Abstract: A system and method are provided for very short pitch detection and coding for speech or audio signals. The system and method include detecting whether there is a very short pitch lag in a speech or audio signal that is shorter than a conventional minimum pitch limitation using a combination of time domain and frequency domain pitch detection techniques. The pitch detection techniques include using pitch correlations in time domain and detecting a lack of low frequency energy in the speech or audio signal in frequency domain. The detected very short pitch lag is coded using a pitch range from a predetermined minimum very short pitch limitation.

Abstract: An apparatus for level estimation of an encoded audio signal is provided. The apparatus has a codebook determinator for determining a codebook from a plurality of codebooks as an identified codebook. The audio signal has been encoded by employing the identified codebook. Moreover, the apparatus has an estimation unit configured for deriving a level value associated with the identified codebook as a derived level value and for estimating a level estimate of the audio signal using the derived level value.

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: An audio signal transmission device for encoding an audio signal includes an audio encoding unit that encodes an audio signal and a side information encoding unit that calculates and encodes side information from a look-ahead signal. An audio signal receiving device for decoding an audio code and outputting an audio signal includes: an audio code buffer that detects packet loss based on a received state of an audio packet, an audio parameter decoding unit that decodes an audio code when an audio packet is correctly received, a side information decoding unit that decodes a side information code when an audio packet is correctly received, a side information accumulation unit that accumulates side information obtained by decoding a side information code, an audio parameter missing processing unit that outputs an audio parameter upon detection of audio packet loss, and an audio synthesis unit that synthesizes decoded audio from the audio parameter.

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 system and method for transferring a voice from one body of recordings to other recordings.

Abstract: An audio signal transmission device for encoding an audio signal includes an audio encoding unit that encodes an audio signal and a side information encoding unit that calculates and encodes side information from a look-ahead signal. An audio signal receiving device for decoding an audio code and outputting an audio signal includes: an audio code buffer that detects packet loss based on a received state of an audio packet, an audio parameter decoding unit that decodes an audio code when an audio packet is correctly received, a side information decoding unit that decodes a side information code when an audio packet is correctly received, a side information accumulation unit that accumulates side information obtained by decoding a side information code, an audio parameter missing processing unit that outputs an audio parameter upon detection of audio packet loss, and an audio synthesis unit that synthesizes decoded audio from the audio parameter.

Abstract: The invention relates to search for cases in a database. According to the proposed method and apparatus, similarity matching is performed between an input case and a set of cases in an initial search to receive similar cases by using a given matching criterion. Then statistics on image and/or non-image-based features associated with the similar cases are calculated and presented to the user with the similar cases. In a search refinement the similar cases are refined by additional features that are determined by the user based on the statistics. The search refinement can be iterative depending on the user's need.

Abstract: A multi-channel signal encoding method and an encoder, where the encoding method includes obtaining a multi-channel signal of a current frame, determining an initial multi-channel parameter of the current frame, determining a difference parameter based on the initial multi-channel parameter of the current frame and multi-channel parameters of previous K frames of the current frame, where the difference parameter represents a difference between the initial multi-channel parameter of the current frame and the multi-channel parameters of the previous K frames, and K is an integer greater than or equal to one, determining a multi-channel parameter of the current frame based on the difference parameter and a characteristic parameter of the current frame, and encoding the multi-channel signal based on the multi-channel parameter of the current frame. Hence, the method and the encoder ensure better accuracy of inter-channel information of a multi-channel signal.

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: Systems and methods are described for concealing packet loss in a received audio stream. Packets of the audio stream may be received in a non-lapped transform domain format, where at least one packet is missing in the stream. The received packets are decoded, and each missing packet in the decoded stream is replaced by a reduced-energy signal block. Each reduced-energy signal block may also be modified at a beginning or ending boundary, and shifted such that a start or end of each missing packet does not coincide with a peak of a transform window of a lapped transform domain format. The raw audio signal may then be encoded into transform windows having the lapped transform domain format. Packet loss concealment may then be performed for selected transform windows that include modified reduced-energy blocks, either prior to transmission or after transmission by the receiving endpoint.

Abstract: An encoding apparatus and a decoding apparatus in a transform between a Modified Discrete Cosine Transform (MDCT)-based coder and a different coder are provided. The encoding apparatus may encode additional information to restore an input signal encoded according to the MDCT-based coding scheme, when switching occurs between the MDCT-based coder and the different coder. Accordingly, an unnecessary bitstream may be prevented from being generated, and minimum additional information may be encoded.

Abstract: In a method of video decoding in a decoder, a merge candidate list of a current coding block is constructed for processing the current coding block with a triangular prediction mode (TPM). The merge candidate list can include merge candidates each having one or two motion vectors. Each motion vector can be associated with a first reference picture list or a second reference picture list. A first motion vector and a second motion vector are determined from the motion vectors of the merge candidates on the merge candidate list. The current block is processed with the TPM with the first and second motion vectors as two motion vector predictors (MVPs) of two triangular partitions of the current coding block.

Abstract: Certain aspects involve optimizing neural networks or other models for assessing risks and generating explanatory data regarding predictor variables used in the model. In one example, a system identifies predictor variables. The system generates a neural network for determining a relationship between each predictor variable and a risk indicator. The system performs a factor analysis on the predictor variables to determine common factors. The system iteratively adjusts the neural network so that (i) a monotonic relationship exists between each common factor and the risk indicator and (ii) a respective variance inflation factor for each common factor is sufficiently low. Each variance inflation factor indicates multicollinearity among the common factors. The adjusted neural network can be used to generate explanatory indicating relationships between (i) changes in the risk indicator and (ii) changes in at least some common factors.

Abstract: A method and apparatus for handling input Line Spectral Frequency, LSF, coefficients. The method comprises determining LSF residual coefficients as first compressed LSF coefficients subtracted from the input LSF coefficients, and transforming the LSF residual coefficients into a warped domain. One of a plurality of gain-shape coding schemes is applied on the transformed LSF residual coefficients in order to achieve gain-shape coded LSF residual coefficients, where the plurality of gain-shape coding schemes have mutually different trade-offs in one or more of gain resolution and shape resolution for one or more of the transformed LSF residual coefficients. A representation of the first compressed LSF coefficients, the gain-shape coded LSF residual coefficients, and information on the applied gain-shape coding scheme are transmitted over a communication channel to a decoder.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for speaker diarization are disclosed. In one aspect, a method includes the actions of receiving audio data corresponding to an utterance. The actions further include determining that the audio data includes an utterance of a predefined hotword spoken by a first speaker. The actions further include identifying a first portion of the audio data that includes speech from the first speaker. The actions further include identifying a second portion of the audio data that includes speech from a second, different speaker. The actions further include transmitting the first portion of the audio data that includes speech from the first speaker and suppressing transmission of the second portion of the audio data that includes speech from the second, different speaker.

Abstract: A method for managing the assistance to a person in response to the emission of an alert includes emitting an alert from a piece of mobile equipment of a first user to a plurality of users; establishing a first two-way communication between the first equipment and a given terminal of the first set of an assisting user; automatic generating of a plurality of first notifications to a subset of terminals of the first set, each one of the notifications including at least one piece of data that identifies the assisting user; automatic generating of a plurality of second notifications to the second subset, each second notification including a status relative to the processing of the alert by the assisting user.

Abstract: The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described.

Abstract: A circuit for sound activity detection includes a transducer (106) adapted to generate an electrical signal based on detected sound; a variable gain amplifier adapted to amplify the electrical signal to generate an amplified electrical signal; a comparator adapted to compare the amplified electrical signal with at least one first threshold level to generate a comparison signal indicating comparator events; and a control circuit adapted to generate, based on the comparison signal, a gain control signal for controlling the gain of the variable gain amplifier, and a sound activity alert signal indicating the detection of sound activity.

Abstract: According to an aspect of the present invention an encoder for encoding an audio signal has an analyzer configured for deriving prediction coefficients and a residual signal from a frame of the audio signal. The encoder has a formant information calculator configured for calculating a speech related spectral shaping information from the prediction coefficients, a gain parameter calculator configured for calculating a gain parameter from an unvoiced residual signal and the spectral shaping information and a bitstream former configured for forming an output signal based on an information related to a voiced signal frame, the gain parameter or a quantized gain parameter and the prediction coefficients.

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 of redundantly encoding data includes formatting the data into columns and rows, and generating first and second sets of projections of the data using an encoding transform. For each set of projections generated, an encoding parameter of the encoding transform is set to a different value. The first and second sets of projections are stored as the encoded data. A decoding method reads settings including an indication of a number of data fragments. The number of data fragments is compared to a number of projections in a first set of projections of the encoded data in order to determine whether to use a first or a second decoding mode. The encoded data is then decoded according to the selected decoding mode and the result is outputted.

Abstract: The present disclosure provides a method and an apparatus for correcting an input speech based on artificial intelligence. The method includes: receiving a speech input by a user; performing recognition on the speech to obtain a current recognition text; obtaining at least one candidate phrase of a first phrase to be corrected in the current recognition text and displaying the at least one candidate phrase to the user; detecting a select operation of the user, the select operation being configured to select one of the at least one candidate phrase as a target candidate phrase; and correcting the first phrase in the current recognition text by using the target candidate phrase, to obtain a target recognition text.

Abstract: 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. Therefore, the method and the apparatus can accurately distinguish an active voice and an inactive voice.

Abstract: A method and apparatus for performing coding and decoding for high-frequency bandwidth extension. The decoding apparatus may include: a mode checking unit to check mode information of each of frames included in a bitstream; a first core decoding unit to perform code excited linear prediction (CELP) decoding on a CELP coded frame, when a core coding mode of a low-frequency signal indicates a CELP coding mode; a first extension decoding unit to generate a decoded signal of a high-frequency band by using at least one of a result of the performing the CELP decoding and an excitation signal of the low-frequency signal; a second core decoding unit to perform audio decoding on an audio coded frame, when the core coding mode indicates an audio coding mode; and a second extension decoding unit to generate a decoded signal of the high-frequency band by performing frequency-domain (FD) extension decoding.

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: Encoding of a video file includes determining a plurality of scenes associated with a video file, and determining at least one group of pictures (GOP). Starting sequentially from a beginning frame of the video file, the system identifies a first GOP having a first encoding error characteristic. The system changes a bitrate allocation setting from a first setting to a second setting based on the encoding error characteristic. The system identifies a second frame having a second encoding error characteristic, and changes a second bitrate allocation setting from the second setting to a third setting based on the second encoding error characteristic. The system generates an encoded video file that includes an encoded plurality of scenes.

Abstract: Systems, methods, and devices are disclosed for detecting an active speaker in a two-way conference. Real time audio in one or more sub band domains are analyzed according to an echo cancellor model. Based on the analyzed real time audio, one or more audio metrics are determined from output from an acoustic echo cancellation linear filter. The one or more audio metrics are weighted based on a priority, and a speaker status is determined based on the weighted one or more audio metrics being analyzed according to an active speaker detection model. For an active speaker status, one or more residual echo or noise is removed from the real time audio based on the one or more audio metrics.

Abstract: An audio packet error concealment system includes an encoding unit for encoding an audio signal consisting of a plurality of frames, and an auxiliary information encoding unit for estimating and encoding auxiliary information about a temporal change of power of the audio signal. The auxiliary information is used in packet loss concealment in decoding of the audio signal. The auxiliary information about the temporal change of power may contain a parameter that functionally approximates a plurality of powers of subframes shorter than one frame, or may contain information about a vector obtained by vector quantization of a plurality of powers of subframes shorter than one frame.

Abstract: The present disclosure relates to a compression/encoding apparatus and method, a decoding apparatus and method, and a program that allow for provision of a lossless compression technology with higher compression ratio. A GOB data configuration section configures GOB data with a group of digital data that includes a plurality of blocks by treating a frame of delta-sigma-modulated digital data as a block. A table generation section generates a conversion table for encoding the GOB data. An encoding section compresses and encodes the digital data of each block included in the GOB data by using the conversion table. The present technology is applicable, for example, to audio signal compression and encoding, and so on.

Abstract: An envelope sequence is provided that can improve approximation accuracy near peaks caused by the pitch period of an audio signal. A periodic-combined-envelope-sequence generation device according to the present invention takes, as an input audio signal, a time-domain audio digital signal in each frame, which is a predetermined time segment, and generates a periodic combined envelope sequence as an envelope sequence. The periodic-combined-envelope-sequence generation device according to the present invention comprises at least a spectral-envelope-sequence calculating part and a periodic-combined-envelope generating part. The spectral-envelope-sequence calculating part calculates a spectral envelope sequence of the input audio signal on the basis of time-domain linear prediction of the input audio signal.

Abstract: A linear prediction-based noise signal processing method, includes obtaining a linear prediction coefficient of the noise signal, filtering a signal derived from the noise signal based on the linear prediction coefficient in order to obtain a linear prediction residual signal, obtaining excitation energy of the linear prediction residual signal and a spectral envelope of the linear prediction residual signal, and the spectral envelope, the excitation energy and the linear prediction coefficient are encoded.

Abstract: The invention provides methods and devices for outputting a stereo audio signal having a left channel and a right channel. The apparatus includes a demultiplexer, decoder, and upmixer. The upmixer is configured operate either in a prediction mode or a non-prediction mode based on a parameter encoded in the audio bitstream.

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: An encoder for encoding an audio signal, audio transmission system and method for determining correction values includes an analyzer for analyzing the audio signal and for determining analysis prediction coefficients from the audio signal. Including 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 and 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 obtaining a quantized representation of the converted prediction coefficients.

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: 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 for differentiator-based compression of digital data includes (a) using a subtraction module, subtracting a predicted signal from a sample of an original signal to obtain an error signal, (b) using a quantization module, quantizing the error signal to obtain a quantized error signal, and (c) generating the predicted signal using a least means square (LMS)-based filtering method.

Abstract: The coding efficiency of an audio codec using a controllable—switchable or even adjustable—harmonic filter tool is improved by performing the harmonicity-dependent controlling of this tool using a temporal structure measure in addition to a measure of harmonicity in order to control the harmonic filter tool. In particular, the temporal structure of the audio signal is evaluated in a manner which depends on the pitch. This enables to achieve a situation-adapted control of the harmonic filter tool so that in situations where a control made solely based on the measure of harmonicity would decide against or reduce the usage of this tool, although using the harmonic filter tool would, in that situation, increase the coding efficiency, the harmonic filter tool is applied, while in other situations where the harmonic filter tool may be inefficient or even destructive, the control reduces the appliance of the harmonic filter tool appropriately.

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 method including decomposing a magnitude part of a signal spectrum of a mixture signal into spectral components, each spectral component including a frequency part and a time activation part; and clustering the spectral components to obtain one or more clusters of spectral components, wherein the clustering of the spectral components is computed in the time domain.

Abstract: A speech-based human-machine interface that parses words spoken to detect a complete parse and, responsive to so detecting, computes a hypothesis as to whether the words are a prefix to another complete parse. The duration of no voice activity period to determine an end of a sentence depends on the prefix hypothesis. The user's typical speech speed profile and a short-term measure of speech speed also scale the period. Speech speed is measured by the time between words, and the period scaling uses a continuously adaptive algorithm. The system uses a longer cut-off period after a system wake-up event but before it detects any voice activity.

Abstract: The current disclosure provides a method for transmitting encoded information signals through a control system and to a decoder. The encoded information signals are transmitted along with control signals as an encoded message. The information signals are encoded based at least in part on a control-coding capacity of the control system.

Abstract: A Unified Speech and Audio Codec (USAC) that may process a window sequence based on mode switching is provided. The USAC may perform encoding or decoding by overlapping between frames based on a folding point when mode switching occurs. The USAC may process different window sequences for each situation to perform encoding or decoding, and thereby may improve a coding efficiency.

Abstract: A method and device for automatically increasing the spectral bandwidth of an audio signal including generating a “mapping” (or “prediction”) matrix based on the analysis of a reference wideband signal and a reference narrowband signal, the mapping matrix being a transformation matrix to predict high frequency energy from a low frequency energy envelope, generating an energy envelope analysis of an input narrowband audio signal, generating a resynthesized noise signal by processing a random noise signal with the mapping matrix and the envelope analysis, high-pass filtering the resynthesized noise signal, and summing the high-pass filtered resynthesized noise signal with the input narrowband audio signal. Other embodiments are disclosed.

Abstract: According to an example aspect of the present invention, there is provided an apparatus comprising at least one processing core, at least one memory including computer program code, the at least one memory and the computer program code being configured to, with the at least one processing core, cause the apparatus at least to provide an input data item to a first convolutional layer of an artificial neural network comprising a set of convolutional layers, process the input data item in the set of convolutional layers, define, in a feature map output from a last convolutional layer of the set of convolutional layers, a first feature map patch and a second feature map patch, and provide the first feature map patch to a first classifier and the second feature map patch to a second classifier.

Abstract: An encoder for encoding an audio signal has: an analyzer configured for deriving prediction coefficients and a residual signal from an unvoiced frame of the audio signal; a gain parameter calculator configured for calculating a first gain parameter information for defining a first excitation signal related to a deterministic codebook and for calculating a second gain parameter information for defining a second excitation signal related to a noise-like signal for the unvoiced frame; and a bitstream former configured for forming an output signal based on an information related to a voiced signal frame, the first gain parameter information and the second gain parameter information.

Abstract: A decoder for processing an audio signal receives an audio bitstream, decodes the bitstream to obtain a set of spectral frequency parameters that are arranged in an order of frequencies, determines a minimum spectral frequency parameter difference from a plurality of calculated spectral frequency parameter differences, determines a start frequency bin for predicting a high band excitation signal according to the minimum spectral frequency parameter difference, generates the high band excitation signal by selecting a frequency band with a preset bandwidth selected from a low band excitation signal according to the start frequency bin, and synthesizes a wideband signal based on the generated high band excitation signal.

Abstract: A transmitting device for generating a plurality of encoded portions of a video to be transmitted to a receiving device over a network configured to: receive an error message over a feedback channel from the receiving device indicating at least one of said plurality of encoded portions that has been lost at the receiving device; encode a recovery portion responsive to said receiving said error message; and transmit said recovery portion to the receiving device over said network; wherein said error message includes information pertaining to a decoded portion successfully decoded at the receiving device and said recovery portion is encoded relative to said decoded portion.

Abstract: An apparatus for generating a bandwidth extended signal from a bandwidth limited audio signal, the bandwidth limited audio signal The patch generator is configured to perform a harmonic patching algorithm to obtain the patched signal. The signal manipulator is configured for manipulating a signal before patching or the patched signal. The timely preceding bandwidth limited time block timely precedes the current bandwidth limited time block in the plurality of consecutive bandwidth limited time blocks of the bandwidth limited audio signal. The combiner is configured for combining the bandwidth limited audio signal having the core frequency band and the manipulated patched signal having the upper frequency band to obtain the bandwidth extended signal.