Abstract: The signal separation device includes: cross product calculation means receiving an input of an observed signal that is a mixture of a plurality of target signals, and calculating a cross product of the observed signal; model calculation means updating a parameter of a model for estimating the cross product with a predetermined algorithm using an inverse matrix of a matrix that represents an estimate of the cross product; inverse matrix calculation means calculating the inverse matrix of a matrix by a SIMD command when the parameter is updated; and separation means calculating the target signals using a matrix representing an estimate of the cross product, the updated parameter, and the observed signal.

Abstract: An apparatus, comprising: an audio input for receiving an input audio signal; an tuning input for receiving a tuning signal; a filter chain comprising a plurality of filters for filtering the audio signal to produce a filtered input audio signal, the filter chain comprising: a first filter module operating at a first sampling rate; and a second filter module operating at a second sampling rate greater than the first sampling rate, wherein a phase response of the first filter module is dependent on the tuning input and wherein a magnitude response of the first filter module is substantially independent of the tuning input.

Abstract: The present disclosure proposes a speech conversion scheme for non-parallel corpus training, to get rid of dependence on parallel text and resolve a technical problem that it is difficult to achieve speech conversion under conditions that resources and equipment are limited. A voice conversion system and a training method therefor are included. Compared with the prior art, according to the embodiments of the present disclosure: a trained speaker-independent automatic speech recognition model can be used for any source speaker, that is, the speaker is independent; and bottleneck features of audio are more abstract as compared with phonetic posteriorGram features, can reflect decoupling of spoken content and timbre of the speaker, and meanwhile are not closely bound with a phoneme class, and are not in a clear one-to-one correspondence relationship. In this way, a problem of inaccurate pronunciation caused by a recognition error in ASR is relieved to some extent.

Abstract: An audio processor for generating a frequency enhanced audio signal from a source audio signal has: an envelope determiner for determining a temporal envelope of at least a portion of the source audio signal; an analyzer for analyzing the temporal envelope to determine temporal values of certain features of the temporal envelope; a signal synthesizer for generating a synthesis signal, the generating having placing pulses in relation to the determined temporal values, wherein the pulses are weighted using weights derived from amplitudes of the temporal envelope related to the temporal values, where the pulses are placed; and a combiner for combining at least a band of the synthesis signal that is not included in the source audio signal and the source audio signal to obtain the frequency enhanced audio signal.

Abstract: Methods and apparatus to extract a pitch-independent timbre attribute from a media signal are disclosed. An example apparatus includes an audio characteristic extractor to determine a logarithmic spectrum of an audio signal; transform the logarithmic spectrum of the audio signal into a frequency domain to generate a transform output; determine a magnitude of the transform output; and determine a timbre attribute of the audio signal based on an inverse transform of the magnitude.

Abstract: To provide a bandwidth extension method which allows reduction of computation amount in bandwidth extension and suppression of deterioration of quality in the bandwidth to be extended. In the bandwidth extension method: a low frequency bandwidth signal is transformed into a QMF domain to generate a first low frequency QMF spectrum; pitch-shifted signals are generated by applying different shifting factors on the low frequency bandwidth signal; a high frequency QMF spectrum is generated by time-stretching the pitch-shifted signals in the QMF domain; the high frequency QMF spectrum is modified; and the modified high frequency QMF spectrum is combined with the first low frequency QMF spectrum.

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: A method for training parameters of a first domain adaptation model. The method includes evaluating a cycle consistency objective using a first task specific model associated with a first domain and a second task specific model associated with a second domain, and evaluating one or more first discriminator models to generate a first discriminator objective using the second task specific model. The one or more first discriminator models include a plurality of discriminators corresponding to a plurality of bands that corresponds domain variable ranges of the first and second domains respectively. The method further includes updating, based on the cycle consistency objective and the first discriminator objective, one or more parameters of the first domain adaptation model for adapting representations from the first domain to the second domain.

Abstract: A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

Abstract: An adjustment method of sound output is disclosed. The adjustment method includes the following steps of: receiving an audio message having a vowel message; determining whether the audio message is a whispered voice message; if the audio message is a whispered voice message, outputting a normal voice message, wherein the spoken content of the normal voice message is the same as that of the audio message, and the normal voice message has a normal voice vowel message, wherein the sound energy of the low-frequency part of the normal voice vowel message is 1.5-1,000,000 times that of the vowel message.

Abstract: A signal processing unit performs, on the basis of a received electric field signal from an antenna by which a beam is scanned within a predetermined azimuthal angle and a signal of an azimuthal angle of the scanned beam, a Fourier transform on a distribution function of the received electric field signal into a frequency domain of the azimuthal angle, divides a signal according to a first spectral function by a signal according to a second spectral function, the first spectral function being obtained by performing the Fourier transform, the second spectral function being obtained by performing a Fourier transform on an antenna pattern of the antenna into a frequency domain of the azimuthal angle, and subjects the divided signal to fitting by using Prony's method with exponential functions including real parts and imaginary parts in arguments.

Abstract: An audio decoder for providing at least four bandwidth-extended channel signals on the basis of an encoded representation provides first and second downmix signals on the basis of a jointly encoded representation of the first and second downmix signals using a multi-channel decoding and provides at least first and second audio channel signals on the basis of the first downmix signal using a multi-channel decoding, and provides at least third and fourth audio channel signals on the basis of the second downmix signal using a multi-channel decoding. It performs a multi-channel bandwidth extension on the basis of the first and third audio channel signals, to obtain first and third bandwidth-extended channel signals, and performs a multi-channel bandwidth extension on the basis of the second and fourth audio channel signals, to obtain second and fourth bandwidth extended channel signals. An audio encoder uses a related concept.

Abstract: A method for decoding a digital signal encoded using predictive coding and transform coding, comprising the following steps: predictive decoding of a preceding frame of the digital signal, encoded by a set of predictive coding parameters; detecting the loss of a current frame of the encoded digital signal; generating by prediction, from at least one predictive coding parameter encoding the preceding frame, a frame for replacing the current frame; generating by prediction, from at least one predictive coding parameter encoding the preceding frame, an additional segment of digital signal; temporarily storing said additional segment of digital signal.

Abstract: The purpose of the present invention is to reduce distortion a frequency band component encoded with a small number of bits in a time domain and improve quality. An audio decoding device (10) decodes an encoded audio signal and outputs the audio signal. A decoding unit (10a) decodes an encoded sequence containing an encoded audio signal and obtains a decoded signal. A selective temporal envelope shaping unit (10b) shapes a temporal envelope of a decoded signal in the frequency band on the basis of decoding related information concerning decoding of the encoded sequence.

Abstract: The present invention relates to a method for receiving data transmitted acoustically. The method includes receiving an acoustically transmitted signal encoding data; processing the received signal to minimise environmental interference within the received signal; and decoding the processed signal to extract the data. The data encoded within the signal using a sequence of tones. A method for encoding data for acoustic transmission is also disclosed. This method includes encoding data into an audio signal using a sequence of tones. The audio signal in this method is configured to minimise environmental interference. A system and software are also disclosed.

Abstract: Embodiments of the present disclosure provide a quantization method of feature vector, search method, apparatus and storage medium, wherein the quantization method comprises: setting quantization codewords; selecting, from the set quantization codewords, at least one quantization codeword of which distance from an original feature vector is smaller than first predetermined distance as a quantization codeword subset; determining quantization feature vectors corresponding to the original feature vector by using at least two quantization codewords, wherein at least one quantization codeword from the at least two quantization codewords are in the quantization codeword subset. The search method is a method of performing a search among quantization feature vectors after a quantization using the quantization method.

Abstract: A system and method for masking an identity of a speaker of natural language speech, such as speech clips to be labeled by humans in a system generating voice transcriptions for training an automatic speech recognition model. The natural language speech is morphed prior to being presented to the human for labeling. In one embodiment, morphing comprises pitch shifting the speech randomly either up or down, then frequency shifting the speech, then pitch shifting the speech in a direction opposite the first pitch shift.

Abstract: A method of operating a hearing aid system in order to provide improved performance for a multitude of hearing aid system processing stages and a hearing aid system (400) for carrying out the method.

Abstract: In a system for improving performance of tensor-based computations and for minimizing the associated memory usage, computations associated with different non-zero tensor values are performed while exploiting an overlap between the respective index tuples of those non-zero values. While performing computations associated with a selected mode, when an index corresponding to a particular mode in a current index tuple is the same as the corresponding index from another, previously processed index tuple, the value already stored in a buffer corresponding to that particular mode is reused either wholly or in part, minimizing the processor usage and improving performance. Certain matrix operations may be iterated more than once so as to avoid the need to store a large partial result obtained from those operations. The performance overhead of the repeated operations is not significant, but the reduction in memory usage is.

Abstract: A controller and method of classifying a user into one of a plurality of user classes. One or more voice samples are received from the user, from which a frequency spectrum is generated. One or more values defining respective features of the frequency spectrum are extracted from the frequency spectrum. Each of the respective features are defined by values of frequency, amplitude, and/or position in the spectrum. One or more of the respective features are resonant frequencies in the voice of the user. A user profile of the user is generated and comprises the extracted one or more values. The user profile is supplied to a machine learning algorithm that is trained to classify users as belonging to one of the plurality of user classes based on the one or more values in their respective user profile.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed to reduce noise from harmonic noise sources. An example apparatus includes a contour tracer to determine a first point of comparatively large amplitude of a frequency component in a frequency spectrum of an audio sample, determine a set of points in the frequency spectrum having amplitude values within an amplitude threshold of the first point, frequency values within a frequency threshold of the first point, and phase values within a phase threshold of the first point, increment a counter when a distance between (1) a second point in the set of points and (2) the first point satisfies a distance threshold, and when the counter satisfies a counter threshold, generate the contour trace, the contour trace including the set of points, and a subtractor to remove the contour trace from the audio sample when the amplitude values satisfy an outlier threshold.

Abstract: A method for optimising the transmission power of an FM radio broadcasting transmitter includes sampling a signal representative of the audio content to be broadcasted by the FM radio broadcasting transmitter; continuously calculating the constituent parameters of the representative signal from frequency, amplitude, dynamic range, temporal distribution, energy and power; continuously analysing the parameters by comparison with a model of psycho-acoustic data; generating a controlling signal for controlling the power of the transmitter as a function of the results of the analysis and of the calculations allowed by the constituent parameters and the continuous psycho-acoustic data; driving of the RF power of the transmitter by means of the controlling signal. A device for implementing the method in an FM radio broadcasting transmitter is also proposed.

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 a method, a device, a system and/or a manufacture of control and security of a voice controlled device. In one embodiment, an apparatus includes a signal input for receiving a command signal and a soundwave damper to dampen a sound receiving capability of a voice controlled device. The voice controlled device is capable of initiating a responsive action in response to a voice command. A processor determinations that the command signal is associated with the voice command and in response to the determination generate with a speaker a soundwave carrying the voice command bypassing the soundwave damper such that the soundwave carrying the voice command is communicable to the microphone of the voice controlled device. The apparatus also includes a cap housing the speaker and optionally the soundwave damper. As an example, the soundwave damper may be a barrier, a flexible membrane, or a sound interference generator.

Abstract: A signal processing apparatus that handles an adaptive enhancement of a speech signal, receives a first signal and a second signal from a determined source. At least one of a speech signal or at least one noise signal is present in the first signal or the second signal. The first signal and the received second signal are processed to obtain a processed signal for amplification of a gain associated with the speech signal present in the first signal and the second signal by a determined factor. A signal-to-noise ratio (SNR) associated with the processed signal is greater than or equal to a threshold value. A reference noise signal is obtained from the second signal based on subtraction of an estimated the speech signal present in the received second signal from the processed signal. A processed speech signal is determined based on filtration of the obtained reference noise signal.

Abstract: A decoder for decoding an encoded audio signal to obtain a phase-adjusted audio signal is provided. The decoder has a decoding unit and a phase adjustment unit. The decoding unit is adapted to decode the encoded audio signal to obtain a decoded audio signal. The phase adjustment unit is adapted to adjust the decoded audio signal to obtain the phase-adjusted audio signal. The phase adjustment unit is configured to receive control information depending on a vertical phase coherence of the encoded audio signal. Moreover, the phase adjustment unit is adapted to adjust the decoded audio signal based on the control information.

Abstract: A voice signal decoding device includes a first decoder, a second decoder, a signal switch, and a noise adder. The first decoder decodes first encoded data encoded by a first encoding method. The second decoder decodes second encoded data encoded by a second encoding method. The second encoded data has a narrower band than a band of the first encoded data. The signal switch switches an output signal of the first decoder and an output signal of the second decoder. The noise adder adds a noise signal to a high-frequency band in the output signal of the second decoder when the signal switch switches an output signal from the output signal of the first decoder to the output signal of the second decoder. The high-frequency band is a band where a signal component is lacking as compared with the output signal of the first decoder.

Abstract: The invention discloses systems and methods for enhancing the sound of vocal utterances of interest in an acoustically cluttered environment. The system generates canceling signals (sound suppression signals) for an ambient audio environment and identifies and characterizes desired vocal signals and hence a vocal stream or multiple streams of interest. Each canceling signal, or collectively, the noise canceling stream, is processed so that signals associated with the desired audio stream or streams are dynamically removed from the canceling stream. This modified noise canceling stream is combined (electronically or acoustically) with the ambient to effectuate a destructive interference of all ambient sound except for the removed audio streams, thus “enhancing” the vocal streams with respect to the unwanted ambient sound. Cepstral analysis may be used to identify a fundamental frequency associated with a voiced human utterance.

Abstract: An apparatus for analysing a magnitude spectrogram of an audio signal is provided. The apparatus includes a frequency change determiner being configured to determine a change of a frequency for each time-frequency bin of a plurality of time-frequency bins of the magnitude spectrogram of the audio signal depending on the magnitude spectrogram of the audio signal. Moreover, the apparatus includes a classifier being configured to assign each time-frequency bin of the plurality of time-frequency bins to a signal component group of two or more signal component groups depending on the change of the frequency determined for the time-frequency bin.

Abstract: An audio decoder for providing at least four bandwidth-extended channel signals on the basis of an encoded representation provides first and second downmix signals on the basis of a jointly encoded representation of the first and second downmix signals using a multi-channel decoding and provides at least first and second audio channel signals on the basis of the first downmix signal using a multi-channel decoding, and provides at least third and fourth audio channel signals on the basis of the second downmix signal using a multi-channel decoding. It performs a multi-channel bandwidth extension on the basis of the first and third audio channel signals, to obtain first and third bandwidth-extended channel signals, and performs a multi-channel bandwidth extension on the basis of the second and fourth audio channel signals, to obtain second and fourth bandwidth extended channel signals. An audio encoder uses a related concept.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed to reduce noise from harmonic noise sources. Example methods disclosed herein include determining a point of comparatively large amplitude for a frequency in an audio sample, generating a contour trace beginning at the point, and calculating a parameter for the contour trace. The example method includes, in response to the contour trace being an outlier based on the parameter, removing the contour trace from the audio sample.

Abstract: A wireless communication device includes an equalizer configured to adjust a frequency characteristic of a user audio signal on the basis of a predetermined gain setting and to generate an adjusted signal, or an adjusted audio signal; an encoder configured to encode the adjusted signal with a preset scheme to generate an encoded signal; a decoder configured to decode the encoded signal to generate a decoded signal; a sound quality evaluator configured to carry out a PESQ evaluation of the decoded signal and to generate an evaluation result with the user audio signal serving as a reference signal for evaluation; and a selector configured to change the gain setting and to select a gain setting yielding a high evaluation result.

Abstract: An apparatus for selecting one of a first encoding algorithm and a second encoding algorithm includes a filter configured to receive the audio signal, to reduce the amplitude of harmonics in the audio signal and to output a filtered version of the audio signal. First and second estimators are provided for estimating first and second quality measures in the form of SNRs of segmented SNRs associated with the first and second encoding algorithms without actually encoding and decoding the portion of the audio signal using the first and second encoding algorithms. A controller is provided for selecting the first encoding algorithm or the second encoding algorithm based on a comparison between the first quality measure and the second quality measure.

Abstract: A speaker identification/verification system comprises at least one feature extractor for extracting a plurality of audio features from speaker voice data, a plurality of speaker-specific subsystems, and a decision module. Each of the speaker-specific subsystem comprises: a neural network configured to generate an estimate of the plurality of extracted audio features based on the plurality of extracted audio features, and an error module. Each of the plurality of neural networks is associated with one of a plurality of speakers, and the one speaker associated with each of the plurality of neural networks is different for all neural networks. The error module is configured to estimate an error based on the plurality of extracted audio features and the estimate of the plurality of extracted audio features generated by the associated neural network. The neural networks are speaker-specific auto-encoders trained for one user and therefore calibrated on that particular user's speech.

Abstract: An audio encoder has a first information sink oriented encoding branch, a second information source or SNR oriented encoding branch, and a switch for switching between the first encoding branch and the second encoding branch, wherein the second encoding branch has a converter into a specific domain different from the spectral domain, and wherein the second encoding branch furthermore has a specific domain coding branch, and a specific spectral domain coding 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 for decoding a signal, and a third domain decoder and two cascaded switches for switching between the decoders.

Abstract: In a transmit method, a set of data eigenvectors that are based on a Prometheus Orthonormal Set (PONS) code construction and orthogonal to each other are stored, wherein the data eigenvectors are mapped to unique multi-bit words. A pilot sequence representing a pilot eigenvector that is based on the PONS code construction and orthogonal to each of the data eigenvectors is generated. Input data are grouped into multi-bit words and data eigenvectors among the data eigenvectors are selected based on the multi-bit words. A spread data sequence including the selected data eigenvectors and that is synchronized to the pilot sequence is generated. An acoustic signal including the synchronized pilot sequence and the spread data sequence is generated. The acoustic signal is transmitted.

Abstract: Systems and methods for an audio signal processing system for transforming an input audio signal. A processor implements steps of a module by inputting an input audio signal into a spectrogram estimator to extract an audio feature sequence, and process the audio feature sequence to output a set of estimated spectrograms. Processing the set of estimated spectrograms and the audio feature sequence using a spectrogram refinement module, to output a set of refined spectrograms. Wherein the processing of the spectrogram refinement module is based on an iterative reconstruction algorithm. Processing the set of refined spectrograms for the one or more target audio signals using a signal refinement module, to obtain the target audio signal estimates. An output interface to output the optimized target audio signal estimates. Wherein the module is optimized by minimizing an error using an optimizer stored in the memory.

Abstract: A sound source direction estimation device includes: a phase difference calculator which calculates, from an acoustic signal obtained by a microphone array, a first phase difference of a pair of microphone units; a similarity calculator which calculates similarities between the calculated first phase difference and second phase differences precalculated for directions and stored in a phase difference database; a peak searcher which searches for a direction for which a highest similarity is calculated by the similarity calculator, and estimates the direction searched out to be a sound source direction; a feature quantity calculator which uses the calculated similarities, the estimated sound source direction, and an acoustic feature quantity obtained from the obtained acoustic signal, to calculate a feature quantity obtained by correcting the acoustic feature quantity; and a speech/non-speech determiner which determines whether the obtained acoustic signal indicates speech, using the feature quantity calculated

Abstract: The present invention is provided with: a voice input section that receives a remark (a question) via a voice signal; a reply creation section that creates a voice sequence of a reply (response) to the remark; a pitch analysis section that analyzes the pitch of a first segment (e.g., word ending) of the remark; and a voice generation section (a voice synthesis section, etc.) that generates a reply, in the form of voice, represented by the voice sequence. The voice generation section controls the pitch of the entire reply in such a manner that the pitch of a second segment (e.g., word ending) of the reply assumes a predetermined pitch (e.g., five degrees down) with respect to the pitch of the first segment of the remark. Such arrangements can realize synthesis of replying voice capable of giving a natural feel to the user.

Abstract: Audio objects that are present in input audio content in one or more frames are determined. Output clusters that are present in output audio content in the one or more frames are also determined. Here, the audio objects in the input audio content are converted to the output clusters in the output audio content. One or more spatial error metrics are computed based at least in part on positional metadata of the audio objects and positional metadata of the output clusters.

Abstract: A method and device are provided for determining an optimized scale factor to be applied to an excitation signal or a filter during a process for frequency band extension of an audio frequency signal. The band extension process includes decoding or extracting, in a first frequency band, an excitation signal and parameters of the first frequency band including coefficients of a linear prediction filter, generating an excitation signal extending over at least one second frequency band, filtering using a linear prediction filter for the second frequency band. The determination method includes determining an additional linear prediction filter, of a lower order than that of the linear prediction filter of the first frequency band, the coefficients of the additional filter being obtained from the parameters decoded or extracted from the first frequency and calculating the optimized scale factor as a function of at least the coefficients of the additional filter.

Abstract: A method and device are provided for determining an optimized scale factor to be applied to an excitation signal or a filter during a process for frequency band extension of an audio frequency signal. The band extension process includes decoding or extracting, in a first frequency band, an excitation signal and parameters of the first frequency band including coefficients of a linear prediction filter, generating an excitation signal extending over at least one second frequency band, filtering using a linear prediction filter for the second frequency band. The determination method includes determining an additional linear prediction filter, of a lower order than that of the linear prediction filter of the first frequency band, the coefficients of the additional filter being obtained from the parameters decoded or extracted from the first frequency and calculating the optimized scale factor as a function of at least the coefficients of the additional filter.

Abstract: A method for processing a digital audio signal, including a series of samples distributed in consecutive frames, is implemented when decoding the signal in order to replace at least one signal frame lost during decoding. The method includes the following steps: a) searching, in a valid signal segment available when decoding, for at least one period in the signal, determined in accordance with the valid signal; b) analyzing the signal in the period, in order to determine spectral components of the signal in the period; c) synthesizing at least one frame for replacing the lost frame, by construction of a synthesis signal from: an addition of components selected among the predetermined spectral components, and a noise added to the addition of components. In particular, the amount of noise added to the addition of components is weighted in accordance with voice information of the valid signal, obtained when decoding.

Abstract: Disclosed are a method and an apparatus for high frequency decoding for bandwidth extension. The method for high frequency decoding for bandwidth extension comprises the steps of: decoding an excitation class; transforming a decoded low frequency spectrum on the basis of the excitation class; and generating a high frequency excitation spectrum on the basis of the transformed low frequency spectrum. The method and apparatus for high frequency decoding for bandwidth extension according to an embodiment can transform a restored low frequency spectrum and generate a high frequency excitation spectrum, thereby improving the restored sound quality without an excessive increase in complexity.

Abstract: An apparatus for generating an encoded signal includes: a window sequence controller for generating a window sequence information for windowing an audio or image signal, the window sequence information indicating a first window for generating a first frame of spectral values, a second window function and at least one third window function for generating a second frame of spectral values, wherein the first window function, the second window function and the one or more third window functions overlap within a multi-overlap region; a preprocessor for windowing a second block of samples corresponding to the second window function and the at least one third window functions using an auxiliary window function to acquire a second block of windowed samples, a spectrum converter for applying an aliasing-introducing transform; and a processor for processing the first frame and the second frame to acquire encoded frames of the audio or image signal.

Abstract: Methods and systems for audio source separation in real-time are described. In an embodiment, the present disclosure describes reading and decoding an audio source into PCM samples, fragmenting Pulse Code Modulation (PCM) samples into fragments, transforming fragments into spectrograms, performing audio source separation using a training database that includes a training dictionary and non-negative matrix factorization (NMF) to generate a set of component signals, and streaming the component signals to a playback engine. In an embodiment, a semantic equalizer graphical user allows for fading of individual component signals.

Abstract: An audio encoder has a first information sink oriented encoding branch, a second information source or SNR oriented encoding branch, and a switch for switching between the first encoding branch and the second encoding branch, wherein the second encoding branch has a converter into a specific domain different from the spectral domain, and wherein the second encoding branch furthermore has a specific domain coding branch, and a specific spectral domain coding 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 for decoding a signal, and a third domain decoder and two cascaded switches for switching between the decoders.

Abstract: A voice conversion device includes: a parameter learning unit in which a probabilistic model that uses speech information, speaker information, and phonological information as variables to thereby express relationships among binding energies between any two of the speech information, the speaker information and the phonological information by parameters is prepared, wherein the speech information is obtained based on a speech, the speaker information corresponds to the speech information, and the phonological information expresses the phoneme of the speech, and in which the parameters are determined by performing learning by sequentially inputting the speech information and the speaker information into the probabilistic model; and a voice conversion processing unit that performs voice conversion processing of the speech information obtained on the basis of the speech of an input speaker, based both on the parameters determined by the parameter learning unit and on the speaker information of a target speaker.

Abstract: A method for controlling a plurality of environmental factors that trigger a negative emotional state is provided. The method may include analyzing a plurality of user data when a user experiences a plurality of various environmental factors. The method may also include determining an emotional state experienced by the user when each of the plurality of various environmental factors is present based on the plurality of user data. The method may include receiving a plurality of calendar information associated with a user account. The method may also include identifying an upcoming event based on the plurality of calendar information. The method may include identifying an environmental factor within the plurality of various environmental factors is present at the upcoming event. The method may also include, in response to determining the environmental factor causes the user to experience a negative emotional state, executing an accommodation method based on the environmental factor.

Abstract: According to an aspect of the present invention, a method for reconstructing an audio signal having a baseband portion and a highband portion is disclosed. The method includes obtaining a decoded baseband audio signal by decoding an encoded audio signal and obtaining a plurality of subband signals by filtering the decoded baseband audio signal. The method further includes generating a high-frequency reconstructed signal by copying a number of consecutive subband signals of the plurality of subband signals and obtaining an envelope adjusted high-frequency signal. The method further includes generating a noise component based on a noise parameter. Finally, the method includes adjusting a phase of the high-frequency reconstructed signal and obtaining a time-domain reconstructed audio signal by combining the decoded baseband audio signal and the combined high-frequency signal to obtain a time-domain reconstructed audio signal.