Abstract: A non-transitory computer-readable recording medium stores a program for causing a computer to execute a process, the process includes identifying first named entities and first verbs in a dependent relationship respectively with the first named entities from each of a plurality of sentences, vectorizing each of the plurality of sentences based on the first named entities and the first verbs, identifying, out of the plurality of sentences, one or more sentences each having a similarity greater than or equal to a threshold with respect to a specific sentence based on a plurality of vectors generated by the vectorizing, the similarity between two sentences being defined as a similarity between vectors generated by vectorizing the two sentences, and executing training of a machine learning model based on the one or more sentences.

Abstract: A sentence structure vectorization device includes processing circuitry to generate a plurality of morphemes by performing morphological analysis on an input sentence; to generate a dependence structure graph regarding the plurality of morphemes by performing dependency parsing on the plurality of morphemes; and to generate a sentence structure vector by extracting a plurality of pieces of partial structure information from the dependence structure graph and converting a morpheme string corresponding to the plurality of pieces of partial structure information into a numerical sequence.

Abstract: The quality of encoded signals can be improved by reclassifying AUDIO signals carrying non-speech data as VOICE signals when periodicity parameters of the signal satisfy one or more criteria. In some embodiments, only low or medium bit rate signals are considered for re-classification. The periodicity parameters can include any characteristic or set of characteristics indicative of periodicity. For example, the periodicity parameter may include pitch differences between subframes in the audio signal, a normalized pitch correlation for one or more subframes, an average normalized pitch correlation for the audio signal, or combinations thereof. Audio signals which are re-classified as VOICED signals may be encoded in the time-domain, while audio signals that remain classified as AUDIO signals may be encoded in the frequency-domain.

Abstract: A frequency domain long-term prediction system and method for estimating and applying an optimum long term predictor. Embodiments of the system and method include determining parameters of a single-tap predictor using a frequency-domain analysis having an optimality criteria based on spectral flatness measure. Embodiments of the system and method also include determining parameters of the long-term predictor by accounting for the performance of the vector quantizer in quantizing the various subbands. In some embodiments other encoder metrics (such as signal tonality) are used as well. Other embodiments of the system and method include determining the optimal parameters of the long-term predictor by accounting for some of the decoder operation. Other embodiments of the system and method include extending a 1-tap predictor to a k-th order predictor by convolving the 1-tap predictor with a pre-set filter and selecting from a table of such pre-set filters based on a minimum energy criteria.

Abstract: Techniques for content-based indexing of videos at web-scale are described. As one example, a computer-implemented method includes receiving a video file, splitting the video file into video frames and audio for the video frames, determining audial features for the audio, clustering each of a plurality of subsets of the audial features into a respective audio centroid for a shared set of bases, determining a first adjacency matrix of distances between the respective audio centroids, determining visual features for the video frames, clustering each of a plurality of subsets of the visual features into a respective video centroid, and determining a second adjacency matrix of distances between the respective video centroids.

Abstract: A system and method determining a match between sound fragments is provided. Each wave that makes up a sequence within the fragment is identified. An average amplitude and frequency of each wave is determined. An index of amplitudes and frequencies is determined by summating the square of the difference between the amplitude and frequencies, respectively, of each wave and the average amplitude and frequency, respectively, of the sequence. A single index is determined by averaging the index of amplitudes and frequencies. Matches between sound fragments may be determined by comparing the various indexes.

Abstract: Provided is pitch enhancement processing having little unnaturalness even in time segments for consonants, and having little unnaturalness to listeners caused by discontinuities even when time segments for consonants and other time segments switch frequently. A pitch emphasis apparatus obtains an output signal by executing pitch enhancement processing on each of time segments of a signal originating from an input audio signal.

Abstract: A method and device for allocating a bit-budget to a plurality of first parts and to a second part of a CELP core module of (a) an encoder for encoding a sound signal or (b) a decoder for decoding the sound signal. In a frame of the sound signal comprising sub-frames, respective bit-budgets are allocated to the first CELP core module parts and a bit-budget remaining after allocating to the first CELP core module parts their respective bit-budgets is allocated to the second CELP core module part. According to an alternative, the second CELP core module part bit-budget is distributed between the sub-frames of the frame and a larger bit-budget is allocated to at least one of the sub-frames of the frame. The at least one sub-frame may be the first sub-frame of the frame, at least one sub-frame following the first sub-frame, or the sub-frame using a glottal-impulse-shape codebook.

Abstract: Methods, systems and computer programs are presented for classifying malware using audio signal processing. One method includes an operation for converting a non-audio data file to an audio signal. Audio features are extracted from the audio signal and are used to classify the non-audio data file.

Abstract: There is provided a method and apparatus for motion estimation in a sequence of video images. The method comprises a) subdividing each field or frame of a sequence of video images into a plurality of blocks, b) assigning to each block in each video field or frame a respective set of candidate motion vectors, c) determining for each block in a current video field or frame, which of its respective candidate motion vectors produces a best match to a block in a previous video field or frame, d) forming a motion vector field for the current video field or frame using the thus determined best match vectors for each block, and e) forming a further motion vector field by storing a candidate motion vector derived from the best match vector at a block location offset by a distance derived from the candidate motion vector. Finally, steps a) to e) are repeated for a video field or frame following the current video field or frame.

Abstract: A low-complexity method and apparatus for detection of voiced speech and pitch estimation is disclosed that is capable of dealing with special constraints given by applications where low latency is required, such as in-car communication (ICC) systems. An example embodiment employs very short frames that may capture only a single excitation impulse of voiced speech in an audio signal. A distance between multiple such impulses, corresponding to a pitch period, may be determined by evaluating phase differences between low-resolution spectra of the very short frames. An example embodiment may perform pitch estimation directly in a frequency domain based on the phase differences and reduce computational complexity by obviating transformation to a time domain to perform the pitch estimation. In an event the phase differences are determined to be substantially linear, an example embodiment enhances voice quality of the voiced speech by applying speech enhancement to the audio signal.

Abstract: Voice-based authentication can include methods, systems, devices, and computer program products for providing user-specific services or access based at least in part on an utterance. In one method, an audio clip having an utterance is obtained. The utterance has an activation trigger portion and a command portion. A first distance between a vector representation of the activation trigger portion and a registered activation trigger vector is determined; and a second distance between a vector representation of the command portion and a registered command vector is determined. Responsive to the first distance satisfying a first distance threshold, and the second distance satisfying a second distance threshold, access is provided to a service associated with a registered user.

Abstract: A method of determining a distribution of bits for coding a transition frame, said method being implemented in a coder/decoder for coding/decoding a digital signal, the transition frame being preceded by a predictive coded preceding frame, coding the transition frame comprising transform coding and predictive coding a single sub-frame of the transition frame, the method comprising the following steps: assigning a bit rate for predictive coding the transition sub-frame, said bit rate being equal to the minimum between the bit rate for transform coding the transition frame and a first predetermined bit rate value; determining a first number of bits allocated for predictive coding the transition sub-frame for said bit rate; and calculating a second number of bits allocated for transform coding the transition frame from the first number of bits and a number of bits available for coding the transition frame.

Abstract: Embodiments of the present disclosure provide systems, methods, and computer storage media for mitigating latencies associated with the encoding of digital assets. Instead of waiting for codebook generation to complete in order to encode a digital asset for storage, embodiments described herein describe a shifting codebook generation and employment technique that significantly mitigates any latencies typically associated with encoding schemes. As a digital asset is received, a single codebook is trained based on each portion of the digital asset, or in some instances along with each portion of other digital assets being received. The single codebook is employed to encode subsequent portion(s) of the digital asset as it is received. The process continues until an end of the digital asset is reached or another command to terminate the encoding process is received. To encode an initial portion of the digital asset, a bootstrap codebook can be employed.

Abstract: Vehicle systems and methods are provided for capturing audio during operation for subsequent presentation and analysis. One method involves obtaining a plurality of audio segments via an audio input device onboard an aircraft, classifying each audio segment of the plurality of audio segments into one or more of a plurality of topic categories based at least in part on the content of the respective audio segment, and providing a graphical user interface (GUI) display depicting the plurality of audio segments in a time-ordered sequence. The GUI display includes GUI elements for selectively removing subsets of audio segments classified into particular topic categories from the time-ordered sequence.

Abstract: Vector Quantizer and method therein for vector quantization, e.g. in a transform audio codec. The method comprises comparing an input target vector with four centroids C0, C1, C0,flip and C1,flip, wherein centroid C0,flip is a flipped version of centroid C0 and centroid C1,flip is a flipped version of centroid C1, each centroid representing a respective class of codevectors. A starting point for a search related to the input target vector in the codebook is determined, based on the comparison. A search is performed in the codebook, starting at the determined starting point, and a codevector is identified to represent the input target vector. A number of input target vectors per block or time segment is variable. A search space is dynamically adjusted to the number of input target vectors. The codevectors are sorted according to a distortion measure reflecting the distance between each codevector and the centroids C0 and C1.

Abstract: The present disclosure relates to a 5G or pre-5G communication system for supporting a higher data transmission rate than in a 4G communication system such as LTE. The present disclosure relates to uplink transmission in a wireless communication system, and an operating method of a terminal includes mapping codes that are included in at least one codebook onto data symbols. and transmitting the data symbols spread by using the at least one codebook, and the data symbols are used for a base station to detect at least one active terminal including the terminal.

Abstract: Embodiments of the present invention may provide the capability to detect complex events while providing improved detection and performance. In an embodiment of the present invention, a method for detecting an event may comprise receiving data representing measurement or detection of physical parameters, conditions, or actions, quantizing the received data and selecting a number of samples from the quantized data, generating a hidden Markov model representing events to be detected using initial model values based on ideal conditions, wherein a desired output is defined as a sequence of states, and wherein a number of states of the hidden Markov model is less than or equal to the number of samples of the quantized data, adjusting the quantized data and the initial model values to improve accuracy of the model, determining a state sequence of the hidden Markov model, and outputting an indication of a detected event.

Abstract: Disclosed are a system and a method for singing expression transplantation. A singing expression transplantation method performed by a singing expression transplantation system according to an embodiment may comprise the steps of: synchronizing each of a first sound source and a second sound source, which include different pieces of voice information with regard to an identical song; modifying the pitch of the first sound source on the basis of pitch information extracted from each of the first sound source and the second sound source, which have been synchronized; and extracting volume information from each of the first sound source and the second sound source and adjusting the magnitude of the volume regarding the first sound source, the pitch of which has been modified, according to each piece of extracted volume information.

Abstract: An apparatus includes a speech processing engine configured to receive data corresponding to speech and to determine whether a first characteristic associated with the speech differs from a reference characteristic by at least a threshold amount. The apparatus further includes a selection circuit responsive to the speech processing engine. The selection circuit is configured to select a particular speech codebook from among a plurality of speech codebooks based on the first characteristic differing from the reference characteristic by at least the threshold amount. The particular speech codebook is associated with the first characteristic.

Abstract: Secure analytics using homomorphic and injective format-preserving encryption are disclosed herein. An example method includes encoding an analytic parameter set using a homomorphic encryption scheme as a set of homomorphic analytic vectors; transmitting the set of homomorphic analytic vectors to a server system; and receiving a homomorphic encrypted result from the server system, the server system having utilized the homomorphic encryption scheme and a first injective, format-preserving encryption scheme to evaluate the set of homomorphic analytic vectors over a datasource.

Abstract: A computer-implemented method, according to one embodiment, includes: receiving a complex audio signal which includes an intended audio signal and at least one interfering audio signal. The complex audio signal is converted into text which represents a plurality of words included in the complex audio signal, and at least some of the text is identified as representing words which correspond to the at least one interfering audio signal. The identified text is discarded, and a remaining portion of the text is evaluated to determine whether the remaining portion of the text represents words which convey the voice-based command at an accuracy that is in a predetermined range. Furthermore, the remaining portion of the text is output in response to determining that the remaining portion of the text represents words which convey the voice-based command at an accuracy that is in the predetermined range.

Abstract: A method for utterance section detection includes: executing pitch gain calculation processing that includes calculating a pitch gain indicating an intensity of periodicity of an audio signal expressing a voice of a speaker for each of frames that are obtained by dividing the audio signal and that each have a predetermined length; and executing utterance section detection processing that includes determining that an utterance section on the audio signal starts when the pitch gain becomes greater than or equal to a first threshold value after a non-utterance section on the audio signal lasts, wherein the utterance section detection processing further includes determining that the utterance section ends when the pitch gain becomes less than a second threshold value lower than the first threshold value after the utterance section lasts.

Abstract: A method of authenticating a user based on voice recognition of a keyword includes generating, at a processor, clean speech statistics. The clean speech statistics are generated from an audio recording of the keyword spoken by the user during an enrollment phase. The method further includes separating speech data and noise data from noisy input speech using the clean speech statistics during an authentication phase. The method also includes authenticating the user by comparing the speech data to the clean speech statistics or by comparing the noisy input speech to noisy speech statistics. The noisy speech statistics are based at least in part on the noise data.

Abstract: A speech recognition system includes an input device to receive voice sounds, one or more processors, and one or more storage devices storing parameters and program modules including instructions which cause the one or more processors to perform operations.

Abstract: A speech-enhancing noise filter is disclosed. The noise filter comprises a microphone for acquiring speech data from a user; a feature extraction module configured to extract a plurality of features characterizing the speech data; a neural network configured to receive the plurality of extracted features and to estimate a noise profile from the plurality of extracted features; a noise removal module configured to remove the noise profile from the noisy speech data; and a reconstruction module configured to generate a waveform from the plurality of frames after removal of the noise profile from each of those frames. The neural network is trained to isolate various types of noise from the user speech in the speech data and then subtract the noise from the speech data, thus leaving only the user speech free of noise.

Abstract: A method comprising: receiving lattice vector quantised parameter data, the parameter data representing at least one audio signal; determining within the data at least one bit error; and controlling the decoding of the data to generate an audio signal based on the determining of the bit error.

Abstract: Vector Quantizer and method therein for vector quantization, e.g. in a transform audio codec. The method comprises comparing an input target vector with four centroids C0, C1, C0,flip and C1,flip, wherein centroid C0,flip is a flipped version of centroid C0 and centroid C1,flip is a flipped version of centroid C1, each centroid representing a respective class of codevectors. A starting point for a search related to the input target vector in the codebook is determined, based on the comparison. A search is performed in the codebook, starting at the determined starting point, and a codevector is identified to represent the input target vector. A number of input target vectors per block or time segment is variable. A search space is dynamically adjusted to the number of input target vectors. The codevectors are sorted according to a distortion measure reflecting the distance between each codevector and the centroids C0 and C1.

Abstract: Disclosed is an apparatus for processing an input signal, having a perceptual weighter and a quantizer. The perceptual weighter has a model provider and a model applicator. The model provider provides a perceptual weighted model based on the input signal. The model applicator provides a perceptually weighted spectrum by applying the perceptual weighted model to a spectrum based on the input signal. The quantizer is configured to quantize the perceptually weighted spectrum and for providing a bitstream. The quantizer has a random matrix applicator and a sign function calculator. The random matrix applicator is configured for applying a random matrix to the perceptually weighted spectrum in order to provide a transformed spectrum. The sign function calculator is configured for calculating a sign function of components of the transformed spectrum in order to provide the bitstream. The invention further refers to an apparatus for processing an encoded signal and to corresponding methods.

Abstract: An apparatus for speech processing: calculates a pitch gain indicating a magnitude of periodicity of an audio signal for each frame, the audio signal representing speaker's voice to be divided into the frames each having a predetermined length; determines that a speech production interval has started, when the pitch gain becomes equal or greater than a first threshold after a non-speech production interval; sets a second threshold that is lower than the first threshold by a reduction amount corresponding to a value acquired by subtracting a second representative value of the pitch gain in an interval preceding the start of the speech production interval from a first representative value of the pitch gain in the speech production interval; and determines that the speech production interval has terminated, when the pitch gain becomes smaller than the second threshold after the speech production interval has started.

Abstract: Systems, controllers and methods for contextual-based searching are provided. A system includes one or more devices configured to collect at least one audio stream and a contextual search system. The contextual search system includes a data analyzer and a search engine. The data analyzer is configured to receive the at least one audio stream from among the one or more devices and to determine contextual information from the received at least one audio stream. The search engine is configured to perform a search of at least one search term using the contextual information, to produce a search result.

Abstract: An information encoder for encoding an information signal includes: a converter for converting the linear prediction coefficients of the predictive polynomial A(z) to frequency values f1 . . . fn of a spectral frequency representation of the predictive polynomial A(z), wherein the converter is configured to determine the frequency values f1 . . . fn by analyzing a pair of polynomials P(z) and Q(z) being defined as P(z)=A(z)+z?m?lA(z?1) and Q(z)=A(z)?z?m?lA(z?1), wherein m is an order of the predictive polynomial A(z) and l is greater or equal to zero, wherein the converter is configured to obtain the frequency values by establishing a strictly real spectrum derived from P(z) and a strictly imaginary spectrum from Q(z) and by identifying zeros of the strictly real spectrum derived from P(z) and the strictly imaginary spectrum derived from Q(z).

Abstract: A system and method are provided for analyzing a video. The method comprises: sampling the video to generate a plurality of spatio-temporal video volumes; clustering similar ones of the plurality of spatio-temporal video volumes to generate a low-level codebook of video volumes; analyzing the low-level codebook of video volumes to generate a plurality of ensembles of volumes surrounding pixels in the video; and clustering the plurality of ensembles of volumes by determining similarities between the ensembles of volumes, to generate at least one high-level codebook. Multiple high-level codebooks can be generated by repeating steps of the method. The method can further include performing visual event retrieval by using the at least one high-level codebook to make an inference from the video, for example comparing the video to a dataset and retrieving at least one similar video, activity and event labeling, and performing abnormal and normal event detection.

Abstract: A method of transmission over multiple wireless channels in a multiple antenna system includes storing channel modulation matrices at a transmitter; receiving quantized channel state information at the transmitter from plural receivers; selecting a transmission modulation matrix using the quantized channel state information from the stored channel modulation matrices; and transmitting over the multiple channels to the plural receivers using the selected transmission modulation matrix.

Abstract: A method of enhancing speech quality includes: generating a high-frequency signal by using a low-frequency signal in a time domain; combining the low-frequency signal with the high-frequency signal; transforming the combined signal into a spectrum in a frequency domain; determining a class of a decoded speech signal; predicting an envelope from a low-frequency spectrum obtained in the transforming; and generating a final high-frequency spectrum by applying the predicted envelope to a high-frequency spectrum obtained in the transforming.

Abstract: A speech encoder that analyzes and classifies each frame of speech as being periodic-like speech or non-periodic like speech where the speech encoder performs a different gain quantization process depending if the speech is periodic or not. If the speech is periodic, the improved speech encoder obtains the pitch gains from the unquantized weighted speech signal and performs a pre-vector quantization of the adaptive codebook gain GP for each subframe of the frame before subframe processing begins and a closed-loop delayed decision vector quantization of the fixed codebook gain GC. If the frame of speech is non-periodic, the speech encoder may use any known method of gain quantization.

Abstract: A method is provided for processing attenuation of pre-echo in a digital audio signal decoded by transform decoding. The method includes the following acts: decomposition of the decoded signal into at least two sub-signals according to a pre-determined decomposition criterion; calculation of attenuation factors per sub-signal and per sample of a previously determined pre-echo zone; attenuation of pre-echo in the pre-echo zone of each of the sub-signals by applying attenuation factors to the sub-signals; and production of the attenuated signal by addition of the attenuated sub-signals. Also provided are a processing device implementing the acts of the described method, and a decoder including such a device.

Abstract: A method of wirelessly communicating a packet can include generating, at a wireless device, a packet including a plurality of symbols. The method further includes segmenting an input bit vector into a plurality of symbol vectors according to one of a sequential or distributed segmentation procedure. The method further includes splitting each of the plurality of symbol vectors into two or more split vectors according to one of a sequential or round-robin split procedure. The method further includes mapping each of the split vectors into the plurality of symbols according to one of a block-level repetition or a symbol-level repetition. The method further includes transmitting the packet.

Abstract: A spectrum filler for filling non-coded residual sub-vectors of a transform coded audio signal includes a sub-vector compressor configured to compress actually coded residual sub-vectors. A sub-vector rejecter is configured to reject compressed residual sub-vectors that do not fulfill a predetermined sparseness criterion. A sub-vector collector is configured to concatenate the remaining compressed residual sub-vectors to form a first virtual codebook. A coefficient combiner is configured to combine pairs of coefficients of the first virtual codebook to form a second virtual codebook. A sub-vector filler is configured to fill non-coded residual sub-vectors below a predetermined frequency with coefficients from the first virtual codebook, and to fill non-coded residual sub-vectors above the predetermined frequency with coefficients from the second virtual codebook.

Abstract: Provided is a method for user training of an information dialog system. The method may include activating a user input subsystem, receiving a training request entered by the user, converting the training request into text by the user input subsystem, sending the text of the training request obtained as a result of the conversion to a dialog module, processing the text of the training request by the dialog module, forming a response to the training request by the dialog module, and sending the response to the training request to the user. The response to the training request may be formed in a form of one or more of the following: a voice cue, a text, and an action performed by the information dialog system.

Abstract: In an embodiment, a system on a chip (SOC) may include one or more central processing units (CPUs), a memory controller, and a circuit configured to remain powered on when the rest of the SOC is powered down. The circuit may be configured to receive audio samples and match those audio samples against a predetermined pattern. The circuit may operate according to a first clock during the time that the rest of the SOC is powered down. In response to detecting the predetermined pattern in the samples, the circuit may cause the memory controller and processors to power up. During the power up process, a second clock having one or more better characteristics than the first clock may become available. The circuit may switch to the second clock while preserving the samples, or losing at most one sample, or no more than a threshold number of samples.

Abstract: An apparatus comprising: a vector generator configured to generate at least one vector of parameters defining at least one audio signal; a lattice vector quantizer configured to sort the at least one vector of parameters according to an ordering of at least one vector absolute tuples to generate an associated at least one ordered vector of parameters; the lattice vector quantizer configured to select from a list of leader classes at least one potential code vector; the lattice vector quantizer configured to determine a distance between the at least one potential code vector and the at least one ordered vector of parameters; the lattice vector quantizer configured to determine at least one leader class associated with a potential code vector which generates the smallest associated distance; the lattice vector quantizer configured to transpose the at least one leader class to generate an output lattice quantized codevector.

Abstract: Provided are, among other things, systems, methods and techniques for compressing an audio signal. According to one representative embodiment, an audio signal that includes quantization indexes, identification of segments of such quantization indexes, and indexes of entropy codebooks that have been assigned to such segments is obtained, with a single entropy codebook index having been assigned to each such segment. Potential merging operations in which adjacent ones of the segments potentially would be merged with each are identified, and bit penalties for the potential merging operations are estimated. At least one of the potential merging operations is performed based on the estimated bit penalties, thereby obtaining a smaller updated set of segments of quantization indexes and corresponding assigned codebooks. The quantization indexes in each of the segments in the smaller updated set are then entropy encoded by using the corresponding assigned entropy codebooks, thereby compressing the audio signal.

Abstract: A noise attenuation apparatus receives an audio signal comprising a desired and a noise signal component. Two codebooks (109, 111) comprise respectively desired signal candidates representing a possible desired signal component and noise signal contribution candidates representing possible noise contributions. A segmenter (103) segments the audio signal into time segments and for each time segment a noise attenuator (105) generates estimated signal candidates by for each of the desired signal candidates generating an estimated signal candidate as a combination of a scaled version of the desired signal candidate and a weighted combination of the noise signal contribution candidates. The noise attenuator (105) minimizes a cost function indicative of a difference between the estimated signal candidate and the audio signal in the time segment. A signal candidate is then determined for the time segment from the estimated signal candidates and the audio signal is noise compensated based on this signal candidate.

Abstract: A method, device, and apparatus provide the ability to predict a portion of a polyphonic audio signal for compression and networking applications. The solution involves a framework of a cascade of long term prediction filters, which by design is tailored to account for all periodic components present in a polyphonic signal. This framework is complemented with a design method to optimize the system parameters. Specialization may include specific techniques for coding and networking scenarios, where the potential of each enhanced prediction is realized to considerably improve the overall system performance for that application. One specific technique provides enhanced inter-frame prediction for the compression of polyphonic audio signals, particularly at low delay. Another specific technique provides improved frame loss concealment capabilities to combat packet loss in audio communications.

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

Abstract: An audio signal decoding apparatus is provided that includes a receiver that receives an encoded information, a memory, and a processor that demultiplexes low-band encoding parameters, index information, and scale factor information from the encoded information. The processor also decodes the low-band encoding parameters to obtain a synthesized low frequency spectrum, replicates a high frequency subband spectrum based on the index information using the synthesized low frequency spectrum, and adjusts an amplitude of the replicated high frequency subband spectrum using the scale factor information. The processor further estimates a frequency of a harmonic component in the synthesized low frequency spectrum, adjusts a frequency of a harmonic component in the high frequency subband spectrum using the estimated harmonic frequency spectrum, and generates an output signal using the synthesized low frequency spectrum and the high frequency subband spectrum.

Abstract: An audio encoding apparatus to encode an audio signal using lossless coding or lossy coding and an audio decoding apparatus to decode an encoded audio signal are disclosed. An audio encoding apparatus according to an exemplary embodiment may include an input signal type determination unit to determine a type of an input signal based on characteristics of the input signal, a residual signal generation unit to generate a residual signal based on an output signal from the input signal type determination unit, and a coding unit to perform lossless coding or lossy coding using the residual signal.

Abstract: A method for measuring speech signal quality by an electronic device is described. The method includes obtaining a modified single-channel speech signal. The method also includes estimating multiple objective distortions based on the modified single-channel speech signal. The multiple objective distortions include at least one foreground distortion and at least one background distortion. The method further includes estimating a foreground quality and a background quality based on the multiple objective distortions. The method additionally includes estimating an overall quality based on the foreground quality and the background quality.

Abstract: In an embodiment, a system on a chip (SOC) may include one or more central processing units (CPUs), a memory controller, and a circuit configured to remain powered on when the rest of the SOC is powered down. The circuit may be configured to receive audio samples and match those audio samples against a predetermined pattern. The circuit may operate according to a first clock during the time that the rest of the SOC is powered down. In response to detecting the predetermined pattern in the samples, the circuit may cause the memory controller and processors to power up. During the power up process, a second clock having one or more better characteristics than the first clock may become available. The circuit may switch to the second clock while preserving the samples, or losing at most one sample, or no more than a threshold number of samples.