85 results about "Spectral envelope" patented technology

The invention relates to the technical field of audio processing, in particular to an audio optimization method and device based on artificial intelligence, computer equipment and a medium. According to the method, a linear layer is used for mapping a spectrum envelope of a to-be-optimized audio to obtain an envelope feature, an embedding layer is used for embedding a standard audio parameter into a parameter feature, a prediction model is used for predicting a fusion feature of the envelope feature and the parameter feature to obtain a prediction pitch curve, and a noise adding model is used for adding noise to a Mel spectrum of the to-be-optimized audio to obtain a predicted pitch curve. Calculating the noise of the noise addition result by using a noise estimation model to obtain predicted noise, updating the noise estimation model according to the predicted noise, the real noise and the predicted pitch curve, calculating the reference noise of the noise addition result by using the updated noise estimation model, and de-noising the noise addition result according to the reference noise. The noise estimation model is optimized in combination with pitch information, so that the denoising process meets the pitch requirement, and the audio optimization effect is improved.

The invention belongs to the technical field of radio electronics and signal processing, and particularly relates to a color noise environment signal frequency domain detection method and system based on spectrum envelope extraction, and the system comprises a signal acquisition unit, a signal conversion unit, a signal judgment unit and a signal output unit. The method is a research summary of a color noise environment signal frequency domain detection algorithm based on spectrum envelope extraction, and the algorithm is used for blind signal detection under the conditions of uneven noise spectrum and broadband reception, has good robustness, can be used for low signal-to-noise ratio occasions, and is high in signal detection accuracy.

The invention relates to the technical field of sleep quality evaluation, in particular to a sleep monitoring optimization system based on artificial intelligence. The system comprises the steps of collecting sleep monitoring data of a user at different collection moments, extracting dominant frequency components in snore audio data through spectrum envelope analysis, extracting audio primitives based on the dominant frequency components, inputting the audio primitives into a snore classification model, outputting a snore category corresponding to the snore audio data of the user, and outputting the snore category corresponding to the snore audio data of the user. Carrying out Bayesian inversion analysis on the brain wave data to obtain a posterior sample of random field model parameters, executing reliability analysis based on the posterior sample to calculate a posterior failure probability, obtaining a first sleep quality index of the user based on the posterior failure probability, obtaining a second sleep quality index of the user based on vital sign data analysis, and sending the second sleep quality index to the user; and obtaining a user sleep quality evaluation result based on the snore type, the user first sleep quality index and the user second sleep quality index. The accuracy and efficiency of user sleep quality evaluation can be improved.

The present invention discloses a continuous speech formant extraction method based on peak selection, comprising: performing a preprocessing operation on a single frame of input speech; using a linear prediction method to preliminarily estimate the peak value in the spectral envelope of the speech frame; establishing a reference point and a formant trough, and then using a peak selection method to establish a mapping relationship between the peak value and the reference point; using the mapping relationship between the peak value and the reference point and the formant trough to determine the formant of the speech frame; and performing formant estimation on the continuous speech: dividing the continuous speech into frames according to different frame numbers, using the above algorithm to loop 100 times to obtain the formant parameters under different frame number tests, averaging the results after 100 loops, and obtaining the final result after smoothing. The method of the present invention can eliminate the influence of merged peaks and false peaks, and has a fast convergence speed and strong robustness.

The embodiment of the invention provides an acoustic emission positioning method, device and equipment. The method comprises the following steps: acquiring time domain waveform data of an acoustic emission event of an observation target on a plurality of channels through acquisition equipment; according to the multi-channel time domain waveform data of the acoustic emission event, obtaining a preliminary positioning result of the acoustic emission event by adopting a cross-correlation algorithm; for the time-domain waveform data of each channel, intercepting the time-domain waveform data within a preset duration before and after the arrival time of the impact signal by adopting a preset window function to obtain multi-channel local time-domain waveform data of the acoustic emission event; acquiring a spectrum envelope corresponding to the multi-channel local time domain waveform data of the acoustic emission event; and according to the frequency spectrum envelope, correcting the preliminary positioning result by adopting a coherent superposition algorithm to obtain a final positioning result of the acoustic emission event. Local waveform spectrum envelope is used as a characteristic function, waveform generality of the same acoustic emission event distributed in different channels is amplified, noise interference can be overcome, and positioning precision is improved.

The invention discloses a micro-seismic signal classification method based on genetic algorithm optimization. The method comprises the following steps: data acquisition: acquiring waveform data containing multiple types of micro-seismic event forms; preprocessing: carrying out mean value removal and band-pass filtering preprocessing on the obtained waveform data, and removing baseline drift and high-frequency noise interference; envelope calculation: respectively calculating a time domain envelope and a frequency spectrum envelope for the preprocessed microseismic signal; feature extraction: extracting a plurality of statistics and time frequency parameters from the time domain envelope and the frequency spectrum envelope to form a feature complete set; feature selection and classifier optimization: utilizing a genetic algorithm to perform joint search optimization on structure parameters and feature subsets of the multi-layer perceptron; and training and verification: training the multilayer perceptron by using the optimized feature subset and the network structure, and evaluating the classification accuracy on a verification set. The method can realize high-precision classification, has robustness to noise, and can be widely applied to the fields of microseism monitoring, blasting event monitoring, other vibration source identification and the like.

The application discloses a bone conduction speech conversion method based on spectral envelope mapping, comprising the following steps: pre-processing and linear prediction analysis of the bone conduction speech signal, and calculating the LP filter coefficient; mapping the LSF coefficient of the air conduction speech signal corresponding to the bone conduction speech signal by using the trained neural network; controlling the minimum error of the original bone conduction speech signal and the synthesized air conduction speech signal according to the timbre weighting characteristic of the bone conduction speech characteristic; estimating the integer pitch of the bone conduction speech signal through the timbre weighting filter, obtaining the reference signal by passing the linear prediction residual signal through the timbre weighting filter, estimating the fractional pitch to obtain the adaptive codebook vector; obtaining the new reference signal by subtracting the adaptive codebook vector from the reference signal, searching for the optimal excitation in the fixed codebook; synthesizing the air conduction speech signal by combining the optimal excitation and the LP filter of the air conduction speech signal, and correcting the air conduction speech signal.

An apparatus for generating a bandwidth extended signal from an input signal includes a patch generator and a combiner. The input signal is represented for first and second bands by first and second resolution data, respectively, the second resolution being lower than the first. The patch generator generates first and second patches from the first band of the input signal according to first and second patching algorithms, respectively. A spectral density of the second patch generated using the second patching algorithm is higher than a spectral density of a first patch generated using the first patching algorithm. The combiner combines both patches and the first band of the input signal to obtain the bandwidth extended signal. The apparatus scales the input signal according to the first and second patching algorithms or scales the first and second patches, so that the bandwidth extended signal fulfills a spectral envelope criterion.

The invention provides a method and device for determining a fracturing crack based on a fracturing pump stop water hammer signal, and the method comprises the steps: firstly obtaining the fracturing pump stop water hammer signal of a target well, and determining a first cluster crack and the crack position of the first cluster crack through cepstrum analysis according to the fracturing pump stop water hammer signal; determining a plurality of candidate cracks according to the crack positions of the first cluster cracks; determining a plurality of actually opened fracturing fractures and fracture positions of the fracturing fractures from the plurality of candidate fractures according to the fracturing pump stop water hammer signal of the target well by determining and according to the related frequency spectrum envelope; according to the fracturing pump stop water hammer signal of the target well, determining a reflection coefficient of a fracturing crack by determining and according to a target reflection response sequence; and according to the reflection coefficient of the fracturing crack, determining the liquid inlet strength of the fracturing crack. Therefore, the number and the positions of the underground fracturing cracks can be accurately identified, and meanwhile, the liquid inlet strength of the underground fracturing cracks can be accurately determined.

An improved concept for coding sample values of a spectral envelope is obtained by combining spectrotemporal prediction on the one hand and context-based entropy coding the residuals, on the other hand, while particularly determining the context for a current sample value dependent on a measure of a deviation between a pair of already coded / decoded sample values of the spectral envelope in a spectrotemporal neighborhood of the current sample value. The combination of the spectrotemporal prediction on the one hand and the context-based entropy coding of the prediction residuals with selecting the context depending on the deviation measure on the other hand harmonizes with the nature of spectral envelopes.

An apparatus for generating a bandwidth extended signal from an input signal includes a patch generator and a combiner. The input signal is represented for first and second bands by first and second resolution data, respectively, the second resolution being lower than the first. The patch generator generates first and second patches from the first band of the input signal according to first and second patching algorithms, respectively. A spectral density of the second patch generated using the second patching algorithm is higher than a spectral density of a first patch generated using the first patching algorithm. The combiner combines both patches and the first band of the input signal to obtain the bandwidth extended signal. The apparatus scales the input signal according to the first and second patching algorithms or scales the first and second patches, so that the bandwidth extended signal fulfills a spectral envelope criterion.

This invention discloses a cross-receiver radio frequency fingerprinting method, system, and terminal device, relating to the field of wireless communication security technology. The invention estimates the carrier frequency offset of the received IQ signal by decoupling the transmitter and receiver carrier frequency offsets; extracts the slope and curvature of the frequency domain phase as domain-invariant features; constrains the consistency of features of the same device across different receivers based on contrastive learning; employs optimal transmission alignment of the spectral envelope distribution of different receivers to reduce inter-domain offset; and achieves cross-receiver device identification through joint optimization of invariant features. By estimating the carrier frequency offset and decoupling the transmitter and receiver frequency offsets, extracting the frequency domain phase slope and curvature as domain-invariant features, and combining contrastive learning to constrain the consistency of features of the same device across receivers with optimal transmission alignment of the spectral envelope distribution of different receivers, this invention achieves reliable device access authentication and identification under receiver-less conditions, improving the model's generalization ability and stability.

An apparatus for generating a bandwidth extended signal from an input signal includes a patch generator and a combiner. The input signal is represented for first and second bands by first and second resolution data, respectively, the second resolution being lower than the first. The patch generator generates first and second patches from the first band of the input signal according to first and second patching algorithms, respectively. A spectral density of the second patch generated using the second patching algorithm is higher than a spectral density of a first patch generated using the first patching algorithm. The combiner combines both patches and the first band of the input signal to obtain the bandwidth extended signal. The apparatus scales the input signal according to the first and second patching algorithms or scales the first and second patches, so that the bandwidth extended signal fulfills a spectral envelope criterion.

The invention is applicable to the technical field of optical biosensors, and provides an optical biosensor with a single ring-double ring mode switching function, which comprises a broadband light source, an optical signal is transmitted to an optical switch I, an optical sensing chip is respectively connected with the optical switch I and an optical switch II through an optical fiber array, and the optical switch II is connected with a spectrum analyzer. The optical fiber array is connected with the optical sensing chip and the optical switch and used for controlling switching and detection of a single-ring mode or a double-ring mode, when liquid to be detected causes high absorption of the sensing ring, the single-ring sensing mode is adopted, otherwise, the double-ring sensing mode is adopted, and when double-ring sensing is adopted, the single-ring sensing mode is adopted. The reference ring is tuned at the two ends of the thermode of the reference ring, the effective refractive index of the reference ring is changed, when the change of the refractive index is small, characterization of a to-be-measured substance is achieved through the spectral envelope drift distance, when the change of the refractive index is large, characterization of the to-be-measured substance is conducted through the change of FSR, and high-sensitivity and large-range sensing is achieved.

An improved concept for coding sample values of a spectral envelope is obtained by combining spectrotemporal prediction on the one hand and context-based entropy coding the residuals, on the other hand, while particularly determining the context for a current sample value dependent on a measure of a deviation between a pair of already coded / decoded sample values of the spectral envelope in a spectrotemporal neighborhood of the current sample value. The combination of the spectrotemporal prediction on the one hand and the context-based entropy coding of the prediction residuals with selecting the context depending on the deviation measure on the other hand harmonizes with the nature of spectral envelopes.

The systems and methods disclosed herein generate a personalized inner voice audio output that replicates or otherwise shares the acoustic characteristics of a speaker's self-perceived voice, which can differ from their externally perceived voice due to bone conduction effects. The systems and methods disclosed herein can generate a provisional voice clone (e.g., parameters of a voice model) of the speaker's voice (e.g., using a recording of the speaker's voice), and can apply a frequency shift to compensate for the absence of bone-conducted low-frequency adjustment that occurs during natural speech production. A trained artificial intelligence model predicts and applies one or more additional acoustic parameter adjustment values (e.g., formant structure, spectral envelope, prosodic patterns) to the provisional voice clone based on one or more factors (e.g., demographic, anatomical, content, environment). The provisional voice clone can be iteratively refined based on received user feedback (e.g., until the output aligns with the speaker's perception of their inner voice).

Systems and techniques are provided for processing audio data. For example, a process can include generating a first sub-vector of audio features corresponding to a first combination of spectral envelope features of the audio, energy features of the audio, and pitch features of the audio, and generating a second sub-vector of audio features corresponding to a second combination of the spectral envelope features, energy features, and pitch features. A first neural network-based autoencoder can be used to generate a first encoded representation corresponding to the first sub-vector of audio features. A second neural network-based autoencoder can be used to generate a second encoded representation corresponding to the second sub-vector of audio features. The first encoded representation and the second encoded representation can be transmitted to a decoder.

The present invention relates to a method for synthesising a speech signal, the method comprising the following steps: - obtaining a coded stream coded with mixed-excitation linear prediction coding; - extracting the parameters of the coded stream; - applying a neural network to extracted parameters in order to obtain a set of output parameters comprising a first set of gains and first spectral envelope parameters, wherein the gain and the spectral envelope are the gain and the spectral envelope of the speech signal to be synthesised in the 4 kHz - 8 kHz frequency band; - merging parameters derived from the set of output parameters and the extracted parameters of the coded stream; and - applying a mixed-excitation linear prediction synthesis technique to the parameters obtained after the merging step in order to obtain the speech signal to be synthesised.

The invention relates to the technical field of voice generation, in particular to a personification voice generation method and system for a carry-on AI partner, and the method comprises the following steps: collecting an external environment sound simulation signal through a carry-on equipment microphone array, obtaining an environment acoustic harmonic feature set, storing the environment acoustic harmonic feature set, and obtaining a voice simulation signal; and obtaining environment frequency spectrum characteristic data. According to the method, after a sound channel regular curled spectrum sequence is subjected to time domain smoothing to form frequency domain curled modulation data, a high-frequency-band spectrum envelope interval is extracted, amplitude gain improvement is executed according to a lip radiation impedance coefficient, then the frequency domain curled modulation data is superposed, and anthropomorphic interaction voice signals are generated. And high-frequency friction and gingival blasting details are enhanced without damaging the overall spectrum continuity, the environmental suitability, the anti-interference stability, the formant controllability and the high-frequency detail intelligibility, so that a chain gain is formed.

An apparatus for generating a bandwidth extended signal from an input signal includes a patch generator and a combiner. The input signal is represented for first and second bands by first and second resolution data, respectively, the second resolution being lower than the first. The patch generator generates first and second patches from the first band of the input signal according to first and second patching algorithms, respectively. A spectral density of the second patch generated using the second patching algorithm is higher than a spectral density of a first patch generated using the first patching algorithm. The combiner combines both patches and the first band of the input signal to obtain the bandwidth extended signal. The apparatus scales the input signal according to the first and second patching algorithms or scales the first and second patches, so that the bandwidth extended signal fulfills a spectral envelope criterion.

This application provides an audio comparison method, apparatus, device, computer-readable storage medium, and product. The method includes: acquiring a first audio channel and a second audio channel; sampling the first and second audio channels respectively to obtain a first frame signal set and a second frame signal set; determining the spectral envelope window of each frame signal in the second frame signal set based on the second frame signal set and a preset critical bandwidth; shifting the spectral envelope window of each frame signal in the second frame signal set to the spectral envelope window of the corresponding frame signal in the first frame signal set to obtain the audio similarity between each frame signal in the second frame signal set and the corresponding frame signal in the first frame signal set; and determining the audio comparison result of the first and second audio channels based on the audio similarity. This application's embodiments transform the tedious audio testing task into a fully automated comparison test, improving testing efficiency and accuracy.

An audio processing method includes: filtering an audio signal to obtain a low-frequency signal and a high-frequency signal; encoding the low-frequency signal to obtain a bitstream of the low-frequency signal; performing frequency domain transform on the low-frequency signal and the high-frequency signal respectively, to obtain a low-frequency spectrum and a high-frequency spectrum; performing spectral envelope extraction on the low-frequency spectrum and the high-frequency spectrum to obtain spectral envelope information, and performing spectral flatness extraction on the high-frequency spectrum to obtain spectral flatness information; and performing quantization encoding on the spectral flatness information and the spectral envelope information to obtain a bandwidth extension bitstream, and combining the bandwidth extension bitstream and the bitstream of low-frequency signal into an encoded bitstream.

The present application discloses an audio processing method, device and readable storage medium, the method comprising: obtaining initial audio, extracting the initial spectrum of the initial audio and the spectrum envelope corresponding to the timbre; performing scaling processing on the spectrum envelope to obtain a target spectrum envelope; using the target spectrum envelope and the spectrum envelope to obtain a weight coefficient; using the weight coefficient, processing the initial spectrum to obtain a target spectrum; performing inverse Fourier transform on the target spectrum to obtain a target audio after the timbre is changed. In the present application, since scaling processing is performed on the spectrum envelope corresponding to the timbre will not cause synchronous changes in pitch, the target audio obtained after the timbre change processing will not have the problem of synchronous changes in pitch. That is, the difference between the target audio after the timbre change and the initial audio is only a change in timbre, but no change in pitch, so the target audio will be more realistic.

An echo suppressing device includes: an echo canceller that suppresses a linear echo signal from an input signal acquired by a microphone; a spectrum envelope extraction unit that extracts spectrum envelope information from a reception signal to be output to a loudspeaker; a nonlinear echo estimation unit that estimates spectrum envelope information of a nonlinear echo signal included in the input signal from the spectrum envelope information extracted from the reception signal; and a nonlinear echo suppression unit that suppresses the nonlinear echo signal from an output signal of the echo canceller by using the estimated spectrum envelope information of the nonlinear echo signal.

The present document relates an audio encoding and decoding system (referred to as an audio codec system). In some embodiments, a method of audio signal encoding comprises: receiving an input audio signal; transforming a sequence of samples of the input audio signal into a block of transform coefficients, the transform coefficients indicative of the spectral energy of the block; estimating a spectral envelope of the block from the transform coefficients; adjusting the transform coefficients using the spectral envelope and a spectral shaper, the spectral shaper including one or more parameters indicative of a fundamental frequency of a multi-sinusoidal signal model, where the fundamental frequency corresponds to a time domain delay; and entropy coding the adjusted transform coefficients.

This invention discloses a flat multi-wavelength vector vortex fiber laser, which utilizes a dual-injection recirculating frequency-shifting fiber cavity and a few-mode long-period fiber grating to achieve the generation of a flat multi-wavelength vector vortex beam. The laser mainly includes an injection seed source, a pump source, a wavelength division multiplexer, an erbium-doped fiber, an RF drive signal, a phase modulator, a few-mode long-period fiber grating, a partial mirror, a fiber delay unit, a fiber circulator, a polarization controller, and a fiber coupler. This laser uses a recirculating frequency-shifting fiber cavity to suppress mode competition in the erbium-doped fiber, achieving multi-wavelength oscillation. By injecting two seed lights externally, it generates complementary triangular spectral envelopes, achieving a flat multi-wavelength vector vortex beam output between the injected wavelengths. Since this laser does not use an interferometric filter as a multi-wavelength oscillation device, it exhibits excellent time stability.

Embodiments of the present invention provide a linear prediction-based noise signal processing method, a linear prediction-based noise signal generation method, an encoder, a decoder, and an encoding and decoding system. The noise signal processing method according to the embodiments of the present invention 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 invention, 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.

This application discloses an audio signal processing method and apparatus, and belongs to the field of audio signal processing technologies. The method includes: obtaining a plurality of sub-bands of an audio signal and a scale factor of each sub-band; determining, based on the scale factors of the plurality of sub-bands, a reference value used for shaping a spectral envelope of the audio signal; and shaping the spectral envelope of the audio signal by using the reference value as a baseline, to obtain an adjustment factor of each sub-band corresponding to a shaped spectral envelope. The adjustment factor is used to quantize a spectral value of the audio signal, and / or the adjustment factor is used to dequantize a code value of the spectral value. In this application, compression efficiency of encoding an audio signal is improved while sound quality is ensured.

The invention discloses an electromagnetic compatibility intelligent detection system and method based on an intelligent sensor, and relates to the technical field of electromagnetic compatibility detection, and the system comprises an input source collection module, a prediction analysis module and an interference recognition module. The method is technically characterized by comprising the following steps: acquiring the environment temperature of a to-be-detected area, and synchronously acquiring historical calibration data of an active antenna; wherein the historical calibration data at least comprises an electromagnetic near-field distribution map and a spectrum envelope feature; based on the historical calibration data, establishing a logic regression model to predict a drift rule of the active antenna along with the environment temperature, and obtaining a gain change curve; a one-dimensional convolution analysis electromagnetic signal is introduced to identify an electromagnetic interference event, a built-in rule engine is used for executing judgment analysis on measurement gain deviation of each batch, and an adjustment strategy is triggered through the electromagnetic interference event; according to the invention, intelligent detection of electromagnetic compatibility is realized, real-time calibration and correction are carried out in the detection process, and the detection precision is improved.

This application belongs to the field of radio electronics and signal processing technology, and specifically relates to a method and system for detecting signals in the frequency domain in a colored noise environment based on spectral envelope extraction, including a signal acquisition unit, a signal conversion unit, a signal judgment unit, and a signal output unit; The present invention is a research summary of an algorithm for detecting signals in the frequency domain in a colored noise environment based on spectral envelope extraction. This algorithm is used for blind signal detection under broadband reception with an uneven noise spectrum and has good robustness. It can be used in low signal-to-noise ratio scenarios and has a high signal detection accuracy.