23 results about "Spectral envelope" patented technology

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 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.

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.

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).

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 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.

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.

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.

The invention provides a foreign language learning loud-reading quality analysis system and method based on voice interaction, and the method comprises the steps: collecting a loud-reading audio signal of a learner, extracting a fundamental frequency contour and a spectrum envelope feature through a Mel-frequency cepstral coefficient algorithm, and obtaining a preliminary voice feature set; aligning duration information and energy distribution of the learner audio and the standard reference audio by adopting a dynamic time warping algorithm according to the preliminary voice feature set, and determining an aligned deviation vector; processing the combination of rhythm adjustment indexes and the preliminary voice feature set by adopting a long-short-term memory network to obtain a standardized feature vector representation form; aiming at the standardized feature vector representation form, through calculating the Euclidean distance between the standardized feature vector representation form and a standard pronunciation vector, determining a cross-language comparison difference degree; and generating an adaptive improved sequence according to the targeted feedback data to obtain a final reading quality analysis result. According to the method, through fusion of rhythm adjustment and cross-language difference analysis, the accuracy and personalized improvement effect of reading quality evaluation are remarkably improved, and efficient technical support is provided for language learning.

The invention relates to the technical field of voice processing, and particularly discloses a PTE spoken language question and answer evaluation method and system based on intelligent voice, and the method comprises the steps: obtaining an original voice signal, and extracting an acoustic feature sequence; calculating an instantaneous speech speed through phase-space reconstruction and a recursion rate matrix, and generating a time length regularity factor when the instantaneous speech speed exceeds a threshold value; determining a fractional Fourier transform order according to the factor, calculating a cosine distance matrix weighted by the instantaneous frequency deviation, and dynamically planning and backtracking to obtain a regular feature sequence; performing multi-fractal detrending fluctuation analysis to position a singular fluctuation candidate section, judging a fuzzy phoneme section by combining a certainty percentage and a laminar flow percentage of a recurrence plot, and extracting a phoneme label of an adjacent stable section as a reconstruction condition; performing phase reconstruction on the clear feature sequence, calculating a distance between a phase cross-correlation peak value and a spectral envelope Riemann geodesic line to obtain a pronunciation confidence coefficient, and performing weighted summation on the pronunciation confidence coefficient and a fluency parameter to obtain an evaluation score; according to the method, the problem of score distortion caused by duration mismatch and continuous reading boundary fuzziness at the supernormal speech speed is solved.

An approximation-free and iteration-free method for spectral analysis of an intracavity electro-optic modulation type optical frequency comb, includes: calculating a residual phase delay of a single propagation of laser in a resonant cavity, analyzing outgoing transmission characteristics of a light source of the intracavity electro-optic modulation type optical frequency comb, accumulating laser electric field intensities corresponding to all cyclic propagation times n to obtain an outgoing laser electric field intensity E, obtaining a new approximate-free outgoing laser electric field intensity E′ of the intracavity electro-optic modulation type optical frequency comb, obtaining an outgoing laser electric field intensity Ek′ of kth-order comb teeth, calculating an outgoing laser light intensity Ik of the kth-order comb teeth and accurately analyzing a spectrum of the intracavity electro-optic modulation type optical frequency comb, determining a working state according to a simulated spectral envelope curve, and guiding the subsequent optimization design and debugging.

The invention discloses a regional event detection method based on a distributed acoustic sensing system, and belongs to the technical field of event detection. According to the invention, a self-adaptive multi-scale threshold is adopted in the process of processing an initial event detection fusion signal to obtain a final event fusion signal, and a wavelet coefficient is corrected by using the self-adaptive multi-scale threshold; the corrected wavelet coefficient is used for reconstructing the signal to obtain a final event fusion signal, so that residual noise can be further eliminated, fine signal features possibly lost due to threshold processing can be recovered, and the overall quality and robustness of signal reconstruction are remarkably improved. According to the method, tuning factors are added in the feature extraction process, and the dynamic characteristics of signals in time and frequency can be deeply mined; and meanwhile, a third-order discrete cosine transform coefficient is introduced as a compensation item for feature extraction, so that more complex and finer fluctuating changes in a frequency spectrum envelope are fitted more accurately, and feature information with higher distinction degree is provided for a subsequent classification task.

The invention discloses an optical frequency comb generating device based on a thin-film lithium tantalate grating FP (Fabry-Perot) microcavity, which is characterized by comprising a pumping unit, a microwave modulation unit and a normal dispersion grating FP resonant cavity unit. A low-power / high-power single-frequency laser pump enters an input coupling film lithium tantalate waveguide, and then is coupled to a normal dispersion grating FP resonant cavity unit through an Euler bending-based S-shaped waveguide and a small-perimeter annular micro-cavity, effective excitation of a fundamental mode is ensured based on Euler bending, a high-order mode is suppressed, a sideband is generated through electro-optical modulation, and a high-frequency laser is generated. Multi-frequency laser is generated, spectrum broadening and spectrum envelope planarization of the optical frequency comb are achieved through the combined effect of normal dispersion, the electro-optical modulation effect, the third-order nonlinear effect and high-reflectivity broadband apodized grating filtering, and finally the generated optical frequency comb is output through the output coupling waveguide. Complex mode cross coupling regulation and control are avoided, and generation of a high-efficiency optical frequency comb with a relatively flat normal dispersion area is realized.

The embodiment of the invention provides a frequency spectrum control method, circuit, chip and system for an ultrasonic signal. The method comprises the following steps: generating a frequency spectrum envelope corresponding to a frequency spectrum envelope parameter according to the frequency spectrum envelope parameter; according to a preset frequency point selection strategy, sequentially selecting a plurality of resident frequency points between a starting frequency point and a cut-off frequency point of the spectrum envelope to obtain a resident frequency point sequence; and determining the residence period of each residence frequency point in the residence frequency point sequence according to the total number of the residence periods and the spectrum envelope, and obtaining a frequency hopping pattern composed of the residence frequency point sequence and the residence periods. In the embodiment of the invention, only a small amount of frequency spectrum envelope parameters need to be stored, the recovery of the frequency spectrum envelope is realized through the frequency spectrum envelope parameters during use, and the signal with the corresponding frequency spectrum envelope shape can be generated based on the corresponding resident frequency point and resident period selection strategy. Therefore, storage resources are saved, flexible spectrum control can be realized, and the performance and the application range of the ultrasonic detection system are further improved.

This application discloses an AI-based image identification and feature extraction system for mine hardware faults, relating to the field of power equipment fault diagnosis technology. The proposed solution includes: forming a multimodal feature tensor by tensor concatenating the time-frequency matrix and temperature gradient field, and then using this tensor to weightedly modulate the spectral envelope feature vector to generate a modulated feature vector. This solves the problem that a fixed discrimination threshold cannot adapt to the nonlinear changes of multimodal features, achieving dynamic fusion and enhancement of multi-physics field features. Furthermore, by calculating the confidence score of candidate aging segments using the modulated feature vector, low-score regions are filtered out. The confidence score is then multiplied and fused with a spatial attention map to generate a refined activation map. Connected regions are extracted through adaptive threshold segmentation, and the coordinates of the aging segments are output. This solves the problem of coarse localization in traditional methods, achieving precise localization of aging areas and reducing unnecessary maintenance operations.

This invention discloses a spectral envelope shaping device and method based on an adjustable mask array. The device includes a light source to be shaped, a collimation unit, a first dispersion unit, a first imaging unit, an adjustable mask array disposed on the spectral modulation surface or its conjugate surface, a spectral reconstruction unit, and a coupling output unit; it may also include a spectroscopic monitoring unit and a closed-loop control unit. After collimation, dispersion, and imaging, the light to be shaped forms a wavelength spatial distribution along the dispersion direction on the spectral modulation surface. The adjustable mask array independently blocks the beam in the corresponding target wavelength range or band through multiple independent blocking units, achieving selective continuous attenuation of energy in each band by changing the blocking area or blocking ratio. The modulated beam is then reconstructed and output after inverse dispersion compensation or equivalent inverse transformation by the spectral reconstruction unit. This invention is applicable to scenarios such as broadband spectral flattening, multi-passband selective output, and attenuation-type spectral envelope shaping of spectrally calibrated light sources.

This invention discloses an intelligent electromagnetic compatibility (EMC) detection system and method based on smart sensors, belonging to the field of EMC detection technology. The system includes an input source acquisition module, a predictive analysis module, and an interference identification module. Its key technical points are: acquiring the ambient temperature of the area to be tested and simultaneously obtaining historical calibration data of the active antenna; wherein the historical calibration data includes at least an electromagnetic near-field distribution map and spectral envelope characteristics; based on the historical calibration data, establishing a logistic regression model to predict the drift law of the active antenna with ambient temperature and obtaining the gain change curve; introducing one-dimensional convolutional analysis of electromagnetic signals to identify electromagnetic interference events, using a built-in rule engine to perform judgment analysis on the gain deviation of each batch of measurements, and triggering adjustment strategies through electromagnetic interference events; this aspect achieves intelligent EMC detection, with real-time calibration and correction during the detection process, improving detection accuracy.

A technology of accurately coding and decoding coefficients which are convertible into linear prediction coefficients even for a frame in which the spectrum variation is great while suppressing an increase in the code amount as a whole is provided. A coding device includes: a first coding unit that obtains a first code by coding coefficients which are convertible into linear prediction coefficients of more than one order; and a second coding unit that obtains a second code by coding at least quantization errors of the first coding unit if (A−1) an index Q commensurate with how high the peak-to-valley height of a spectral envelope is, the spectral envelope corresponding to the coefficients which are convertible into the linear prediction coefficients of more than one order, is larger than or equal to a predetermined threshold value Th1 and / or (B−1) an index Q′ commensurate with how short the peak-to-valley height of the spectral envelope is, is smaller than or equal to a predetermined threshold value Th1′.

The application relates to the technical field of speech processing, and particularly discloses a PTE oral language question and answer evaluation method and system based on intelligent speech, which comprises the following steps: acquiring an original speech signal and extracting an acoustic feature sequence; calculating an instantaneous speech speed through phase space reconstruction and a recursive rate matrix, generating a time length regularization factor when the value exceeds a threshold; determining a fractional Fourier transform order according to the factor, calculating an instantaneous frequency deviation weighted cosine distance matrix, and dynamically programming backtracking to obtain a regularized feature sequence; performing multiple fractal detrended fluctuation analysis to locate a singular fluctuation candidate segment, determining an ambiguous phoneme segment in combination with a recursive graph certainty percentage and a laminar flow percentage, and extracting adjacent stable segment phoneme labels as reconstruction conditions; performing phase reconstruction on a clear feature sequence, calculating a phase mutual correlation peak value and a spectral envelope Riemann geodesic distance, obtaining pronunciation confidence, and weighting and summing fluency parameters to obtain an evaluation score; and the application solves the score distortion problem caused by time length mismatch and unclear read-through boundaries under an abnormal speech speed.

An embodiment according to the present invention includes an audio decoder configured to decode, from a data stream, a quantized spectrum, a spectral envelope representation based on linear prediction coefficients, and a fundamental frequency related parameter for a predetermined frame of successive frames. The decoder is configured to determine a spectral shaping function from a spectral envelope representation based on linear prediction coefficients using a first manner below a fundamental tone frequency determined from the fundamental frequency related parameters and a second manner above the fundamental tone frequency, to spectrally shape the quantized spectrum using the spectral shaping function to obtain a dequantized spectrum, and to decode the dequantized spectrum using the dequantized spectrum. And reconstructing the predetermined frame using the dequantized spectrum. Furthermore, the audio decoder is configured such that, if the pitch frequency is spectrally higher than a predetermined spectral position, the spectral shaping function is lower at the predetermined spectral position than if the pitch frequency is spectrally lower than the predetermined spectral position. A decoder, an encoder, and a method are also disclosed.

The present invention relates to a method for performing high frequency reconstruction of an audio signal, the method comprising: receiving an encoded audio bitstream, the encoded audio bitstream including audio data representing a lowband portion of the audio signal and high frequency reconstruction metadata; decoding the audio data to generate a decoded lowband audio signal; extracting from the encoded audio bitstream the high frequency reconstruction metadata, the high frequency reconstruction metadata including operating parameters for a high frequency reconstruction process, the operating parameters including a patching mode parameter located in a backward-compatible extension container of the encoded audio bitstream, wherein a first value of the patching mode parameter indicates spectral translation and a second value of the patching mode parameter indicates harmonic transposition by phase-vocoder frequency spreading, and wherein the backwardcompatible extension container further includes a flag indicating whether additional preprocessing is used to avoid discontinuities in a shape of a spectral envelope of the highband portion when the patching mode parameter equals the first value; filtering the decoded lowband audio signal to generate a filtered lowband audio signal; regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata, wherein the regenerating the highband portion of the audio signal includes spectral translation if the patching mode parameter is the first value and the regenerating the highband portion of the audio signal includes harmonic transposition by phase-vocoder frequency spreading if the patching mode parameter is the second value.

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.

For spectral Doppler imaging, the search range for tracking the spectral envelope is dynamically set. The limits of the search envelope are established by a spectral-by-spectral placement between frequency bands. This search can be aided by plotting the spectrum from 0 to 2π. The limits vary over time to better separate the frequency bands, thus avoiding aliasing for subsequent tracking.