Abstract: A computer-implemented method for determining a relevance of a node in a network. A digital representation of a local neighborhood structure of the node in the network is obtained in a computer-readable non-volatile memory. A numerical value characteristic of the node's relevance is determined, and output to a user. The numerical value is determined based on the neighborhood structure of the node.

Abstract: Proposed is a method and apparatus for determining a weighting function for quantizing a linear predictive coding (LPC) coefficient and having a low complexity. The weighting function determination apparatus may convert an LPC coefficient of a mid-subframe of an input signal to one of a immittance spectral frequency (ISF) coefficient and a line spectral frequency (LSF) coefficient, and may determine a weighting function associated with an importance of the ISF coefficient or the LSF coefficient based on the converted ISF coefficient or LSF coefficient.

Abstract: A method includes decoding a low-band mid channel bitstream to generate a low-band mid signal and a low-band mid excitation signal. The method further includes decoding a high-band mid channel bandwidth extension bitstream to generate a synthesized high-band mid signal. The method also includes determining an inter-channel bandwidth extension (ICBWE) gain mapping parameter corresponding to the synthesized high-band mid signal. The ICBWE gain mapping parameter is based on a selected frequency-domain gain parameter that is extracted from a stereo downmix/upmix parameter bitstream. The method further includes performing a gain scaling operation on the synthesized high-band mid signal based on the ICBWE gain mapping parameter to generate a reference high-band channel and a target high-band channel. The method includes outputting a first audio channel and a second audio channel. The first audio channel is based on the reference high-band channel, and the second audio channel is based on target high-band channel.

Abstract: Disclosed are some examples of systems, apparatus, methods and computer program products implementing techniques for extending the range of a set of decoded parameter values for a sequence of frequency bands in an identifiable time frame of an audio signal. In some implementations, the parameter values vary in relation to a sequence of time frames of the audio signal and in relation to a sequence of frequency bands in each time frame. In some implementations, it is determined that a decoded value corresponds to a minimum of a first range of values of a first coding protocol of a set of coding protocols. The determined value is modified to be below the minimum of the first range of values to produce an extended value. A modified set of decoded values including one or more extended values can thus be provided.

Abstract: Systems and methods for the matching of datasets, such as input audio segments, with known datasets in a database are disclosed. In an illustrative embodiment, the use of the presently disclosed systems and methods is described in conjunction with recognizing known network message recordings encountered during an outbound telephone call. The methodologies include creation of a ternary fingerprint bitmap to make the comparison process more efficient. Also disclosed are automated methodologies for creating the database of known datasets from a larger collection of datasets.

Abstract: A method includes selecting, at an encoder of a first device, a left channel or a right channel as a non-reference target channel based on a high-band reference channel indicator. The method also includes generating a synthesized non-reference high-band channel based on a non-reference high-band excitation corresponding to the non-reference target channel. The method also includes generating a high-band portion of the non-reference target channel. The method further includes estimating one or more spectral mapping parameters based on the synthesized non-reference high-band channel and the high-band portion of the non-reference target channel. The method also includes applying the one or more spectral mapping parameters to the synthesized non-reference high-band channel to generate a spectrally shaped synthesized non-reference high-band channel.

Abstract: The present disclosure relates methods and apparatus for recovering a lost frame in a received audio signal. One example method includes obtaining an initial high-frequency band signal of a current lost frame in the received audio signal, calculating a ratio R, where the ratio R is a ratio of a high frequency excitation energy of a previous frame of the current lost frame to a high frequency excitation energy of the current lost frame, obtaining a global gain of the current lost frame according to the ratio R and a global gain of the previous frame of the current lost frame, and recovering a high-frequency band signal of the current lost frame according to the initial high-frequency band signal of the current lost frame and the global gain of the current lost frame.

Abstract: A calculator for determining phase correction data for an audio signal includes a variation determiner for determining a variation of a phase of the audio signal in a first and a second variation mode, a variation comparator for comparing a first variation determined using the first variation mode and a second variation determined using the second variation mode, and a correction data calculator for calculating the phase correction data in accordance with the first variation mode or the second variation mode based on a result of the comparing.

Abstract: A hearing aid includes a microphone for converting sound into an audio input signal, a hearing loss processor, an adaptive feedback suppressor configured for generation of a feedback suppression signal by modelling a feedback signal path of the hearing aid, a subtractor coupled to an output of the adaptive feedback suppressor, wherein the subtractor is configured for subtracting the feedback suppression signal from the audio input signal, and outputting a feedback compensated audio signal to the hearing loss processor, and wherein the hearing loss processor is configured for processing the feedback compensated audio signal in accordance with a hearing loss of a user of the hearing aid, a synthesizer configured for generation of a synthesized signal based at least on a sound model and the audio input signal, and a receiver for generating an output sound signal based at least on the synthesized signal.

Abstract: An audio signal decoding apparatus is provided that includes a receiver that receives an encoded information, a memory, and a processor that demultiplexes the encoded information, including encoding parameters that are used for decoding a low frequency spectrum and index information that identifies a most, correlated portion from a low frequency spectrum for one or more high frequency subbands. The processor also replicates a high frequency subband spectrum based on the index information using a synthesized low frequency spectrum, the synthesized low frequency spectrum being obtained by decoding the encoding parameters. 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, and generates an output signal using the synthesized low frequency spectrum and the high frequency subband spectrum.

Abstract: A method of encoding audio information for forward error correction reconstruction of a transmitted audio stream over a lossy packet switched network, the method including the steps of: (a) dividing the audio stream into audio frames; (b) determining a series of corresponding audio frequency bands for the audio frames; (c) determining a series of power envelopes for the frequency bands; (d) encoding the envelopes as a low bit rate version of the audio frame in a redundant transmission frame.

Abstract: A method includes decoding a low-band portion of an encoded mid channel to generate a decoded low-band mid channel. The method also includes filtering the decoded low-band mid channel according to one or more filter coefficients to generate a low-band filtered mid channel. The method also includes generating an inter-channel predicted signal based on the low-band filtered mid channel and the inter-channel prediction gain. The method further includes generating a low-band left channel and a low-band right channel based on an up-mix factor, the decoded low-band mid channel, and the inter-channel predicted signal.

Abstract: A method to detect a packet includes: receiving an input sequence including preambles; detecting a transition from a noise period to a signal period in the input sequence; dynamically adjusting a gain of the input sequence in response to the signal period being initiated; and distinguishing an intended packet from other packets, among packets received in the preambles.

Abstract: A voice activity detection (VAD) system includes an input processing module configured to receive an acoustic signal, convert the acoustic signal into an analog signal, and subsequently, a digital signal; an energy-based detection module configured to receive one of the analog/digital signals and determine a sound activity decision; an area-function-based detection module configured to derive an area-related function from the digital signal and use a machine learning method to output an area-based decision according to the area related function; and a VAD decision module configured to make a final VAD decision based on the sound activity decision from the energy-based detection module and the area-based decision from the area-function-based detection module.

Abstract: A terminal apparatus is provided. The terminal apparatus includes an input unit configured to receive an input of a search word related to a content to be searched for, a search unit including a processor configured to search for a time range corresponding to the content using the input search word and pre-stored log information, and to search for log information within the searched time range, and a display unit configured to display information regarding a list of contents corresponding to the searched log information.

Abstract: A method and apparatus for performing coding and decoding for high-frequency bandwidth extension. The coding apparatus may classify a coding mode of a low-frequency signal of an input signal based on characteristics of the low-frequency signal of an input signal, perform code excited linear prediction coding or audio coding on the LPC excitation signal of the low-frequency signal of an input signal, and perform time-domain (TD) extension coding or frequency-domain (FD) extension coding on a high-frequency signal of an input signal. When the FD extension coding is performed, the coding apparatus may generate a base excitation signal for a high band using an input spectrum, obtain an energy control factor of a sub-band in a frame using the base excitation signal and the input spectrum, generate an energy signal based on the input spectrum and the energy control factor, for the sub-band in the frame, and quantize the energy signal.

Abstract: In a method for coding of information for enhancing a background noise representation, voice activity of an input speech signal is determined. A noisiness parameter is determined for an inactive speech signal, wherein the noisiness parameter is based on a ratio of prediction gains of two Linear Predictive Coder (LPC) prediction filters with different orders. The noisiness parameter is quantized, and the quantized noisiness parameter is encoded for transmission.

Abstract: A coding method and a decoding method are provided which can use in combination a predictive coding and decoding method which is a coding and decoding method that can accurately express coefficients which are convertible into linear prediction coefficients with a small code amount and a coding and decoding method that can obtain correctly, by decoding, coefficients which are convertible into linear prediction coefficients of the present frame if a linear prediction coefficient code of the present frame is correctly input to a decoding device.

Abstract: An apparatus and method for encoding and decoding a signal for high frequency bandwidth extension are provided. An encoding apparatus may down-sample a time domain input signal, may core-encode the down-sampled time domain input signal, may transform the core-encoded time domain input signal to a frequency domain input signal, and may perform bandwidth extension encoding using a basic signal of the frequency domain input signal.

Abstract: A method for determining a noise shaped quantized parameter contributing to generation of an output signal comprises estimating an error within the output signal using a quantization of the parameter and a quantization of a further parameter contributing to generation of the output signal. The quantization of the parameter is used as the noise shaped quantized parameter according to a selection criterion.

Abstract: Example embodiments disclosed herein relate to separated audio analysis and processing. A system for processing an audio signal is disclosed. The system includes an audio analysis module configured to analyze an input audio signal to determine a processing parameter for the input audio signal, the input audio signal being represented in time domain. The system also includes an audio processing module configured to process the input audio signal in parallel with the audio analysis module. The audio processing module includes a time domain filter configured to filter the input audio signal to obtain an output audio signal in the time domain, and a filter controller configured to control a filter coefficient of the time domain filter based on the processing parameter determined by the audio analysis module. Corresponding method and computer program product of processing an audio signal are also disclosed.

Abstract: A speech recognition method acquires sound information via multiple microphones, detects a sound source interval including sound from the sound information, acquires an estimated direction of speech by conducting direction estimation on a speech interval from among the sound source interval, conducts an adaptation process of using the sound information to estimate filter coefficients, decides a buffer size of the sound information to hold in a buffer, based on sound source interval information, estimated direction information, and adaptation process convergence state information, holds the sound information in the buffer according to the buffer size, conducts a beamforming process using the sound information held in the buffer and the filter coefficients to acquire speech information, and conducts speech recognition on the speech information acquired by the beamforming process.

Abstract: A spectrum encoding method includes selecting an important spectral component in band units for a normalized spectrum and encoding information of the selected important spectral component for a band, based on a number, a position, a magnitude and a sign thereof. A spectrum decoding method includes obtaining from a bitstream, information about an important spectral component for a band of an encoded spectrum and decoding the obtained information of the important spectral component, based on a number, a position, a magnitude and a sign of the important spectral component.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation using a time domain excitation signal.

Abstract: Provided is a method and apparatus for determining a signal coding mode. The signal coding mode may be determined or changed according to whether a current frame corresponds to a silence period and by using a history of speech or music presence possibilities.

Abstract: The present disclosure provides a method and a device for processing a speech based on artificial intelligence. The method includes: receiving a speech processing request, in which the speech processing request includes a first digital speech signal and a first sample frequency corresponding to the first digital speech signal; selecting a target speech processing model from a pre-trained speech processing model base according to the first sample frequency; performing up-sampling processing on the first digital speech signal using the target speech processing model to generate a second digital speech signal having a second sample frequency, in which the second sample frequency is larger than the first sample frequency.

Abstract: An encoder for encoding an audio signal includes an analyzer for analyzing the audio signal and for determining analysis prediction coefficients from the audio signal. The encoder includes a converter for deriving converted prediction coefficients from the analysis prediction coefficients, a memory for storing a multitude of correction values and a calculator. The calculator includes a processor for processing the converted prediction coefficients to obtain spectral weighting factors. The calculator includes a combiner for combining the spectral weighting factors and the multitude of correction values to obtain corrected weighting factors. A quantizer of the calculator is configured for quantizing the converted prediction coefficients using the corrected weighting factors to obtain a quantized representation of the converted prediction coefficients.

Abstract: A speech signal decoding device includes an adaptive code-book decoder, a fixed code-book decoder, a ratio calculator, a determination unit, and an attenuator. The adaptive code-book decoder generates an adaptive code-book vector using an adaptive code-book code of an encoded signal encoded by a CELP method. The fixed code-book decoder generates a fixed code-book vector using a fixed code-book code of the encoded signal. The ratio calculator calculates an amplitude ratio or an energy ratio between the adaptive code-book vector and the fixed code-book vector. The determination unit determines whether the amplitude ratio or the energy ratio exceeds a predetermined threshold. When the amplitude ratio or the energy ratio is determined to exceed the predetermined threshold, the attenuator attenuates an excitation signal that has been defined by adding the adaptive code-book vector to the fixed code-book vector.

Abstract: A system and method for determining users personality types, profiles, and infer the likely intent based on the users' musical taste and listening habits. The process involves identifying the listener's musical preferences and linking this to a predetermined scale within the psychological Big 5 that in turn links to a dictionary of descriptors that defines the personality type. This can be further fine tuned by knowing the persons age, gender, and many other lifestyle variables, as well as how a user may use the music.

Abstract: In accordance with an embodiment, a method of decoding an encoded audio bitstream at a decoder includes receiving the audio bitstream, decoding a low band bitstream of the audio bitstream to get low band coefficients in a frequency domain, and copying a plurality of the low band coefficients to a high frequency band location to generate high band coefficients. The method further includes processing the high band coefficients to form processed high band coefficients. Processing includes modifying an energy envelope of the high band coefficients by multiplying modification gains to flatten or smooth the high band coefficients, and applying a received spectral envelope decoded from the received audio bitstream to the high band coefficients. The low band coefficients and the processed high band coefficients are then inverse-transformed to the time domain to obtain a time domain output signal.

Abstract: A method and an apparatus for predicting a high band excitation signal are disclosed. The method includes: acquiring, according to a received low band bitstream, a set of spectral frequency parameters that are arranged in an order of frequencies, calculating a spectral frequency parameter difference between every two spectral frequency parameters that have a same position interval; acquiring a minimum spectral frequency parameter difference from the calculated spectral frequency parameter differences; determining, according to a frequency bin that corresponds to the minimum spectral frequency parameter difference, a start frequency bin for predicting a high band excitation signal from a low band; and predicting the high band excitation signal from the low band according to the start frequency bin. By implementing embodiments of the present invention, a high band excitation signal can be better predicted, thereby improving performance of the high band excitation signal.

Abstract: An apparatus for decoding an encoded audio signal, includes: a spectral domain audio decoder for generating a first decoded representation of a first set of first spectral portions being spectral prediction residual values; a frequency regenerator for generating a reconstructed second spectral portion using a first spectral portion of the first set of first spectral portions, wherein the reconstructed second spectral portion additionally includes spectral prediction residual values; and an inverse prediction filter for performing an inverse prediction over frequency using the spectral residual values for the first set of first spectral portions and the reconstructed second spectral portion using prediction filter information included in the encoded audio signal.

Abstract: A linear predictive coding apparatus is provided that performs linear predictive analysis using a pseudo correlation function signal sequence obtained by performing inverse Fourier transform regarding the ?1-th power of the absolute values of the frequency domain sample sequence corresponding to the time-series signal as a power spectrum to obtain coefficients transformable to linear predictive coefficients. The apparatus further adapts values of ? for a plurality of candidates for coefficients transformable to linear predictive coefficients stored in a code book and the coefficients transformable to linear predictive coefficients are obtained by the linear predictive analysis.

Abstract: A method for recovering a lost frame in a received audio signal includes: obtaining an initial high-frequency band signal of a current lost frame in the received audio signal; calculating a ratio R, wherein the ratio R is a ratio of a high frequency excitation energy of a previous frame of the current lost frame to a high frequency excitation energy of the current lost frame; obtaining a global gain of the current lost frame according to the ratio R and a global gain of the previous frame of the current lost frame; and recovering a high-frequency band signal of the current lost frame according to the initial high-frequency band signal of the current lost frame and the global gain of the current lost frame. The method can be used in an audio signal decoding process for low-loss recovery of lost frames of the audio signal.

Abstract: A coding technology that efficiently codes an input sound signal irrespective of the characteristics thereof and can obtain a decoded sound signal that sounds less artificial to a listener. A coding method codes an input sound signal frame by frame of a predetermined time segment by a selected coding processing from a plurality of types of coding processing in the frequency domain, the coding method makes it possible for a selection unit to select coding processing which is different from the coding processing of the preceding frame as coding processing of the present frame if at least one of the magnitude of the energy of high frequency components of the input sound signal of the preceding frame and the magnitude of the energy of high frequency components of the input sound signal of the present frame is smaller than or equal to a predetermined threshold value.

Abstract: The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular, a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described. The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of ?f.

Abstract: Embodiments disclosed herein provide a method for detecting an audio signal and an apparatus, where the method includes determining an input audio signal as a to-be-determined audio signal; determining an enhanced segmental signal-to-noise ratio (SSNR) of the audio signal, where the enhanced SSNR is greater than a reference SSNR; and comparing the enhanced SSNR with a voice activity detection (VAD) decision threshold to determine whether the audio signal is an active signal. According to the method and the apparatus provided in the embodiments, an active voice and an inactive voice can be accurately distinguished.

Abstract: A processor configured to: receive, at a floating-point-input-terminal, an input-block of data comprising a plurality of floating-point numbers each floating-point number comprising a mantissa and an exponent; determine an input-scale-factor based on a previous-input-block-exponent-value associated with a previous-input-block of data; and convert the input-block of data into a fixed-point-block of data in accordance with the input-scale-factor, wherein the fixed-point-block of data comprises a plurality of fixed-point-values that can represent the plurality of floating-point numbers within a particular range.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information. The audio decoder has an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame, wherein the error concealment is configured to modify a time domain excitation signal obtained for one or more audio frames preceding a lost audio frame, in order to obtain the error concealment audio information.

Abstract: An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme.

Abstract: Information identifying a user as having accessed a first network device from an external network connected device is received at an intermediate network connected device from an edge network device. The intermediate network connected device is arranged between the first network device and a second network device. The intermediate network connected device, the first network device and the second network device are within the network and the external network connected device is outside the network. A request to access the second network device is received at the intermediate network connected device from the first network device. It is determined that the user is a source of the request via the first network device based upon the received information. The request from the first network device is evaluated based upon determining the user is the source of the request.

Abstract: A weighting function determination method includes obtaining a line spectral frequency (LSF) coefficient or an immitance spectral frequency (ISF) coefficient from a linear predictive coding (LPC) coefficient of an input signal and determining a weighting function by combining a first weighting function based on spectral analysis information and a second weighting function based on position information of the LSF coefficient or the ISF coefficient.

Abstract: An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.

Abstract: Information related to voice of a question and information related to voice of a response to the question are received. An analysis section acquires a representative pitch of the question (e.g., a pitch of the end of the question), and a representative pitch of the response (e.g., an average pitch of the response) based on the received information. On the basis of comparison between the representative pitch of the question and the representative pitch of the response, an evaluation section evaluates the voice of the response to the question on the basis of how much a difference between the respective representative pitches of the question and the response is away from a predetermined reference value (e.g., a fifth consonant interval). Further, a conversation interval detection section is provided for detecting a conversation interval, i.e., a time interval from the end of the question to the start of the response.

Abstract: A quantizing method is provided that includes quantizing an input signal by selecting one of a first quantization scheme not using an inter-frame prediction and a second quantization scheme using the inter-frame prediction, in consideration of one or more of a prediction mode, a predictive error and a transmission channel state.

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

Abstract: A method for using voiceprint identification to operate voice recognition and electronic device thereof are provided. The method includes the following steps: receiving a specific voice fragment; cutting the received specific voice fragment into a plurality of specific sub-voice clips; performing a voiceprint identification flow to the specific sub-voice clips, respectively; determining whether each of the specific sub-voice clips is an appropriate sub-voice clip according to a result of the voiceprint identification flow; and capturing the appropriate sub-voice clips and operating a voice recognition thereto.

Abstract: A method and system for compressing an audio signal. The method includes receiving a segment of an audio signal and selectively disabling noise suppression for the received segment. The segment is filtered in a noise-suppression module if noise suppression is not disabled. The method also includes calculating an autocorrelation coefficient and an LSP coefficient, predicting a short-term coefficient and long-term coefficients according to the LSP coefficient and calculating one or more bandwidth-expanded correlation coefficients. Further, the method includes determining the type of packet in which to encode the segment. An encoding rate is selected from among a full rate encode, a half-rate encode, and an eight-rate encode if noise suppression is not disabled. An encoding rate is selected from among a full rate encode and a half-rate encode if noise suppression is disabled. Furthermore, the segment is formed into a packet of the determined type and selected rate.

Abstract: Various embodiments describe a dynamic reconfiguration of a media processing system to optimize a latency performance. In an example, a computer system accesses a current latency performance of the media processing system. The latency is associated with performing a codec process on a current configuration of the media processing system. The current configuration includes virtual machines. The computer system estimates, based on the current latency performance and on historical latency performances associated with the current configuration, a next latency performance of the media processing system. The computer system also identifies, from potential configurations, an updated configuration of the media processing system based on a difference between the next latency performance and a target latency performance and on historical performances associated with the potential configurations. The updated configuration specifies an additional number of virtual machines associated with hosting the codec process.

Abstract: A method of performing BandWidth Extension (BWE) includes a frequency band shifting approach to generate an extended high band signal in time domain and a gain determination approach of controlling the energy of the extended high band. The proposed approach allows shifting any size of low band to any size of high band. The BWE scaling gain is estimated by using available filter bank coefficients with extremely low bit rate or without costing any bit, combining three possible gain factors.