Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore, compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

Abstract: Apparatuses, arrangements and methods therein for generation of comfort noise are disclosed. In short, the solution relates to exploiting the spatial coherence of multiple input audio channels in order to generate high quality multi channel comfort noise.

Abstract: An audio encoder for encoding segments of coefficients, the segments of coefficients representing different time or frequency resolutions of a sampled audio signal, the audio encoder including a processor for deriving a coding context for a currently encoded coefficient of a current segment based on a previously encoded coefficient of a previous segment, the previously encoded coefficient representing a different time or frequency resolution than the currently encoded coefficient. The audio encoder further includes an entropy encoder for entropy encoding the current coefficient based on the coding context to obtain an encoded audio stream.

Abstract: Voice data transmission with adaptive redundancy creates a voice data packet by packetizing the voice data payload and a number of redundant payloads selected from a set of previous voice data payloads. The voice data from the voice data payload is analysed to determine whether it is a critical or non-critical payload by classifying the received voice data as voiced or unvoiced. If at least a portion of the voice data is classified as unvoiced, the voice data payload is determined to be a critical payload. If it is a critical payload, then the voice data payload is added to the set of previous voice data payloads for inclusion as a redundant payload in subsequent voice data packets. The voice data packet is then forwarded for transmission over the network.

Abstract: An improved analysis and determination of a relationship between a tone frequency and spurious frequencies is provided. A device under test is operated to generate signals based on multiple different tone frequencies. For each tone frequency a spectrum of the generated signal is measured and spurious frequencies in the generated signal are identified. Based on the measured spectrums a representation of the spurious frequencies versus the related tone frequencies is generated. Trajectories may be inserted into the representation for indicating relationships between the tone frequencies and the spurious frequencies. This representation provides a useful basis for a fast and reliable identification of relationships between spurious frequencies and tone frequencies.

Abstract: An audio encoder has a first information sink oriented encoding branch such as a spectral domain encoding branch, a second information source or SNR oriented encoding branch such as an LPC-domain encoding branch, and a switch for switching between the first and second encoding branches, the second encoding branch having a converter into a specific domain different from the spectral domain such as an LPC analysis stage generating an excitation signal, and the second encoding branch having a specific domain coding branch such as LPC domain processing branch, and a specific spectral domain coding branch such as LPC spectral domain processing branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch. An audio decoder has a first domain decoder, a second domain decoder, and a third domain decoder as well as two cascaded switches for switching between the decoders.

Abstract: Apparatuses, arrangements and methods therein for generation of comfort noise are disclosed. In short, the solution relates to exploiting the spatial coherence of multiple input audio channels in order to generate high quality multi channel comfort noise.

Abstract: A method for representing a second presentation of audio channels or objects as a data stream, the method comprising the steps of: (a) providing a set of base signals, the base signals representing a first presentation of the audio channels or objects; (b) providing a set of transformation parameters, the transformation parameters intended to transform the first presentation into the second presentation; the transformation parameters further being specified for at least two frequency bands and including a set of multi-tap convolution matrix parameters for at least one of the frequency bands.

Abstract: Methods and apparatus relating to platform power consumption reduction via power state switching are described. In one embodiment, control logic causes a processor to enter a first low power consumption state (e.g., S0ix) instead of a second low power consumption state (e.g., S3) based on whether a threshold time period exists between a first wake event (e.g., corresponding to a first one of one or more awake requests) and a second wake event (e.g., corresponding to a second one of the one or more awake requests). Other embodiments are also claimed and disclosed.

Abstract: A data packer forma bit stream for forwarding values to memory. The bit stream includes the values and respective prefixes for identifying the values and the data packer is configured to insert the prefixes at predetermined boundaries in the bit stream, such that a prefix for identifying one value is inserted between bits that define a value identified by a preceding prefix. A data unpacker unpacks a bit stream that comprises values and respective prefixes for identifying those values that are located at predetermined boundaries in the bit stream, such that a prefix for identifying one value is inserted between bits that define a value identified by a preceding prefix. The data unpacker identifies a prefix at a predetermined boundary in the bit stream and determine, in dependence on that prefix and the predetermined boundaries, a location of the next prefix in the bit stream.

Abstract: In general, techniques are described by which to code scaled spatial components. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may store a bitstream including an encoded foreground audio signal and a corresponding quantized spatial component. The one or more processors may perform psychoacoustic audio decoding with respect to the encoded foreground audio signal to obtain a foreground audio signal, and determine, when performing psychoacoustic audio decoding, a bit allocation for the encoded foreground audio signal. The one or more processors may dequantize the quantized spatial component to obtain a scaled spatial component, and descale, based on the bit allocation, the scaled spatial component to obtain a spatial component. The one or more processors may reconstruct, based on the foreground audio signal and the spatial component, scene-based audio data.

Abstract: Methods and systems for detecting and combining audio transmissions containing data are provided. In one embodiment, a method is provided that includes receiving multiple audio signals from multiple receivers. An audio transmission may be detected in a first subset of the audio signals and the audio transmission may contain data. A second subset of the audio signals may be identified with the strongest magnitude of the audio transmission. Portions of the second subset of the audio signals may then be combined to generate a combined audio transmission signal and the combined audio transmission signal may be decoded for further processing.

Abstract: An encoder operable to filter audio signals into a plurality of frequency band components, generate quantized digital components for each band, identify a potential for pre-echo events within the generated quantized digital components, generate an approximate signal by decoding the quantized digital components using inverse pulse code modulation, generate an error signal by comparing the approximate signal with the sampled audio signal, and process the error signal and quantized digital components. The encoder operable to process the error signal by processing delayed audio signals and Q band values, determining the potential for pre-echo events from the Q band values, and determining scale factors and MDCT block sizes for the potential for pre-echo events.

Abstract: Present disclosure provide an audio signal encoding method and communication terminal, which relate to the communications field. The method and communication terminal are used to obtain an analog audio signal and encoding the analog audio signal to obtain a bitstream representing the analog audio signal, in which a proper bit allocation for spectral coefficients can be performed.

Abstract: A method, including the actions of obtaining data based on a current and/or anticipated future wearing implementation of a hearing prosthesis, and adjusting a parameter of the hearing prosthesis based on the current or anticipated future wearing implementation of the hearing prosthesis, and evoking a hearing percept using the hearing prosthesis with the adjusted parameter.

Abstract: A system and method for volumetric sounds generation. The method includes: generating multiple sounds beams from a plurality of microphones within a three-dimensional space; capturing multiple sound signals generated by multiple sounds sources located within the three-dimensional space, where the multiple sound signals are captured based on the multiple sound beams; and, generating a pattern for each of the multiple sound sources.

Abstract: An audio output system for providing an output signal to a micro-speaker provides energy reduction and thermal protection by removing components of an input signal from which the output signal is generated, so that the audio power output system does not expend power reproducing portions of the audio information that would not be perceived by a listener. The micro-speaker has a resonant frequency such that substantial content of the audio information lies below the resonant frequency. A processing subsystem receives the audio information and generates the output signal by filtering the audio information with a response simulating the micro-speaker and an acoustic path to the listener, and comparing the filtered audio information with a frequency-dependent threshold of hearing. The processing subsystem then removes portions of the audio information that have an amplitude below the frequency-dependent threshold of hearing.

Abstract: Methods, systems, and devices for a decision directed multi-modulus searching algorithm are described. A receiver may receive an optical signal including a set of data symbols. The receiver may iteratively determine a set of centroids for demodulating the set of data symbols (e.g., as part of a training procedure). The centroids may be used to demodulate the set of data symbols according to a modulation constellation associated with the set of data symbols. The training procedure may include, for each data symbol of a subset of data symbols, assigning a centroid of the set of centroids to each data symbol and updating the set of centroids based on assigning the centroid to each data symbol. The receiver may demodulate the set of data symbols based on the updated set of centroids.

Abstract: The present disclosure provides a speech/audio signal processing method based on wideband switching and a coding apparatus. The method includes: if a first wideband speech/audio signal is a harmonic signal, adjusting a determining condition for determining that a second wideband speech/audio signal is a harmonic signal, to obtain a first determining condition, where the first wideband speech/audio signal is a signal before wideband switching, and the second wideband speech/audio signal is a signal after the wideband switching; and determining, according to the first determining condition, whether the second wideband speech/audio signal is a harmonic signal. In the case of wideband switching, signal types of speech/audio signals remain as consistent as possible before and after the switching, so that continuity of the speech/audio signal decoded by a decoder device is ensured as much as possible, further improving speech communication service quality.

Abstract: Methods for generating an object based audio program which is renderable in a personalizable manner, e.g., to provide an immersive, perception of audio content of the program. Other embodiments include steps of delivering (e.g., broadcasting), decoding, and/or rendering such a program. Rendering of audio objects indicated by the program may provide an immersive experience. The audio content of the program may be indicative of multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects, and typically also a default set of objects which will be rendered in the absence of a selection by a user) and a bed of speaker channels. Another aspect is an audio processing unit (e.g., encoder or decoder) configured to perform, or which includes a buffer memory which stores at least one frame (or other segment) of an object based audio program (or bitstream thereof) generated in accordance with, any embodiment of the method.

Abstract: There is provided an information processing device to reduce a processing load associated with inner product operations while also guaranteeing the quantization granularity of weight coefficients, the information processing device including: a computational unit configured to compute an output value by performing an inner product operation based on multiple input values and multiple weight coefficients corresponding to each of the input values. The computational unit computes the output value on the basis of the weight coefficients quantized on the basis of a granularity of a vector direction on an N-dimensional hypersphere surface. Moreover, provided is an information processing device including: a multiply-accumulate operation circuit configured to execute a multiply-accumulate operation based on multiple input values and multiple weight coefficients corresponding to each of the input values.

Abstract: The technology disclosed herein may determine timing windows for speech captions of an audio stream. In one example, the technology may involve accessing audio data comprising a plurality of segments; determining, by a processing device, that one or more of the plurality of segments comprise speech sounds; identifying a time duration for the speech sounds; and providing a user interface element corresponding to the time duration for the speech sounds, wherein the user interface element indicates an estimate of a beginning and ending of the speech sounds and is configured to receive caption text associated with the speech sounds of the audio data.

Abstract: A frequency-domain audio codec is provided with the ability to additionally support a certain transform length in a backward-compatible manner, by the following: the frequency-domain coefficients of a respective frame are transmitted in an interleaved manner irrespective of the signalization signaling for the frames as to which transform length actually applies, and additionally the frequency-domain coefficient extraction and the scale factor extraction operate independent from the signalization. By this measure, old-fashioned frequency-domain audio coders/decoders, insensitive for the signalization, would be able to nevertheless operate without faults and with reproducing a reasonable quality. Concurrently, frequency-domain audio coders/decoders able to support the additional transform length would offer even better quality despite the backward compatibility.

Abstract: An audio processor for processing an audio signal to acquire a subband representation thereof includes a cascaded lapped critically sampled transform stage and a time domain aliasing reduction stage, the former being configured to perform a cascaded lapped critically sampled transform on at least two partially overlapping blocks of samples of the audio signal, to acquire a set of subband samples on the basis of a first block of samples of the audio signal, and to acquire a corresponding set of subband samples on the basis of a second block of samples of the audio signal. The latter is configured to perform a weighted combination of two corresponding sets of subband samples, which are acquired on the basis of the first and second blocks of samples of the audio signal, respectively, to acquire an aliasing reduced subband representation of the audio signal.

Abstract: In one implementation, the disclosure provides a system including a detector configured to generate an output of a first log-likelihood ratio for each bit in an input data stream. The system also includes at least one look-up table providing a mapping of the first log-likelihood ratio to a second log-likelihood ratio. The mapping between the first log-likelihood ratio and the second log-likelihood ratio is non-linear. The system also includes a decoder configured to generate an output data stream using the second log-likelihood ratio to generate a value for each bit in the input data stream.

Abstract: Techniques of playing back a looping video file involve providing multiple video codecs for decoding a video file. Each video codec performs its own decoding operation on the looping video file, resulting in multiple buffers of video frames and a buffer of audio frames. Then, as the GPU renders the ending video frames from a first buffer, it begins rendering the beginning video frames from a second buffer. In this way, the beginning of a next video cycle is already rendered for viewing by the time a current video cycle is ending.

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: Scene and/or state information may be used to facilitate processing an input to separate one or more signals within the input, to shape the signal within the input, and/or for other processing of the input or signal(s) within the input. A scene determination may be made based upon location data, time data, data describing the received input, or other basis. A state determination may be made based upon the scene determination, properties of a signal itself, or other information such as location, time, etc. By determining an appropriate scene and/or state, processing of an input and/or a signal within an input may proceed in a fashion determined to provide the most valuable information for output. Systems and methods in accordance with the invention may be implemented in a wide variety of baseband processing systems, such as hearing aids and energy consumption monitoring systems.

Abstract: A coding efficiency of coding spectral coefficients of a spectrum of an audio signal is increased by en/decoding a currently to be en/decoded spectral coefficient by entropy en/decoding and, in doing so, performing the entropy en/decoding depending, in a context-adaptive manner, on a previously en/decoded spectral coefficient, while adjusting a relative spectral distance between the previously en/decoded spectral coefficient and the currently en/decoded spectral coefficient depending on an information concerning a shape of the spectrum.

Abstract: In some aspects, the disclosure is directed to methods and systems for improving signal to noise ratios of signals from multiple communication links. In some embodiments, a system includes a first frequency transformation circuit configured to transform a first signal in a time domain received from a first device into a corresponding second signal in a frequency domain. The system further includes a second frequency transformation circuit configured to transform a third signal in the time domain received from a second device into a corresponding fourth signal in the frequency domain. The system further includes a leg combining circuit configured to select, for a group of subcarriers, one of the first frequency transformation circuit and the second frequency transformation circuit, and cause, for the group of subcarriers, the selected frequency transformation circuit to output one of the second signal and the fourth signal, according to the selection.

Abstract: In one implementation, the disclosure provides a system including a detector configured to generate an output of a first log-likelihood ratio for each bit in an input data stream. The system also includes at least one look-up table providing a mapping of the first log-likelihood ratio to a second log-likelihood ratio. The mapping between the first log-likelihood ratio and the second log-likelihood ratio is non-linear. The system also includes a decoder configured to generate an output data stream using the second log-likelihood ratio to generate a value for each bit in the input data stream.

Abstract: The invention provides a decoder being configured for processing an encoded audio bitstream, wherein the decoder includes: a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal includes at least one decoded frame; a noise estimation device configured to produce a noise estimation signal containing an estimation of the level and/or the spectral shape of a noise in the decoded audio signal; a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; and a combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to obtain an audio output signal.

Abstract: A method, device, system and kit for preventing the intelligible voice recording Is provided. The voice of a subject or Interlocutor Is recorded for a given time Interval thereby providing a voice recording. The voice recording Is cut Into shorter time interval segments thereby providing a set of voice recording segments. The set of voice recording segments is mixed in a randomly rearranged order, The mixed set of voice recording segments is spliced into a single randomly mixed voice recording. Emitting the randomly mixed voice recording during speaking of the subject or interlocutor prevents the Intelligible recording of the voice of the subject or interlocutor.

Abstract: An embedded codec (EBC) circuitry includes encoder circuitry to determine a refinement start position in a bit-plane of an encoded data block based on a random number. The refinement start position is a position in the bit-plane based on a value of the random number. The encoder circuitry determines a refinement order in the bit-plane for refining un-coded bits present in the bit-plane, based on the determined refinement start position and a refinement step size. The refinement order is a sequence of positions of the un-coded bits in the bit-plane that will be refined in that sequence. The encoder circuitry refines the un-coded bits by allocating a refinement bit at the refinement start position in the bit-plane and then followed by the allocation of subsequent refinement bits in the determined refinement order.

Abstract: A decoder generates decoded signals based on quantized signals. The decoder includes an inverse quantizer and a predictor circuit. The quantized signals are generated in an encoder by low-pass filtering an input signal and encoding the filtered signal using adaptive differential pulse code modulation. The predictor circuit has filter coefficients based on a frequency response of the low-pass filter used to filter the input signal.

Abstract: An audio signal processing device includes a high pass filter to convert an input audio signal into a first audio signal and to output the first audio signal, a displacement estimation unit to estimate displacement amplitude of a speaker diaphragm when the input audio signal is inputted, a saturation processing unit to perform saturation processing to the displacement amplitude estimated in the displacement estimation unit or to a signal obtained by correcting the displacement amplitude, an audio signal generation unit to generate a second audio signal by using displacement amplitude obtained after the saturation processing in the saturation processing unit, and an output generation unit to generate an output signal by using the first and the second audio signals. With this configuration, a cracking sound of a speaker is reduced and a user is allowed to feel low frequency components.

Abstract: An encoder generates quantized signal words based on a difference signal. The encoder includes an adaptive quantizer. A step size applied by the adaptive quantizer is generated in a feedback loop and based on a loading factor and quantized signal words generated by the adaptive quantizer. The encoder includes coding circuitry which generates code words based on quantized signal words generated by the adaptive quantizer. The coding circuitry generates an escape code in response to a quantized signal word not being associated with a corresponding coding code word.

Abstract: The present disclosure relates to a method and to an apparatus, both arranged for controlling acoustic signals to be recorded or reproduced by an electro-acoustical sound system. An initial digital filter is determined by solving an inverse problem, wherein the initial digital filter is configured to control acoustic signals to be recorded and/or reproduced by the electro-acoustical sound system; a frequency-dependent articulation parameter is determined by executing a time spectral psychoacoustic automatic audio quality test on the initial digital filter; a frequency-dependent regularization parameter, used for determining the initial digital filter, is tuned by use of the frequency-dependent articulation parameter; and, by use of the tuned frequency-dependent regularization parameter, a digital filter configured to control acoustic signals to be recorded or reproduced by the electro-acoustical sound system is determined.

Abstract: The present disclosure relates to the field of mobile communications, and in particular, to a data compression method and device. In order to solve the problem of low compression efficiency in the prior art, the method comprises: sampling an obtained baseband signal and obtaining a number of discrete baseband signals to achieve an initial compression; calculating amplitude and phase values of each discrete baseband signal, on the basis of bit widths preset for the adjusted amplitude value and the phase value, carrying out a bit-truncating on the adjusted amplitude value and the phase value respectively, and combining the truncated phase value data and amplitude value data to obtain the final compressed discrete baseband signal. In this way, by truncating bits according to the preset bit width without distortion, the data bits of the baseband signal can be reduced accordingly, thereby reducing the amount of transmitted data, effectively improving the compression efficiency, and thus saving fiber resources.

Abstract: A sampler module divides an audio signal into a series of sequential samples. A signal quality detector module identifies, over a plurality of samples at an outset of the audio signal, a spectral variance of a first range of frequencies of the audio signal below a predetermined threshold frequency as being consistently greater than a spectral variance of a second range of frequencies of the audio signal above the predetermined threshold frequency and determines a signal treatment indication responsive to the identification. A signal enhancer module sequentially receives and analyzes one or more sample components of the audio signal to identify lost parts of the audio signal in the one or more sample components of respective sequential samples, and generates, in accordance with the signal treatment indication, a corresponding signal treatment for each of the one or more sample components of respective sequential samples having a corresponding identified lost part.

Abstract: In general, techniques are described by which to perform audio coding based on audio pattern recognition. A source device comprising a memory and a processor may be configured to perform the techniques. The memory may store audio data. The processor may obtain, from a plurality of categories, a category to which the audio data corresponds, and obtain, based on the category, a set of pyramid vector quantization (PVQ) parameters from a plurality of sets of PVQ parameters. The processor may also perform, based on the set of PVQ parameters, PVQ with respect to the audio data to obtain a residual identifier representative of the audio data, and specify, in the bitstream, the residual identifier.

Abstract: A method for playing a media file in an electronic device is provided. The method includes running, on an electronic device having an operating system (OS), a media presentation system, receiving, at the media presentation system, from a source a media file including audio and video signals, wherein the video signals include the video container of a form suitable for representation on a video output component of the electronic device, playing, at the media presentation system, the media file by processing the audio and video signals, and manipulating, at the media presentation system, the video container depending on an operating state of one or both of the media presentation system and the electronic device while the audio and video signals are being processed.

Abstract: An encoder includes a low-pass filter to filter input audio signals. The low-pass filter has fixed filter coefficients. The encoder generates quantized signals based on a difference signal. The encoder includes an adaptive quantizer and a decoder to generate feedback signals. The decoder has an inverse quantizer and a predictor. The predictor has fixed control parameters which are based on a frequency response of the low-pass filter. The predictor may include a finite impulse response filter having fixed filter coefficients. The decoder may include an adaptive noise shaping filter coupled between the low-pass filter and the encoder. The adaptive noise shaping filter flattens signals within a frequency spectrum corresponding to a frequency spectrum of the low-pass filter.

Abstract: An audio encoder for providing an encoded representation on the basis of an audio signal, wherein the audio encoder is configured to obtain a noise information describing a noise included in the audio signal, and wherein the audio encoder is configured to adaptively encode the audio signal in dependence on the noise information, such that encoding accuracy is higher for parts of the audio signal that are less affected by the noise included in the audio signal than for parts of the audio signal that are more affected by the noise included in the audio signal.

Abstract: Methods and devices for processing and voice operated control are provided. The method can include performing a non-difference comparison between a first received sound and a second received sound, determining if speech exists based on the comparison, and transmitting or providing a decision that the speech is present to at least one among the device, a cell phone, a media player, or a portable computing device. Other embodiments are disclosed.

Abstract: Disclosed is an audio signal processing device including a processor for outputting an output audio signal generated based on an input audio signal. The processor may be configured to obtain a first pair of head-related transfer function (HRTF)s including a first ipsilateral HRTF and a first contralateral HRTF based on a position of a virtual sound source corresponding to the input audio signal, from a first set of transfer functions including HRTFs corresponding to each specific position with respect to listener, and generate the output audio signal by performing binaural rendering the input audio signal based on the first pair of HRTFs.

Abstract: Apparatus for decoding an encoded audio signal including an encoded core signal and parametric data, including: a core decoder for decoding the encoded core signal to obtain a decoded core signal; an analyzer for analyzing the decoded core signal before or after performing a frequency regeneration operation to provide an analysis result; and a frequency regenerator for regenerating spectral portions not included in the decoded core signal using a spectral portion of the decoded core signal, the parametric data, and the analysis result.

Abstract: An encoder for encoding an audio signal has a predictor, a factorizer, a transformer and a quantize and encode stage. The predictor is configured to analyze the audio signal to obtain prediction coefficients describing a spectral analog of the audio signal or a fundamental frequency of the audio signal and subject the audio signal to an analysis filter function dependent on the prediction coefficients to output a residual signal of the audio signal. The factorizer is configured to apply a matrix factorization onto an audiocorrelation or covariance matrix of synthesis filter function defined by the prediction coefficients to obtain factorized matrices. The transformer is configured to transform the residual signal based on the factorized matrices to obtain a transformed residual signal. The quantize and decode stage is configured to quantize the transformed residual signal to obtain a quantized transformed residual signal or an encoded quantized transformed residual signal.

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

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part.