23 results about "Effective frequency" patented technology

The invention discloses a low-delay self-adaptive audio transmission method, system and device and a storage medium, and relates to the technical field of audio, and the method comprises the steps: dynamically recognizing an optimal transmission path in wireless dual bands based on an intelligent band detection and self-adaptive channel selection mechanism; decomposing the target audio data through a hierarchical coding algorithm, and executing self-adaptive adjustment of a differential transmission strategy and a quantization parameter; the dual-band performance is monitored in real time based on a load balancing algorithm, the decomposed transmission tasks are dynamically allocated, and path switching is triggered when it is monitored that the transmission quality of the current band is abnormal; the future buffer requirement is predicted based on a predictive buffer algorithm, the size of a buffer area is dynamically adjusted according to the actual playing condition and the network state change so as to ensure the playing continuity and minimize the delay, the technical bottleneck of traditional single-frequency wireless audio transmission is broken through, an effective frequency band coordination mechanism is realized, resource waste or transmission conflicts are avoided, and the transmission efficiency is improved. And the double-frequency advantage is fully exerted.

The invention relates to the technical field of sound correction, and discloses a room sound calibration method based on listening tendency control and related equipment, which are used for improving the matching degree of a calibration result and listening preference of a user. The room sound calibration method based on listening tendency control comprises the following steps: obtaining a listening tendency parameter selected by a user through an interactive interface and a room pulse response of a current room; determining a group of frequency-related window control parameters according to the hearing tendency parameter and the room pulse response; determining a windowing strategy with the window length changing along with the frequency based on the window control parameters; windowing the room impulse response by using a windowing strategy to extract an effective frequency response curve for acoustic calibration; calculating according to the difference between the effective frequency response curve and the target frequency response curve to obtain a correction filter coefficient; and calibrating the audio signal of the audio playing system based on the correction filter coefficient.

The embodiment of the invention provides a sound box and electronic equipment, and relates to the technical field of sound boxes. The sound box comprises a box body, the box body comprises a first sound outlet hole, a first sound cavity, a through hole, a second sound cavity and a second sound outlet hole, the first sound outlet hole is communicated with the first sound cavity, and the second sound outlet hole is communicated with the second sound cavity; the loudspeaker is mounted in the through hole and separates the first sound cavity from the second sound cavity; the first resonance part is arranged in one of the first sound cavity and the second sound cavity, so that one of the first sound cavity and the second sound cavity has a first first-order sound mode frequency, and the first resonance part is of an acoustic superstructure; and the second resonance part is arranged in the other one of the first sound cavity and the second sound cavity, so that the other one of the first sound cavity and the second sound cavity has a second first-order sound mode frequency different from the first first-order sound mode frequency. According to the sound box and the electronic equipment, the low-frequency diving depth of the sound box can be reduced, and the effective frequency band of the sound box is widened.

The application discloses a method for realizing a headphone transparent mode. The prior art method needs to use a device other than the headphone to complete required path measurement, and requires complicated hardware resources, environment and process. The method uses an external microphone and an error microphone of the headphone to complete path measurement of a primary channel, uses a loudspeaker of the headphone and the error microphone to complete path measurement of a secondary channel, establishes a first-stage filter of the transparent mode based on an N-order FIR filter, establishes a cost function, and uses a genetic algorithm to solve coefficients of the FIR filter; a high shelf filter is used to establish a second-stage filter of the transparent mode, and to complete suppression of potential abnormal high-frequency components. The two-stage filters jointly establish a complete transparent mode filter. The application uses conditions of the headphone itself to complete necessary path measurement, concentrates the FIR filter on fitting of effective frequency bands in environmental human voices, and reduces difficulty and complexity of required resources of the transparent mode design.

The present invention discloses an adaptive underwater sound source passive localization method and system. The method includes: acquiring the signals received by a hydrophone array; converting the signals received by the array to the frequency domain by using fast Fourier transform, obtaining the effective frequency band of the target signal source through spectrum analysis, and constructing a steering vector estimation; calculating the covariance matrix of the signals received by the array, and obtaining the eigenvalues and eigenvectors of the covariance matrix through eigenvalue decomposition; constructing a weight matrix by using the eigenvalues of the covariance matrix, weighting all the eigenvalues of the covariance matrix with the weight matrix to form a fully weighted spatial vector, and realizing adaptive fully weighted MUSIC spatial spectrum estimation through iterative operation and setting of iterative termination conditions; and obtaining the estimated value of the arrival azimuth. The present invention avoids the need for estimating the number of signal sources in the conventional MUSIC algorithm, and solves the situation of performance degradation of the beamformer caused by steering vector mismatch and noise subspace division error in practical applications.

The utility model discloses a distributed active and passive diaphragm matrix transmission ultrathin picture frame sound box, comprising a wooden box body, the outside of which is provided with a metal frame; a full-frequency unit, a first passive vibrating diaphragm group and a second passive vibrating diaphragm group are sequentially arranged on the wooden box body from top to bottom; the high-pitch unit is arranged in the middle of the full-frequency unit; the front surface of the wooden box body is provided with sound transmission screen cloth; according to the sound box, a unique active and passive vibrating diaphragm transmission technology is adopted, a matrix structure is applied to carry out omnibearing harmonic resonance distribution, audio signals of which the input end is subjected to DSP sound effect processing are reasonably adapted and smoothly transited in each layer subarea in the sound box, and harmonic resonance and mixing in an effective frequency response range are carried out, so that the sound effect of the sound box is effectively improved. And the audio signal fully shows the characteristics and fusion of the high-fidelity sound effect in the transmission process.

This invention discloses an automatic identification method for the initial arrival of vibration events based on multi-module filter integration. The main steps are as follows: First, the continuously recorded raw signal is preprocessed and then segmented. For each segmented signal, the effective frequency distribution of vibration energy is determined. Then, operations are sequentially performed, including embedding an anti-aliasing low-pass filter, downsampling the signal, embedding an adaptive detrending term filter, embedding a median filter, and embedding a borderless delay filter. Next, the signal envelope energy is calculated based on the borderless delay filter. Finally, a discriminant filter is embedded, and the signal is normalized to obtain the final identification feature function. This invention establishes an automatic picking algorithm integrated with multiple filters. While reducing noise, it constructs a feature function, requiring only the most basic signal parameters and effectively reducing noise interference. This improves the accuracy of automatically picking the occurrence of disaster events.

The invention discloses a high-quality bird sound rapid screening method based on acoustic index fusion judgment, and the method comprises the steps: automatically loading continuously collected audio files in a preset date range, analyzing the collection time information, carrying out the time sorting, carrying out the resampling and channel unified processing of audio data, and carrying out the collection of the collection time information; segmenting the audio signal according to natural minutes and completing time marking; estimating background noise power and calculating a noise reduction acoustic complexity index, and selecting audio segments in a preset proportion range as a candidate segment set; acoustic diversity indexes of the candidate segments are calculated respectively, and the number of effective frequency bands is extracted; and constructing a dynamic discrimination threshold, and obtaining a final high-quality bird sound fragment screening result through multi-condition joint judgment of ADI and FADI. The method has the advantages of being high in calculation efficiency and high in noise interference robustness, is suitable for automatic analysis and processing of large-scale field collected data, and can effectively improve the quality and availability of front-end acoustic data.

The invention belongs to the technical field of electromagnetic wave absorbing materials, and relates to a Fe-MOF-derived MIL-88A (Fe) / rGO composite aerogel wave-absorbing material and a preparation method of the Fe-MOF-derived MIL-88A (Fe) / rGO composite aerogel wave-absorbing material. The composite material is composed of an MIL-88A (Fe) MOF carbon-based derivative, carboxymethyl cellulose and graphite oxide. The preparation is prepared through a solvothermal method, water bath heating, freeze drying and other processes. The dielectric loss and magnetic loss of the sample are regulated and controlled by regulating the temperature of the MIL-88A (Fe) subjected to high-temperature annealing and the ratio of the annealed MIL-88A (Fe) to the graphite oxide, so that the dielectric loss capability, interface polarization and relaxation loss of the composite material are improved, the microwave absorption performance is improved, and the width of an effective frequency band is increased.

The invention relates to a compressed sampling system suitable for high dynamic range broadband signals and a rapid reconstruction method, and the method is used for processing aero-engine blade tip vibration monitoring signals. Based on the signal Nyquist rate fNYQ, the bandwidth B, the effective frequency band number K and the maximum amplitude Amax, hardware parameters of a modulation broadband converter MWC and a threshold value of a self-recovery analog-to-digital converter SR-ADC are set in a priori mode; performing multi-channel frequency band demodulation, low-pass filtering and analog-to-digital sampling on the signal by using a modulation broadband converter and a dual-threshold self-recovery analog-to-digital converter to obtain the output of each channel; correct amplitude information is recovered from the collected signals according to the dual-threshold information, and continuous signals are constructed to a finite field conversion module; adaptively solving a spectrum support set by using variational Bayesian inference, energy aggregation, FDR saliency detection and a BIC model reduction algorithm; and performing up-sampling and frequency band modulation on the low-rate sample sequence based on the frequency spectrum support set, and reconstructing a high-dynamic range time domain signal meeting the Nyquist rate and a complete frequency spectrum located in an original frequency band.

The application relates to the technical field of decision support, and discloses a bidding whole-process automation and decision support system based on multi-agent cooperation. First, the updating intensity sequence and global risk score data of the demand extraction side and the review and early warning side are collected, and the instantaneous phase and effective frequency are extracted through signal analysis. Then, phase relationship and frequency ratio analysis are carried out, the frequency mismatch and coupling strength are calculated, and whether the system is in a locked state is determined. When the locking effect is detected, the real risk signal is extracted through the unlocking residual error module to eliminate the false fluctuations caused by the model rhythm difference. Then, the residual error signal quality is evaluated by using a signal-to-noise ratio calculation module, and a document-level risk score is comprehensively output in combination with a mapping function, so that accurate identification and intelligent decision support of the bidding document risk are realized, and the accuracy and stability of risk prediction are significantly improved.

The invention discloses a resolution improving method and system based on audio frequency spectrum analysis, and the method comprises the steps: enabling a pressure sensor to convert the vibration of a mute drum leather into a voltage signal, outputting the voltage signal, and converting the voltage signal into a current signal through a coupling circuit; constructing a consensus frequency component template based on the current signal, and filtering the frequency spectrum of the audio signal based on the consensus frequency component template; and smoothing the de-noised audio signal spectrum according to the time frame, and then carrying out inverse discrete Fourier transform to obtain a reconstructed audio signal with clear tone. According to the invention, by collecting multiple coherent current signals, the distinguishing and maintaining capability of each frequency component is remarkably improved, and effective frequency components can be reserved to the greatest extent while noise is suppressed; furthermore, signal fluctuation and distortion are reduced through signal change processing in the time dimension, audio feedback with high fidelity and high definition is finally generated, and more accurate and reliable practice reference is provided for a user.

The application relates to an audio processing method, a computer device and a storage medium. By determining an effective frequency band of a target user, an effective frequency band part and a target frequency band part to be processed of target audio are obtained according to the effective frequency band, a target part audio signal corresponding to the target frequency band part is obtained, the target part audio signal is down-regulated to obtain a down-regulated target part audio signal, and an audio signal corresponding to the effective frequency band part of the target audio and the target part audio signal are synthesized to obtain target audio suitable for the target user. Compared with a traditional audio playing mode, the scheme combines the effective frequency band that can be heard by the target user, down-regulates the high-frequency part of the audio, and realizes the effect that the hearing-impaired user can hear the complete signal of the audio.

The utility model relates to a kind of bone conduction sound production device and bone conduction earphone, bone conduction sound production device includes support, vibrator assembly and stator assembly.Vibrator assembly is movably arranged in the inside of support, vibrator assembly includes magnetic circuit assembly, and the at least outer edge of the both ends of magnetic circuit assembly along first direction is formed first magnetic pole part and second magnetic pole part respectively, and the polarity of first magnetic pole part and second magnetic pole part is opposite;Stator assembly is fixedly arranged in the inside of support, and stator assembly includes first coil and second coil, and the current direction of first coil and second coil is opposite, and first magnetic guide support is arranged between first coil and second coil;First coil is encircled in the outside of first magnetic pole part, and second coil is encircled in the outside of second magnetic pole part;The utility model greatly improves the vibration intensity of vibrator assembly, so that the sensitivity of bone conduction sound production device in effective frequency domain range is effectively improved, thereby the loudness of bone conduction earphone is improved.

The invention provides an animal sound source positioning method and electronic equipment, and the method comprises the steps: inputting a collected sound signal into a deep learning model, and judging whether the sound signal contains animal sound or not; according to the method, the effective frequency band range is determined in the sound signals containing the animal sound, and the sound signals in the effective frequency band range are determined as the target sound signals, so that the animal sound can be received in a high-definition manner, the direction and distance of an animal sound source can be identified in a high-precision manner, and the sound of a high-frequency sound animal can be better received; animal sound can be effectively captured and positioned; according to the method, the target sound signal is optimized according to the sound optimization algorithm to obtain the optimized sound signal, and noise and echo in the optimized target sound signal are optimized through the sound optimization algorithm, so that the animal sound can be clearly captured, the first target sound source position of the animal sound source in the optimized sound signal is determined, background noise can be suppressed, and the sound quality of the animal sound is improved. The captured sound signals are clear and pure, and the direction and distance of the animal are finally determined.

The utility model discloses a sound box housing capable of eliminating interference. The sound box housing comprises a left chamber 1, a middle chamber and a right chamber, inverter tubes are arranged on the partition plates of the left cavity and the middle cavity in a penetrating mode, and inverter tubes are arranged on the partition plates of the right cavity and the middle cavity in a penetrating mode. And the two inverter tubes are arranged in a staggered manner in the axial direction. Therefore, the left chamber and the right chamber are provided with the inverter tubes, and voice coils with different effective frequency ranges are adapted by adjusting the probing depths of the inverter tubes, so that a series of loudspeaker box products adopting the same loudspeaker box shell can obtain good tone quality performance, special design or debugging for the loudspeaker box shell is not needed, the universality is greatly improved, and the cost is reduced. And the production and design cost is reduced.

The present invention belongs to the field of new energy technology utilization, and provides a floating double-chamber oscillating water column wave energy power generation device equipped with an external open structure, including an open structure, an oscillating water column system, and a mooring and fixing system. The present invention effectively combines a floating double-chamber oscillating water column wave energy conversion device and an open structure, which can not only improve the energy conversion efficiency of the device and broaden the effective frequency band width of the device; at the same time, the open structure can also dissipate a certain amount of energy from the waves, especially short waves, reduce the wave loads received by the device, and improve the survivability of the device.

The invention relates to the technical field of decision support, and discloses a bidding full-process automation and decision support system based on multi-agent cooperation, and the system comprises the steps: firstly collecting the update intensity sequences and global risk scoring data of a demand extraction side and an examination early warning side, and extracting an instantaneous phase and an effective frequency through signal analysis; and then analyzing a phase relation and a frequency ratio, and calculating frequency mismatch and coupling strength so as to judge whether the system is in a locked state or not. When a locking effect is detected, a real risk signal is extracted through a locking residual module, and false fluctuation caused by model rhythm difference is eliminated; and a signal-to-noise ratio calculation module is utilized to evaluate the quality of the residual signal, and a mapping function is combined to comprehensively output a document-level risk score, so that accurate identification and intelligent decision support of bidding document risks are realized, and the accuracy and stability of risk prediction are remarkably improved.

The invention provides a preparation process of an electromagnetic protective coating, and relates to the field of preparation of electromagnetic protective coatings, and the preparation process is characterized by comprising the following steps: step 1, preparing CIP coated Fe3O4 binary core-shell composite wave-absorbing functional particles: by adopting a wet chemical oxidation method, putting degreased CIP (carbonyl iron powder) into an oxidation solution, and carrying out ultrasonic treatment to obtain the CIP coated Fe3O4 binary core-shell composite wave-absorbing functional particles; and oxidizing the iron on the surface to generate compact magnetic ferroferric oxide, so as to obtain the CIP-Fe3O4 binary core-shell composite wave-absorbing functional particle. The method has the advantages that the CIP-Fe3O4 core-shell structure prepared by a wet chemical oxidation method effectively regulates the conductivity of the carbonyl iron powder, enhances the chemical stability and reduces electromagnetic wave reflection; an RGO bridging process is introduced to be combined with a magnetic loss and dielectric loss mechanism, impedance matching is optimized, the wave absorbing performance is greatly improved, and the effective frequency band is widened; by adopting an environment-friendly polyurethane matrix and an innovative process, the problems of particle agglomeration and foaming are solved, and the uniformity and stability of the coating are ensured. The material has the advantages of light weight, broadband absorption, strong durability and the like.

The application discloses a resolution improvement method and system based on audio spectrum analysis, and the method comprises the following steps: a pressure sensor converts the vibration of a mute drumhead into an electric voltage signal output, and converts the electric voltage signal into an electric current signal through a coupling circuit; a consensus frequency component template is constructed based on the electric current signal, and the spectrum of an audio signal is filtered based on the consensus frequency component template; after the spectrum of the denoised audio signal is smoothed according to time frames, inverse discrete Fourier transform is performed to obtain a reconstructed audio signal with clear tone color. The application significantly improves the resolution and retention capability of each frequency component by collecting multiple coherent current signals, can maximize the retention of effective frequency components while suppressing noise, further processes the signal change in the time dimension to reduce signal fluctuation and distortion, and finally generates an audio feedback with high fidelity and high definition, thereby providing more accurate and reliable practice reference for users.

The invention discloses a closed-loop adaptive compensation method for loudspeaker production line calibration, and relates to the technical field of testing devices. The method comprises the steps of S1, task initialization and template loading, S2, continuous excitation playing and synchronous recovery, S3, effective frequency point screening and confidence coefficient generation, S4, actual frequency response calculation and confidence weighted error construction, S5, envelope and residual error double-component compensation updating, and S6, smooth constraint and channel consistency constraint compensation reconstruction. S7, double-buffering hot switching and S8, multi-factor freezing judgment and result output are carried out, and through effective frequency point screening, confidence weighted error construction, envelope and residual error double-component compensation updating, compensation reconstruction and the like, good technical effects are achieved in the aspects of low-credibility frequency point suppression, frequency response flatness improvement, convergence speed increase and freezing stability enhancement.

The present application relates to the technical field of sound calibration, and discloses a room sound calibration method based on hearing tendency control and related equipment, which is used for improving the matching degree of calibration results and user hearing preferences. The room sound calibration method based on hearing tendency control comprises the following steps: obtaining a hearing tendency parameter selected by a user through an interactive interface and a room impulse response of a current room; determining a group of frequency-related window control parameters according to the hearing tendency parameter and the room impulse response; determining a window length varying with frequency windowing strategy based on the window control parameters; performing windowing processing on the room impulse response by using the windowing strategy to extract an effective frequency response curve used for sound calibration; calculating the difference between the effective frequency response curve and a target frequency response curve to obtain correction filter coefficients; and calibrating the audio signal of an audio playback system based on the correction filter coefficients.

This invention discloses a closed-loop adaptive compensation method for loudspeaker production line calibration, relating to the field of testing equipment technology. It includes steps S1 (task initialization and template loading), S2 (continuous excitation playback and synchronous sampling), S3 (effective frequency point screening and confidence coefficient generation), S4 (actual frequency response calculation and confidence-weighted error construction), S5 (envelope and residual dual-component compensation update), S6 (compensation reconstruction of smoothing constraints and channel consistency constraints), S7 (double-buffer hot switching), and S8 (multi-factor freeze determination and result output). Through effective frequency point screening, confidence-weighted error construction, envelope and residual dual-component compensation update, and compensation reconstruction, it achieves excellent technical results in suppressing low-confidence frequencies, improving frequency response flatness, increasing convergence speed, and enhancing freeze stability.