a method comprising: receiving a first sparse audio signal for a first channel; receiving a second sparse audio signal for a second channel; and processing the first sparse audio signal and the second sparse audio signal to produce one or more inter-channel spatial audio parameters.

Abstract: A stream combining apparatus is provided, comprising an input unit that receives the input of first group access units and second group access units from two streams that are generated by overlap transform; a decoder that generates group frames by decoding the group access units and that generates group frames by decoding the group access units; and a combining unit that uses first group frames and second group frames as a frame of reference for the access units, that decodes the frames, that performs selective mixing to generate mixed frames, that encodes said mixed frames, that generates a prescribed number of group access units, and that joins two streams, using a prescribed number of group access units as a joint such that the access units adjacent to each other on the boundary between the two streams and a prescribed number of group access units are stitched so that the information for decoding the same common frames is distributed.

Abstract: There is provided an audio coding device including a first windowing part that multiplies an audio signal by a first window function, a second windowing part that multiplies the audio signal by a second window function having a characteristic different from a characteristic of the first window function, a window selecting part that selects the first window function or the second window function as an optimum window function based on the audio signal multiplied by the first windowing part and the audio signal multiplied by the second windowing part, a coding part that codes a frequency spectrum of the audio signal multiplied by the optimum window function, and a transmitting part that transmits the frequency spectrum coded by the coding part and window function information representing the optimum window function.

Abstract: Embodiments are generally directed to systems and methods for bit allocation and band partitioning for gain-shape vector quantization in an audio codec. An audio codec implements a method that uses an implicit, dynamic scheme to allow an encoder and decoder to recreate a series of bit allocation decisions for gain and shape without transmitting additional side information for each decision, based on the number of bits that are left remaining and available in a given packet. For implementation in practical codecs, the band comprising the allocation of bits for the shape is recursively split into equal partitions until the number of bits allocated to each partition is less than the maximum codebook size.

Abstract: A method and system for performing sample rate conversion is provided. The method may include configuring a system to convert a sample rate of a first audio channel of a plurality of audio channels to produce a first audio stream of samples. The system may be dynamically reconfigured to convert a sample rate of a second of the plurality of audio channels to produce a second audio stream of samples, wherein the first and second audio streams are output from the system at the same time. The method may further include arbitrating between request for additional data from the first and second audio stream of samples, where processing of the first channel is suspended when the request corresponds to a second channel that is of higher priority.

Abstract: For a bandwidth extension of an audio signal, in a signal spreader the audio signal is temporally spread by a spread factor greater than 1. The temporally spread audio signal is then supplied to a demicator to decimate the temporally spread version by a decimation factor matched to the spread factor. The band generated by this decimation operation is extracted and distorted, and finally combined with the audio signal to obtain a bandwidth extended audio signal. A phase vocoder in the filterbank implementation or transformation implementation may be used for signal spreading.

Abstract: A noise filler for providing a noise-filled spectral representation of an audio signal on the basis of an input spectral representation of the audio signal has a spectral region identifier configured to identify spectral regions of the input spectral representation spaced from non-zero spectral regions of the input spectral representation by at least one intermediate spectral region, to obtain identified spectral regions, and a noise inserter configured to selectively introduce noise into the identified spectral regions to obtain the noise-filled spectral representation of the audio signal. A noise filling parameter calculator for providing a noise filling parameter on the basis of a quantized spectral representation of an audio signal has a spectral region identifier, as mentioned above, and a noise value calculator configured to selectively consider quantization errors of the identified spectral regions for a calculation of the noise filling parameter.

Abstract: The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal.

Abstract: A quantizing apparatus is provided that includes a quantization path determiner that determines a path from a first path not using inter-frame prediction and a second path using the inter-frame prediction, as a quantization path of an input signal, based on a criterion before quantization of the input signal; a first quantizer that quantizes the input signal, if the first path is determined as the quantization path of the input signal; and a second quantizer that quantizes the input signal, if the second path is determined as the quantization path of the input signal.

Abstract: A method, medium, and apparatus encoding and/or decoding a multichannel audio signal. The method includes detecting the type of spatial extension data included in an encoding result of an audio signal, if the spatial extension data is data indicating a core audio object type related to a technique of encoding core audio data, detecting the core audio object type; decoding core audio data by using a decoding technique according to the detected core audio object type, if the spatial extension data is residual coding data, decoding the residual coding data by using the decoding technique according to the core audio object type, and up-mixing the decoded core audio data by using the decoded residual coding data. According to the method, the core audio data and residual coding data may be decoded by using an identical decoding technique, thereby reducing complexity at the decoding end.

Abstract: A method for processing an audio signal is disclosed. The method for processing an audio signal includes frequency-transforming an audio signal to generate a frequency-spectrum, deciding a weighting per band corresponding to energy per band using the frequency spectrum, receiving a masking threshold based on a psychoacoustic model, applying the weighting to the masking threshold to generate a modified masking threshold, and quantizing the audio signal using the modified masking threshold.

Abstract: A band broadening apparatus includes a processor configured to analyze a fundamental frequency based on an input signal bandlimited to a first band, generate a signal that includes a second band different from the first band based on the input signal, control a frequency response of the second band based on the fundamental frequency, reflect the frequency response of the second band on the signal that includes the second band and generate a frequency-response-adjusted signal that includes the second band, and synthesize the input signal and the frequency-response-adjusted signal.

Abstract: A method for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources is provided. This method comprises decomposing the multi-channel signal into frequency bands and, per frequency band, obtaining directivity information per sound source of the sound scene, the information being representative of the spatial distribution of the sound source in the sound scene, of selecting a set of sound sources of the sound scene constituting principal sources, of matrixing the selected principal sources to obtain a sum signal with a reduced number of channels and, of coding the directivity information and of forming a binary stream comprising the coded directivity information, the binary stream being transmittable in parallel with the sum signal. A decoding method is also provided that is able to decode the sum signal and the directivity information to obtain a multi-channel signal, to an adapted coder and an adapted decoder.

Abstract: Provided is an apparatus for integrally encoding and decoding a speech signal and an audio signal. An encoding apparatus for integrally encoding a speech signal and an audio signal, may include: a module selection unit to analyze a characteristic of an input signal and to select a first encoding module for encoding a first frame of the input signal; a speech encoding unit to encode the input signal according to a selection of the module selection unit and to generate a speech bitstream; an audio encoding unit to encode the input signal according to the selection of the module selection unit and to generate an audio bitstream; and a bitstream generation unit to generate an output bitstream from the speech encoding unit or the audio encoding unit according to the selection of the module selection unit.

Abstract: Disclosed is an audio encoding device which removes unnecessary inter-channel parameters from the subject to be encoded, improving the encoding efficiency thereby. In this audio encoding device, a principal component analysis unit (301) converts an inputted left signal {Lsb(f)} and an inputted right signal {Rsb(f)} into a principal component signal {PCsb(f)} and an ambient signal {Asb(f)} and calculates for each sub-band, a rotation angle which indicates the degree of conversion; a monophonic encoding unit (303) encodes the principal component signal {Pcsb(f)}; a rotation angle encoding unit (302) encodes the angle of rotation {?b}; a local monophonic decoding unit (603) creates a decoded principal component signal; and a redundant parameter elimination unit (604) identifies the redundant parameters by analyzing the encoding quality of the decoded principal component signal and eliminates the redundant parameters from the signal to be encoded.

Abstract: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

Abstract: The present proposes new methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.