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: 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: 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: 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: 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 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: 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 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: 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.

Abstract: A computer-implemented method, according to one embodiment, includes: splitting a video stream into a plurality of sub-streams, encoding each of the sub-streams in parallel, and transmitting a packet to a receiver, wherein the packet includes a number of the encoded sub-streams. Each of the sub-streams individually provide a unique version of the video stream having a lower video quality, while merging more than one of the sub-streams provides a merged version of the video stream having a relatively higher video quality. Moreover, the number of the encoded sub-streams included in the packet transmitted to the receiver corresponds to a quality of service associated with the receiver. Other systems, methods, and computer program products are described in additional embodiments.

Abstract: The present invention provides a bandwidth extension method and apparatus. The method includes: acquiring a bandwidth extension parameter, where the bandwidth extension parameter includes one or more of the following parameters: a linear predictive coefficient (LPC), a line spectral frequency (LSF) parameter, a pitch period, a decoding rate, an adaptive codebook contribution, and an algebraic codebook contribution; and performing, according to the bandwidth extension parameter, bandwidth extension on a decoded low frequency band signal, to obtain a high frequency band signal. The high frequency band signal recovered by using the bandwidth extension method and apparatus in the embodiments of the present invention is close to an original high frequency band signal, and the quality is satisfactory.

Abstract: The present invention extends to methods, systems, and computing system program products for iteratively calculating autocorrelation function for streamed data in real time by iteratively calculating one or more components of autocorrelation function. Embodiments of the invention include iteratively calculating one or more components of autocorrelation function at a specified range of lags in an adjusted computation window based on the one or more components of the autocorrelation function at the specified range of lags calculated for a previous computation window and then calculating the autocorrelation function at the specified range of lags using the iteratively calculated components. Iteratively calculating autocorrelation function avoids visiting all data elements in the adjusted computation window and performing redundant computations thereby increasing calculation efficiency, saving computing resources and reducing computing system's power consumption.

Abstract: The purpose of the present invention is to estimate, with a small amount of computation, a linear prediction synthesis filter after conversion of an internal sampling frequency. A linear prediction coefficient conversion device is a device that converts first linear prediction coefficients calculated at a first sampling frequency to second linear prediction coefficients at a second sampling frequency different from the first sampling frequency, which includes a means for calculating, on the real axis of the unit circle, a power spectrum corresponding to the second linear prediction coefficients at the second sampling frequency based on the first linear prediction coefficients or an equivalent parameter, a means for calculating, on the real axis of the unit circle, autocorrelation coefficients from the power spectrum, and a means for converting the autocorrelation coefficients to the second linear prediction coefficients at the second sampling frequency.

Abstract: A method includes filtering, at a speech encoder, an audio signal into a first group of sub-bands within a first frequency range and a second group of sub-bands within a second frequency range. The method also includes generating a harmonically extended signal based on the first group of sub-bands. The method further includes generating a third group of sub-bands based, at least in part, on the harmonically extended signal. The third group of sub-bands corresponds to the second group of sub-bands. The method also includes determining a first adjustment parameter for a first sub-band in the third group of sub-bands or a second adjustment parameter for a second sub-band in the third group of sub-bands. The first adjustment parameter is based on a metric of a first sub-band in the second group of sub-bands, and the second adjustment parameter is based on a metric of a second sub-band in the second group of sub-bands.

Abstract: A method and an apparatus for reading/writing data from/into a flash memory, and user equipment are presented. The method includes receiving a read data instruction, where the read data instruction includes a size of to-be-read data and a physical address of the to-be-read data in the flash memory; searching a buffer for the physical address; when the physical address is not found in the buffer, dividing a buffer data area from internal memory according to the size of the to-be-read data and an actual physical block size of the flash memory; and reading the to-be-read data from the flash memory according to the physical address and buffering the to-be-read data into the buffer data area.

Abstract: A method and apparatus for performing coding and decoding for high-frequency bandwidth extension. The coding apparatus may down-sample an input signal, perform core coding on the down-sampled input signal, perform frequency transformation on the input signal, and perform bandwidth extension coding by using a base signal of the input signal in a frequency domain.

Abstract: Present disclosure provide a linear prediction-based noise signal processing method includes: acquiring a noise signal, and obtaining a linear prediction coefficient according to the noise signal; filtering the noise signal according to the linear prediction coefficient, to obtain a linear prediction residual signal; obtaining a spectral envelope of the linear prediction residual signal according to the linear prediction residual signal; and encoding the spectral envelope of the linear prediction residual signal. According to the noise processing method, the noise generation method, the encoder, the decoder, and the encoding and decoding system that are in the embodiments of the present disclosure, more spectral details of an original background noise signal can be recovered, so that comfort noise can be closer to original background noise in terms of subjective auditory perception of a user, and subjective perception quality of the user is improved.

Abstract: An autocorrelation calculating part calculates autocorrelation Ro(i) from an input signal. A predictive coefficient calculating part performs linear predictive analysis using modified autocorrelation R?o(i) obtained by multiplying the autocorrelation Ro(i) by a coefficient wo(i). Here, it is assumed that a case where, for at least part of each order i, the coefficient wo(i) corresponding to each order i monotonically increases as a value having negative correlation with a fundamental frequency of an input signal in a current frame or a past frame increases and a case where the coefficient wo(i) monotonically decreases as a value having positive correlation with a pitch gain in a current frame or a past frame increases, are included.

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: In encoding, pitch periods for time series signals in a predetermined time interval are calculated, and a code corresponding thereto is output. In that encoding, the resolutions for expressing the pitch periods and/or a pitch period encoding mode are switched according to whether an index indicating a periodicity and/or stationarity level of the time series signals satisfies a condition indicating high or low in periodicity and/or stationarity. In that decoding, according to whether an index indicating a periodicity and/or stationarity level, the index being included in or obtained from an input code corresponding to the predetermined time interval, satisfies a condition indicating high periodicity and/or stationarity, a decoding mode for a code, included in the input code, corresponding to pitch periods is switched to decode the code corresponding to the pitch periods to obtain the pitch periods corresponding to the predetermined time interval.

Abstract: In encoding, pitch periods for time series signals in a predetermined time interval are calculated, and a code corresponding thereto is output. In that encoding, the resolutions for expressing the pitch periods and/or a pitch period encoding mode are switched according to whether an index indicating a periodicity and/or stationarity level of the time series signals satisfies a condition indicating high or low in periodicity and/or stationarity. In that decoding, according to whether an index indicating a periodicity and/or stationarity level, the index being included in or obtained from an input code corresponding to the predetermined time interval, satisfies a condition indicating high periodicity and/or stationarity, a decoding mode for a code, included in the input code, corresponding to pitch periods is switched to decode the code corresponding to the pitch periods to obtain the pitch periods corresponding to the predetermined time interval.

Abstract: In encoding, pitch periods for time series signals in a predetermined time interval are calculated, and a code corresponding thereto is output. In that encoding, the resolutions for expressing the pitch periods and/or a pitch period encoding mode are switched according to whether an index indicating a periodicity and/or stationarity level of the time series signals satisfies a condition indicating high or low in periodicity and/or stationarity. In that decoding, according to whether an index indicating a periodicity and/or stationarity level, the index being included in or obtained from an input code corresponding to the predetermined time interval, satisfies a condition indicating high periodicity and/or stationarity, a decoding mode for a code, included in the input code, corresponding to pitch periods is switched to decode the code corresponding to the pitch periods to obtain the pitch periods corresponding to the predetermined time interval.

Abstract: An exemplary method is implemented by end-user communication devices for transmitting and receiving HTTP messages in which acknowledgement of receipt by an addressee of an email is requested by an originator of the email. An acknowledgement request command is stored in a command portion of a user data segment of a received HTTP message. Upon receipt of the HTTP message by the addressee device, a reply HTTP message with an acknowledgement request must be authorized to be transmitted to the originator in order to allow the addressee to be able to view the content of the received HTTP message.

Abstract: A method of performing finite field addition and doubling operations in an elliptic curve cryptography (ECC) authentication scheme as a countermeasure to side-channel attack. The addition and doubling operations are executed using atomic patterns that involve the same sequence and number of operation types, so that the noise consumption and electromagnetic emanation profile of circuitry performing the operations is identical regardless of operation. A subtraction operation using such an atomic pattern is also disclosed.

Abstract: A system, method and computer program product are described for voice activity detection (VAD) within a digitally encoded bitstream. A parameter extraction module is configured to extract parameters from a sequence of coded frames from a digitally encoded bitstream containing speech. A VAD classifier is configured to operate with input of the digitally encoded bitstream to evaluate each coded frame based on bitstream coding parameter classification features to output a VAD decision indicative of whether or not speech is present in one or more of the coded frames.

Abstract: A method for restoring distorted speech components of an audio signal distorted by a noise reduction or a noise cancellation includes determining distorted frequency regions and undistorted frequency regions in the audio signal. The distorted frequency regions include regions of the audio signal in which a speech distortion is present. Iterations are performed using a model to refine predictions of the audio signal at distorted frequency regions. The model is configured to modify the audio signal and may include deep neural network trained using spectral envelopes of clean or undamaged audio signals. Before each iteration, the audio signal at the undistorted frequency regions is restored to values of the audio signal prior to the first iteration; while the audio signal at distorted frequency regions is refined starting from zero at the first iteration. Iterations are ended when discrepancies of audio signal at undistorted frequency regions meet pre-defined criteria.

Abstract: The present disclosure is directed towards a system and method for reducing tandeming effects in a communications system. The method may include receiving, at a speech decoder, an input bitstream associated with an incoming initial speech signal from a speech encoder. The method may further include determining whether or not coding is required and if coding is required, modifying an excitation signal associated with the bitstream. The method may also include providing the modified excitation signal to an adaptive encoder.

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.

Abstract: The invention is a method that combines the following components: a high pass filter designed to have sufficiently small phase delay and roll-off value in transition band as well as sufficiently good attenuation; a distortion detection and reconstruction introduced by the application of the high pass filter by extraction the significant frequency components in relevant frequency band; a signal compensation that reshapes the output of the high pass filter by matching the filter's phase delay and attenuation characteristics so as to approximate low frequency component extraction that would be produced by an ideal filter (very sharp frequency transition and no delay); a time-domain detection and correction method that addresses special circumstances under which the compensation would be inaccurate to achieve real-time estimate in normal circumstances, and a time-domain correction method during and immediately after sudden changes in composite signal (spike) is detected.

Abstract: Techniques are described for detecting and/or validating audible taps, double-taps, or other tapped audio sequences generated by a user. A predicted voice signal is subtracted from an audio signal to produce a prediction error signal. The prediction error signal is analyzed in terms of dynamics and periodicity to determine whether it contains an audio pulse. In certain cases, prediction error signals corresponding to multiple microphones are analyzed to determine position information regarding the individual taps of a double-tap. A double-tap is validated and acted upon when the individual taps originate from the same location and the taps occur within a given time period.

Abstract: A system and method are presented for the synthesis of speech from provided text. Particularly, the generation of parameters within the system is performed as a continuous approximation in order to mimic the natural flow of speech as opposed to a step-wise approximation of the feature stream. Provided text may be partitioned and parameters generated using a speech model. The generated parameters from the speech model may then be used in a post-processing step to obtain a new set of parameters for application in speech synthesis.

Abstract: Embodiments of the present invention provide a signal decoding method and device. The method includes decoding a bit stream of a voice signal or an audio signal to acquire a decoded signal; predicting an excitation signal of an extension band according to the decoded signal, where the extension band is adjacent to a band of the decoded signal, and the band of the decoded signal is lower than the extension band; selecting a first band and a second band from the decoded signal, and predicting a spectral envelope of the extension band according to a spectral coefficient of the first band and a spectral coefficient of the second band; and determining a frequency-domain signal of the extension band according to the spectral envelope of the extension band and the excitation signal of the extension band.

Abstract: A computer-implemented method performed by a data processing apparatus includes receiving an audio signal that includes a frequency-domain representation of an audio file, extracting, from the audio signal, a plurality of frequency-domain data values that correspond to at least a portion of the audio file, compressing the plurality of data values to form a compressed frequency domain value file, and transmitting the compressed frequency domain value file to a server to identify the audio file.

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: A vector joint encoding/decoding method and a vector joint encoder/decoder are provided, more than two vectors are jointly encoded, and an encoding index of at least one vector is split and then combined between different vectors, so that encoding idle spaces of different vectors can be recombined, thereby facilitating saving of encoding bits, and because an encoding index of a vector is split and then shorter split indexes are recombined, thereby facilitating reduction of requirements for the bit width of operating parts in encoding/decoding calculation.

Abstract: A method is provided. The method includes obtaining a low-band spectrum of an audio signal in which a low-band signal is frequency transformed; obtaining phase information of a high-band spectrum of the audio signal based on the low-band spectrum; and outputting a bitstream that comprises the phase information of the high-band spectrum.

Abstract: A method, electronic device, computer program product, system and circuit assembly are provided for allocating one or more redundant pictures by taking into consideration the information content of the primary pictures, with which the redundant pictures would be associated. In particular, primary pictures that are determined to be more sensitive to transmission loss or corruption may be allocated one or more redundant pictures, while those that are less sensitive may not be so allocated. By selectively allocating redundant pictures to only those primary pictures that are more sensitive, the method disclosed reduces the amount of overhead associated with redundant pictures and increases the coding efficiency, without sacrificing the integrity of the video data.

Abstract: Disclosed are methods and systems for significantly compressing sparse multidimensional ordered series data comprised of indexed data sets, wherein each data set comprises an index, a first variable and a second variable. The methods and systems are particularly suited for compression of data recorded in double precision floating point format.

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 high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

Abstract: Provided are a method and apparatus for encoding and decoding an audio signal. According to the present application, a signal of a high frequency band above a preset frequency band is adaptively encoded or decoded in the time domain or in the frequency domain by using a signal of a low frequency band below the preset frequency band. As such, the sound quality of a high frequency signal is not deteriorate even when an audio signal is encoded or decoded by using a small number of bits and thus coding efficiency may be maximized.

Abstract: A system and method for encouraging an energy user to conserve energy by providing the user with musical feedback describing their energy use as a musical piece. Various musical parameters of the musical piece may be manipulated to describe higher- or lower-than-expected energy use, in order to encourage the user to conserve energy. A visual animation may also be added to the musical piece to illustrate it further.

Abstract: The dynamic range and power efficiency of a voice-activated system may be improved by dynamically adjusting the configuration of the voice-activated system's input path. In one embodiment, a first portion of audio may be received through an input path of the voice-activated system having a first configuration. A characteristic of the first portion of audio may be determined and the input path may be adjusted to a second configuration based on the determined characteristic. A second portion of audio may then be received through the input path having the second configuration, and speech analysis may be performed on the second portion of audio.

Abstract: Adapted speech models produce fluent synthesized speech in a voice that sounds as if the speaker were fluent in a language in which the speaker is actually non-fluent. A full speech model is obtained based on fluent speech in the language spoken by a first person who is fluent in the language. A limited set of utterances is obtained in the language spoken by a second person who is non-fluent in the language but able to speak the limited set of utterances in the language. The full speech model of the first person is then processed with the limited set of utterances of the second person to produce an adapted speech model. The adapted speech model may be stored to a multi-lingual speech model as a child node of a root with an associated language selection question and branches pointed to the adapted speech model and other speech models, respectively.