Abstract: A method and an apparatus for calculating a downmixed signal and a residual signal are provided. According to the method, if a first target frame (a current frame or a previous frame of the current frame) is a switching frame, a to-be-encoded downmixed signal and a to-be-encoded residual signal of the subband corresponding to the preset frequency band in the current frame is calculated based on a switch fade-in/fade-out factor of a second target frame, an initial downmixed signal and an initial residual signal of the preset frequency band.

Abstract: A method and a system for vision-based defect detection are proposed. The method includes the following steps. A test audio signal is outputted to a device-under-test (DUT), and a response signal of the DUT with respect to the test audio signal is received to generate a received audio signal. Signal processing is performed on the received audio signal to generate a spectrogram, and whether the DUT has an unacceptable defect with respect to the predefined auditory standard is determined by analyzing a distribution of signal strength according to the spectrogram.

Abstract: A method for encoding video data comprises generating coefficients based on video data; generating coefficient vectors, wherein each of the coefficient vectors includes n of the coefficients; for each of the coefficient vectors, determining an amplitude value for the coefficient vector based on a mapping pattern, wherein for each respective allowed coefficient vector in a plurality of allowed coefficient vectors: the mapping pattern maps the respective allowed coefficient vector to a respective amplitude value in a plurality of amplitude values, and the respective amplitude value is adjacent in an n-dimensional space to at least one other amplitude value in the plurality of amplitude values that is adjacent to the respective amplitude value in a monotonic number line of the amplitude values; and modulating an analog signal based on the amplitude values for the coefficient vectors.

Abstract: A device for signal processing includes a memory and a processor. The memory is configured to store a parameter associated with a bandwidth-extended audio stream. The processor is configured to select a plurality of non-linear processing functions based at least in part on a value of the parameter. The processor is also configured to generate a high-band excitation signal based on the plurality of non-linear processing functions.

Abstract: A voice synthesis method includes: supplying a first trained model with control data including phonetic identifier data to generate a series of frequency spectra of harmonic components; supplying a second trained model with the control data to generate a waveform signal representative of non-harmonic components; and generating a voice signal including the harmonic components and the non-harmonic components based on the series of frequency spectra of the harmonic components generated by the first trained model and the waveform signal representative of the non-harmonic components generated by the second trained model.

Abstract: A pitch enhancement apparatus obtains an output signal by performing, for each time segment, pitch enhancement processing on a signal derived from an input audio signal. The pitch enhancement apparatus includes a pitch enhancement unit that performs, as the pitch enhancement processing, for each time segment, processing to obtain, as an output signal at each time of the time segment, a signal including the signal at the time, a first signal for enhancement that is the signal at a time which is an earlier time than the time by the number of samples T0 corresponding to a pitch period of the time segment, and a second signal for enhancement that is the signal at a time which is an earlier time than the time by the number of samples T?? corresponding to a pitch period of the ?-th time segment previous to the time segment.

Abstract: Systems and methods provide a first sample of audio data and detect speech onset in the first sample of the audio data. Responsive to detecting the speech onset, systems and methods switch from capturing second samples of the audio data at first intervals, to capturing the second samples of the audio data at second intervals. Systems and methods provide contiguous audio data using the second samples of the audio data captured at the first intervals and at least one captured portion of the second samples of the audio data captured at the second intervals.

Abstract: A stereo signal encoding method includes determining a window length of an attenuation window based on an inter-channel time difference, determining a modified linear prediction analysis window based on the window length of the attenuation window, where values of at least some points from a point (L?sub_window_len) to a point (L?1) in the modified linear prediction analysis window are less than values of corresponding points from a point (L?sub_window_len) to a point (L?1) in an initial linear prediction analysis window, and the window length of the modified linear prediction analysis window is equal to a window length of the initial linear prediction analysis window, and performing linear prediction analysis on a to-be-processed sound channel signal based on the modified linear prediction analysis window.

Abstract: A method and apparatus for recalling news based on artificial intelligence, a device and a storage medium. The method comprises: building an index repository according to candidate news, the index repository including M search trees, each search tree being a complete binary tree including at least two layers, each non-leaf node in each search tree corresponding to a semantic index vector, each piece of candidate news corresponding to a leaf node in each search tree; when news needs to be recommended to the user, generating a user's semantic index vector according to the user's interest tag; with respect to each search tree, respectively according to semantic index vectors corresponding to non-leaf nodes therein and the user's semantic index vector, determining a path from a first layer of non-leaf nodes to a leaf node, and regarding candidate news corresponding to the leaf node on the path as a recall result.

Abstract: A method comprises determining a first modification weight according to linear spectral frequency (LSF) differences of the current frame and LSF differences of a previous frame of the current frame, when a signal characteristic of the current frame meets a preset modification condition, modifying the linear predictive parameter of the current frame according to the determined first modification weight, and coding the current frame according to the modified linear predictive parameter.

Abstract: The present technology relates to an information processing device and an information processing method that enable reduction of an amount of data to be transmitted in transmission of data of a plurality of audio objects. An information processing device according to one aspect of the present technology combines audio objects with sounds that are undistinguishable at a predetermined supposed listening position among a plurality of audio objects for the predetermined supposed listening position among a plurality of supposed listening positions and transmits data of a combined audio object obtained by the combination, along with data of other audio objects with sounds that are distinguishable at the predetermined supposed listening position. The present technology can be applied to a device that can process object-based audio data.

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: An apparatus for processing an encoded audio signal having a plurality of downmix signals associated with a plurality of input audio objects and object parameters. The apparatus includes a grouper configured to group the downmix signals into groups of downmix signals associated with a set of input audio objects. The apparatus includes a processor configured to perform at least one processing step individually on the object parameters of each set of input audio objects in order to provide group results. Further, there is a combiner configured to combine the group results or processed group results in order to provide a decoded audio signal. The invention also refers to a corresponding method.

Abstract: An information processing device enables a user to register a wake-up-word for activating a predetermined function by voice recognition. The information processing device includes a receiving unit configured to receive, from a user, an input word for registering a wake-up-word, and a determination unit configured to determine whether the input word received by the receiving unit satisfies conditions for an accuracy of voice recognition.

Abstract: A device for signal processing with a receiver that receives an encoded audio signal comprising a parameter, and based on the value of parameter selects one non-linear processing function for generating a first excitation signal, a second non-linear processing function for generating a second excitation signal, and generates a high-band excitation signal based on the first excitation signal and second excitation signal.

Abstract: A stereo sound decoding method and system decode left and right channels of a stereo sound signal, using received encoding parameters comprising encoding parameters of a primary channel, encoding parameters of a secondary channel, and a factor ?. The primary channel encoding parameters comprise LP filter coefficients of the primary channel. The primary channel is decoded in response to the primary channel encoding parameters. The secondary channel is decoded using one of a plurality of coding models, wherein at least one of the coding models uses the primary channel LP filter coefficients to decode the secondary channel. The decoded primary and secondary channels are time domain up-mixed using the factor ? to produce the decoded left and right channels of the stereo sound signal, wherein the factor ? determines respective contributions of the primary and secondary channels upon production of the left and right channels.

Abstract: A multi-channel audio decoder for providing at least two output audio signals on the basis of an encoded representation is configured to perform a weighted combination of a downmix signal, a decorrelated signal and a residual signal, to obtain one of the output audio signals. The multi-channel audio decoder is configured to determine a weight describing a contribution of the decorrelated signal in the weighted combination in dependence on the residual signal. A multi-channel audio encoder for providing an encoded representation of a multi-channel audio signal is configured to obtain a downmix signal on the basis of the multi-channel audio signal, to provide parameters describing dependencies between the channels of the multi-channel audio signal, and to provide a residual signal. The multi-channel audio encoder is configured to vary an amount of residual signal included into the encoded representation in dependence on the multi-channel audio signal.

Abstract: According to some embodiments, each of a plurality of computer systems to be monitored receives a monitoring solution agent code portion and a monitoring solution agent content portion, the monitoring solution agent code portion and monitoring solution agent content portion together forming a monitoring solution agent. The monitoring solution agent code portion and monitoring solution agent content portion may comprise a remote Monitoring Solution Agent that executes at the computer system to be monitored. According to some embodiments, monitoring solution agent content portion includes a data source layer, a data provider layer, a request layer, and a user interface layer.

Abstract: Methods, computer program products, and systems are presented. The methods include, for instance: obtaining an utterance input from a user agent, and collecting context data of the utterance input. A context tag is generated based on the context data, and one or more ground truth having respective utterance semantically identical to the utterance input is selected. Semantical relationship between the context tag and an intent of the selected ground truth is examined and the selected ground truth is updated with the context tag.

Abstract: An audio coding method and apparatus, where the method includes, for each audio frame in audio, when a signal characteristic of the audio frame and a signal characteristic of a previous audio frame meet a preset modification condition, determining a first modification weight according to linear spectral frequency (LSF) differences of the audio frame and the LSF differences of the previous audio frame, modifying a linear predictive parameter of the audio frame according to the determined first modification weight, and coding the audio frame according to a modified linear predictive parameter of the audio frame. According to the present disclosure, audio having a wider bandwidth can be coded while a bit rate remains unchanged or a bit rate slightly changes and a spectrum between audio frames is steadier.

Abstract: Systems and methods for distributed audio mixing are disclosed, comprising providing one or more predefined constellations, each constellation defining a spatial arrangement of points forming a shape or pattern and receiving positional data indicative of the spatial positions of a plurality of audio sources in a capture space. A correspondence may be identified between a subset of the audio sources and a constellation based on the relative spatial positions of audio sources in the subset. Responsive to said correspondence, at least one action may be applied, for example an audio, video and/or controlling action to audio sources of the subset.

Abstract: Generating structured queries from natural language text may include receiving, using a processor, a natural language text input directed to a database management system and, using the processor, performing natural language processing on the natural language text input using an Unstructured Information Management Architecture. The natural language processing may annotate the natural language text input according to a structure of the database management system. A database operation and query elements may be determined using a processor from the annotated natural language text input. A structured query may be created, using the processor, for the database management system that implements the database operation using the query elements.

Abstract: A thermostat for a building includes a communications interface communicably coupled to an HVAC network and a processing circuit. The processing circuit is configure to determine that a HVAC device is connected to the HVAC network and retrieve a software update from the HVAC device via the communications interface and install the software update in response to a determination that the thermostat must install the software update in order to be configured to communicate with the HVAC device.

Abstract: Systems and methods are provided for training an audio generation model for a first person using a first voice audio data and a first text transcript of the first voice audio data. Using a second voice audio data and a second text transcript of the second voice audio data, a plurality of pitch voice audio data for the second person may be generated with different pitches. The audio generation model may be trained for the second person using the generated plurality of pitch voice audio data with the different pitches for the second person. Output voice audio may be generated for the second person using received text and the model trained with the generated plurality of pitch voice audio data.

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: Example embodiments disclosed herein relate to audio signal processing. The audio signal has multiple audio objects. A method of processing an audio signal is disclosed. The method includes obtaining an object position for each of the audio objects; and determining cluster positions for grouping the audio objects into clusters based on the object positions, a plurality of object-to-cluster gains, and a set of metrics. The metrics indicate a quality of the cluster positions and a quality of the object-to-cluster gains, each of the cluster positions is a centroid of a respective one of the clusters, and one of the object-to-cluster gains defines a ratio of the respective audio object in one of the clusters. The method also includes determining the object-to-cluster gains based on the object positions, the cluster positions and the set of metrics; and generating a cluster signal based on the determined cluster positions and object-to-cluster gains.

Abstract: Provided are a vector quantization device, a voice coding device, a vector quantization method, and a voice coding method which enable a reduction in the calculation amount of voice codec without deterioration of voice quality. In the vector quantization device, a first reference vector calculation unit (201) calculates a first reference vector by multiplying a target vector (x) by an auditory weighting LPC synthesis filter (H), and a second reference vector calculation unit (202) calculates a second reference vector by multiplying an element of the first reference vector by a filter having a high pass characteristic. A polarity preliminary selection unit (205) generates a polar vector by disposing a unit pulse having a positive or negative polarity, which is selected on the basis of the polarity of an element of the second reference vector, in the position of said element.

Abstract: A method which includes: detecting phrases in a transcript of an audiovisual file; applying a speech recognition algorithm to the audiovisual file and to a list of words of the phrase, to output a temporal location of each of the words that are uttered in the audio channel; compiling a list of sub-phrases of each of the phrases; creating a temporal sub-phrase map that comprises a temporal location of each of the sub-phrases; extracting the uttered sub-phrases from the audiovisual file, to create multiple sub-phrase audiovisual files; and constructing a database the multiple sub-phrase audiovisual files and of the sub-phrase uttered in each of the files. The method may also include: receiving a phrase; querying the database for audiovisual files which comprise uttered sub-phrases of the phrase; and splicing at least some of the audiovisual files to a compilation audiovisual file in which the phrase is uttered.

Abstract: A device and method for quantizing a gain of a fixed contribution of an excitation in a frame, including sub-frames, of a coded sound signal, wherein the gain of the fixed excitation contribution is estimated in a sub-frame using a parameter representative of a classification of the frame. The gain of the fixed excitation contribution is then quantized in the sub-frame using the estimated gain. The device and method is used in jointly quantizing gains of adaptive and fixed contributions of an excitation in a frame of a coded sound signal. For retrieving a quantized gain of a fixed contribution of an excitation in a sub-frame of a frame, the gain of the fixed excitation contribution is estimated using a parameter representative of a classification of the frame, a gain codebook supplies a correction factor in response to a received, gain codebook index, and a multiplier multiplies the estimated gain by the correction factor to provide a quantized gain of the fixed excitation contribution.

Abstract: An audio encoding method and a related apparatus are disclosed. The audio coding method includes: performing a time-frequency transformation on a current frame of a time-domain audio signal, to obtain spectral coefficients of the current audio frame; obtaining one or more reference coding parameters of the current frame; and determining whether the reference coding parameters satisfy a set of parameter conditions. If any one of the parameter conditions is satisfied, the spectral coefficients of the current frame are encoded by using a transform coded excitation (TCX) algorithm. If none of the parameter conditions is satisfied, the spectral coefficients of the current audio frame are encoded using a high quality transform coding (HQ) algorithm. The audio encoding method and the related apparatus help improve encoding quality or encoding efficiency in audio signal encoding.

Abstract: The present disclosure relates to a telecommunication method for operating a telecommunication device being coupled to a digital telecommunication network. The telecommunication device has a processor, a display device and a memory with an application. The telecommunication device is connectable via the network to a backend server with access to a database comprising a plurality of datasets. The method has: sending an initializing data request to the backend server requesting a first number of the datasets from the database; in response to receiving an initializing data packet with the requested first number of the datasets; and calculating a second number of the datasets to be requested by a second data request.

Abstract: An information processing method includes receiving a change instruction to change a voice parameter used in synthesizing a voice for a set of texts, changing the voice parameter in accordance with the change instruction to change the voice parameter, changing, in accordance with the change instruction, an image parameter used in synthesizing an image of a virtual object, the virtual object indicating a character that vocalizes the voice that has been synthesized, synthesizing the voice using the changed voice parameter, and synthesizing the image using the changed image parameter.

Abstract: An acoustic processing device comprises: a resonant band detecting means that detects a resonant band of sound output from a speaker based on a measurement result of a predetermined measurement signal reproduced through the speaker; an analyzing means that analyzes the measurement result of the predetermined measurement signal; a control parameter generating means that generates a control parameter for controlling the resonant band detected by the resonant band detecting means based on an analysis result by the analyzing means; and an audio signal controlling means that controls an audio signal input from a predetermined audio signal reproducing device based on the control parameter generated by the control parameter generating means such that a resonant band component of reproduced sound of the audio signal is suppressed to be short on a time axis.

Abstract: A method of regenerating wideband speech from narrowband speech, the method comprising: receiving samples of a narrowband speech signal in a first range of frequencies; modulating received samples of the narrowband speech signal with a modulation signal having a modulating frequency adapted to upshift each frequency in the first range of frequencies by an amount determined by the modulating frequency wherein the modulating frequency is selected to translate into a target band a selected frequency band within the first range of signals; filtering the modulated samples using a target band filter to form a regenerated speech signal in the target band; and combining the narrow band speech signal with the regenerated speech signal in the target band to regenerate a wideband speech signal, the method comprising the step of controlling the modulated samples to lie in a second range of frequencies identified by determining a signal characteristic of frequencies in the first range of frequencies.

Abstract: Preprocessing speech signals from an indirect conduction microphone. One exemplary method preprocesses the speech signal in two stages. In stage one, an external speech sample is characterized using an auto regression model, and coefficients from the model are convolved with the internal speech signal from the indirect conduction microphone to produce a pre-conditioned internal speech signal. In stage two, a training sound is received by the indirect conduction microphone and filtered through a low-pass filter. The result is then modeled using auto regression, and inverted to produce an inverted filter model. The pre-conditioned internal speech signal is convolved with the inverted filter model to remove negative or undesirable acoustic characteristics and loss from the speech signal from the indirect conduction microphone.

Abstract: An audio coding method and apparatus, where the method includes, for each audio frame in audio, when a signal characteristic of the audio frame and a signal characteristic of a previous audio frame meet a preset modification condition, determining a first modification weight according to linear spectral frequency (LSF) differences of the audio frame and LSF differences of the previous audio frame, modifying a linear predictive parameter of the audio frame according to the determined first modification weight, and coding the audio frame according to a modified linear predictive parameter of the audio frame. According to the present disclosure, audio having a wider bandwidth can be coded while a bit rate remains unchanged or a bit rate slightly changes and a spectrum between audio frames is steadier.

Abstract: A method performed in an audio decoder for reconstructing an original audio signal having a lowband portion and a highband portion is disclosed. The method includes receiving an encoded audio signal and extracting reconstruction parameters from the encoded audio signal. The method further includes decoding the encoded audio signal with a core audio decoder to obtain a decoded lowband portion and regenerating the highband portion based at least in part on a cross over frequency and the decoded lowband portion to obtain a regenerated highband portion. The method also includes creating a synthetic sinusoid with a level based at least in part on a spectral envelope value for the particular subband and a noise floor value for the particular subband and adding the synthetic sinusoid to the regenerated highband portion in the particular frequency band specified by the location information.

Abstract: A method performed in an audio decoder for reconstructing an original audio signal having a lowband portion and a highband portion is disclosed. The method includes receiving an encoded audio signal and extracting reconstruction parameters from the encoded audio signal. The method further includes decoding the encoded audio signal with a core audio decoder to obtain a decoded lowband portion and regenerating the highband portion based at least in part on a cross over frequency and the decoded lowband portion to obtain a regenerated highband portion. The method also includes creating a synthetic sinusoid with a level based at least in part on a spectral envelope value for the particular subband and a noise floor value for the particular subband and adding the synthetic sinusoid to the regenerated highband portion in the particular frequency band specified by the location information.

Abstract: A gain contribution of the audio signal for each of the N audio objects to at least one of M speakers may be determined. Determining the gain contribution may involve determining a center of loudness position that is a function of speaker (or cluster) positions and gains assigned to each speaker (or cluster). Determining the gain contribution also may involve determining a minimum value of a cost function. A first term of the cost function may represent a difference between the center of loudness position and an audio object position.

Abstract: A system functions to recommend equipment expansions, additions and/or substitutions; interconnections; supplemental capabilities; features; etc. based upon a knowledge of a consumer's existing audio and/or visual system configuration.

Abstract: Speech traits of an entity imbue an artificial intelligence system with idiomatic traits of persons from a particular category. Electronic units of speech are collected from an electronic stream of speech that is generated by a first entity. Tokens from the electronic stream of speech are identified, where each token identifies a particular electronic unit of speech from the electronic stream of speech, and where identification of the tokens is semantic-free. Nodes in a first speech graph are populated with the tokens to develop a first speech graph having a first shape. The first shape is matched to a second shape of a second speech graph from a second entity in a known category. The first entity is assigned to the known category, and synthetic speech generated by an artificial intelligence system is modified based on the first entity being assigned to the known category.

Abstract: An apparatus for decoding an encoded audio signal is provided. The apparatus includes a pulse information decoder and a signal decoder. The pulse information decoder is adapted to decode a plurality of pulse positions, wherein each one of the pulse positions indicates a position of one of the pulses of the track, wherein the pulse information decoder is configured to decode the plurality of pulse positions by using a track positions number, a total pulses number, and one state number. The signal decoder is adapted to decode the encoded audio signal by generating a synthesized audio signal using the plurality of pulse positions and a plurality of predictive filter coefficients.

Abstract: A diabetes care system for detection of an analyte and method for selective data transmission are disclosed. The diabetes care system has a mobile component and a base station, wherein a data transmission occurring between the mobile component and the base station within a time interval in which a wireless communication link exists, wherein is selectively performed in such a manner that within the time interval, a first partial set of the data is transmitted from the mobile component to the base station. The first partial set is selected using a processor-controlled selection algorithm in such a manner that the data transmitted in the time interval is representative of the entirety of the data stored in the mobile component.

Abstract: In a quantization method for quantizing a received excitation signal in a communication system performing the steps of re-shuffling S301 the elements of the received excitation signal to provide a re-shuffled excitation signal; coding S302 the re-shuffled excitation signal with a variable bit-rate algorithm to provide a coded excitation signal; and reassigning S303 codewords of the coded excitation signal if a number of used bits exceeds a predetermined fixed bit rate requirement to provide a quantized excitation signal.

Abstract: A method for coding and decoding an audio signal or speech signal and an apparatus adopting the method are provided.

Abstract: An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal includes a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also includes an audio stream provider for providing the audio stream such that the audio stream includes information describing an audio content of the frequency bands and information describing the multi-band quantization error. A decoder for providing a decoded representation of an audio signal on the basis of an encoded audio stream representing spectral components of frequency bands of the audio signal includes a noise filler for introducing noise into spectral components of a plurality of frequency bands to which separate frequency band gain information is associated on the basis of a common multi-band noise intensity value.

Abstract: Speech traits of an entity imbue an artificial intelligence system with idiomatic traits of persons from a particular category. Electronic units of speech are collected from an electronic stream of speech that is generated by a first entity. Tokens from the electronic stream of speech are identified, where each token identifies a particular electronic unit of speech from the electronic stream of speech, and where identification of the tokens is semantic-free. Nodes in a first speech graph are populated with the tokens to develop a first speech graph having a first shape. The first shape is matched to a second shape of a second speech graph from a second entity in a known category. The first entity is assigned to the known category, and synthetic speech generated by an artificial intelligence system is modified based on the first entity being assigned to the known category.

Abstract: A system and method for receiving an initial search entry to search text data. The text data may be, for example, an N word lattice, transcribed by a text to speech engine. The difference between the initial search entry and one or more entries in the dictionary may be measured. One or more similar entries may be selected from the dictionary that have the smallest measures of difference to the initial search entry. The text data may be searched for the one or more selected similar entries. Each of the searched similar entries found in the text data may be displayed as a search result.

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

Abstract: The invention discloses a method including: performing in a unit of first timeframe frame length, framing on a continuous voice sample to obtain a plurality of first timeframes, detecting energy of each of the first timeframes, and determining a target first timeframe including a potential abrupt exception of a voice signal by analyzing a relationship between the energy of the plurality of first timeframes; performing, in a unit of second timeframe frame length, framing on the continuous voice sample to obtain a plurality of second timeframes, and processing each of the second timeframes to acquire a tone feature, and determining, by analyzing a tone feature of at least one of the second timeframes including at least one target second timeframe, whether the potential abrupt exception of a voice signal included in the target first timeframe included in the target second timeframe is a real abrupt exception of a voice signal.