12 results about "Overtone" patented technology

The present application relates to a method for obtaining an audio adjustment strategy, a computer device, a storage medium, and a computer program product. After receiving an audio adjustment instruction, the fundamental frequency points and overtone frequency points of the human voice signal in the audio to be adjusted are obtained, the overtone frequency points to be adjusted are determined among the overtone frequency points, and according to the comparison result between the amplitude ratio between the fundamental frequency points and the overtone frequency points and a preset amplitude ratio, the amplitude gain coefficient of the frequency band to be adjusted composed of the overtone frequency points is determined, and the frequency band to be adjusted and the amplitude gain coefficient are determined as the audio amplitude adjustment strategy of the audio to be adjusted. Compared with the scheme of manually determining the strategy to adjust the audio, this scheme determines the amplitude gain coefficient based on the standard proportional relationship between the fundamental frequency and the overtone, and determines the frequency to be adjusted based on the comparison between the frequency range and the preset frequency, so as to determine the strategy for adjusting the audio, improving the accuracy of determining the audio adjustment strategy.

The invention discloses a device, system and method for exploring the sound production problem of an airflow sound source, and the device comprises a first case, the interior of the first case is fixedly connected with a first switch, the two sides of the first switch are provided with sound insulation sleeves, and the interior of each sound insulation sleeve is sleeved with an air pump; the system comprises an audio processing module, a motor control module and a simulation module. The method comprises the following steps: step 1, COMSOL simulation; 2, building an experimental device; step 3, carrying out a pre-experiment; step 4, theoretical analysis; step 5, experimental verification; step 6, practical application; through theoretical derivation and experiments, related physical laws of fundamental tone, overtone and background noise of whistling sound source sound production are verified, and defects in overtone and background noise research in the prior art are made up; according to the device, by drawing the spectrogram, the physical law in the experiment process can be visually displayed, and the device can generate low-noise high-air-volume airflow and is more suitable for teaching research.

The invention belongs to the technical field of audio processing, and relates to an audio processing method and device for converting human voice into a microphone in real time and audio equipment. The method comprises the following steps: acquiring a real-time human voice audio, maintaining octave drop of a single-cycle waveform, and removing the continuity of the voice to obtain a throat corresponding to the real-time human voice audio; acquiring real-time human voice audio, and performing spectrum analysis, pitch analysis and consonant and vowel analysis to obtain a current analysis result; according to the current analysis result, energy calculation is carried out on the energy of the fundamental frequency multiple frequency point of the real-time voice audio, formant analysis is carried out on the real-time voice audio, overtones with the maximum energy or specified overtones are tracked, and whistles corresponding to the real-time voice audio are obtained; and obtaining a microphone calling audio corresponding to the real-time voice audio according to the throat sound and the whistle sound. According to the method and the device, ordinary human sound can be converted into sound with microphone calling characteristics in real time, and a microphone calling simulation function with high naturalness, high real-time performance and low threshold is realized.

This invention relates to the field of music education technology, specifically to a musical instrument playing training system and method based on real-time pitch monitoring. The system includes a musical instrument acoustic waveform analysis module, a tuning frequency tracking module, a pitch deviation analysis module, a muscle memory reshaping guidance module, and a musical instrument mastery blind spot positioning module. In this invention, by sensing the vibration envelope and reconstructing the energy distribution of the overtone series, the core musical tone components are accurately extracted from a complex acoustic environment, improving the accuracy of fundamental frequency identification and enhancing the ability to capture microscopic pitch drift. This effectively solves the problem of poor anti-interference in traditional detection methods, transforming pitch deviation into a concrete physiological cause analysis, optimizing the depth of performance mechanism analysis and enhancing the physical correction directionality. It achieves precise integration of correction actions and performance rhythm, ensuring the scientific nature of muscle memory reshaping and significantly reducing functional pitch deviation. By tracking the rate of movement deformation and tracing the source of technical bottlenecks, it optimizes musical instrument mastery and improves the level of intelligent music training.

The present invention relates to the technical field of musical instruments, and discloses a meteorite-made chime stone, the meteorite-made chime stone comprises a first chime stone and a second chime stone, the first chime stone and the second chime stone are prepared by cutting, grinding and tuning iron meteorite, stone meteorite or stone-iron meteorite, the nickel content of the meteorite is 5-30 wt%, the density is greater than or equal to 7.5 g / cm < 3 >, a plurality of chime stone pieces with different sizes and shapes are formed, and the first chime stone and the second chime stone are arranged in the chime stone. The thicknesses of the first stone chimes and the second stone chimes are 8-15mm, the chime pieces are of sentence-shaped structures, and asymmetric radians are arranged at the bottom edges of drum parts so as to adapt to the influence of the anisotropy of the meteorite material on the vibration mode. By adopting the iron aerolite or stone aerolite with specific components, accurately controlling the chime stone thickness and sentence-shaped structural design and matching with the asymmetric radian of the bottom edge of the drum part, the anisotropic characteristic of the aerolite material is effectively adapted, the structural design enables the vibration mode to better conform to the acoustics principle, the tone purity and the resonance duration are remarkably improved, and the tone quality is improved. The music player is especially suitable for playing music paragraphs with rich high-frequency overtones.

The present application relates to a kind of adaptive band optimization chorus sound separation method and system based on harmonic maintenance, first, the signal is collected by microphone array and time delay compensation and multi-channel time-frequency transformation are carried out;Sound source spatial positioning and initial band division are carried out using MUSIC algorithm;Using the greedy optimization strategy, the band boundary is dynamically adjusted and optimized by maximizing the objective function including signal-to-noise ratio and harmonic integrity measure;Then, the fundamental frequency is obtained by harmonic analysis and detection, and the time-varying harmonic trajectory is tracked across frames by Viterbi algorithm;On this basis, combined with MVDR beam forming and the introduction of harmonic maintenance factor, enhanced time-frequency representation is obtained by soft time-frequency mask;Finally, inverse short-time Fourier transform time-domain reconstruction and synchronous harmonic phase are carried out.The present application realizes the accurate allocation of voice part spectrum, effectively reduces the overlap interference, protects the overtone column of human voice while accurately separating, significantly improves the timbre fidelity and naturalness of separated voice part.

This invention discloses a real-time piano timbre simulation method and system based on machine learning, relating to the field of audio signal processing technology. The method includes: data acquisition and multi-dimensional annotation, acquiring multiple types of piano audio, covering techniques and seven dynamic levels, and simultaneously acquiring information such as key presses and techniques; audio preprocessing, including pre-emphasis compensation for high frequencies, Hanning window framing, Fourier transform to frequency domain, spectral subtraction for noise reduction and normalization; multi-dimensional feature extraction, extracting static features such as MFCC and spectral parameters, dynamic features such as first- and second-order differences, and overtone structures; two-stage model training, using stacked autoencoders for dimensionality reduction; real-time parsing, filtering and converting acquired performance data into parameter sequences; timbre synthesis, where the model generates a spectrum and performs an inverse Fourier transform into a waveform; and dynamic optimization, receiving user feedback. This invention solves the problems of traditional simulation methods; the two-stage model enhances timbre coherence, dynamic control achieves low latency, and multi-scenario adaptation and feedback optimization meet specific needs.

The invention discloses a guqin music evaluation standard system based on multi-modal perception and a playing style library construction method, and relates to the technical field of music information retrieval and cultural heritage digitization. According to the system, vibration, acoustics and visual data of Guqin playing are synchronously collected through a multi-modal hardware group (including a laser Doppler vibration meter, a 48-channel acoustic camera and an infrared motion capture system), unified feature vectors are generated through fusion in combination with a graph convolutional network (GCN), and a quantitative evaluation system covering harmonic distortion, overtone attenuation slope, pie voiceprint indexes and the like is constructed. Acoustic feature fingerprints of the pie (Guangling pie, Huangshan pie and the like) are extracted based on deep learning, an interpretable genre style model is established, and cross-space-time playing similarity analysis and style migration are supported. A reduced character score dynamic analysis engine is innovatively integrated, an ancient book music score is converted into an intelligent music score with strength and tone marks through OCR and knowledge graph technologies, and immersive teaching of genre characteristics is realized in combination with VR / AR.

A noise cancellation system (100) with harmonic filtering for a vehicle audio system, comprising: at least one first input sensor configured to receive a reference signal (x r1 [n], x r2 [n]); at least one second input sensor configured to transmit at least two narrowband input signals, wherein each of the input signals (x r1 [n], x r2 [n]) contains overtone noises; a processor (105) programmed to: Receiving the reference signal, wherein the reference signal comprises at least two narrowband reference signals (x r1 [n], x r2 [n]) includes, Receiving the narrowband input signals, Applying a gain reference control (346) to the reference signals to determine whether the frequencies of each of the reference signals (x r1 [n], x r2[n]) are within a predefined range of each other, Applying a bandpass filter (156) to one of the reference signals (x r1 [n], x r2 [n]), Applying a secondary path (376) based on the filtered reference signals (x r1 [n], x r2 [n]) to generate anti-noise signals, and Summing (378) the anti-noise signals and the input signals to produce an error signal (e m [n]) at the output sensor (145), Applying an adaptive filter (392) to the error signal (e m [n]) to remove overtone noise from the error signal (e m [n]) and Removing one of the reference signals (x r1 [n], x r2 [n]) in response to the frequencies of each of the reference signals (x r1 [n], x r2[n]) are within the predefined range of each other to prevent common overtone content from being detected on both reference signals (x r1 [n], x r2 [n]) during the adaptation of the algorithm.

The invention discloses a piano tone real-time simulation method and system based on machine learning, and relates to the technical field of audio signal processing, and the method comprises the steps: data collection and multi-dimensional marking, collection of multiple types of piano audios, skill coverage and seven levels of strength, and synchronous collection of information such as keys and skill; performing audio preprocessing, pre-emphasis high frequency compensation, Hanning window framing, Fourier transform to frequency domain, spectral subtraction denoising and normalization; multi-dimensional feature extraction: extracting static features such as MFCC and spectrum parameters, dynamic features such as first-order and second-order differences and an overtone structure; carrying out two-stage model training, and carrying out stacking auto-encoder dimensionality reduction; analyzing and acquiring playing data in real time, filtering and converting into a parameter sequence; synthesizing timbre, generating a frequency spectrum by a model, and performing inverse Fourier transform to form a waveform; and dynamically optimizing and receiving user scores. According to the invention, the traditional simulation problem is solved; the dual-stage model enhances tone coherence, and dynamic control realizes low delay, multi-scene adaptation and feedback optimization fitting requirements.

The invention relates to the technical field of music data processing, in particular to a violin playing auxiliary system based on artificial intelligence, and the system comprises an audio collection module which employs a microphone array to capture a violin playing sound field, and generates an original audio stream. A coding adaptation module generates a transmission code stream while extracting side information metadata including a noise masking threshold and a transient signal flag. The feature extraction module outputs a high-dimensional feature vector. The reconstruction module generates reconstructed audio. And the evaluation feedback module performs quantitative evaluation on the reconstructed audio, generates scores of intonation deviation degree, string rubbing depth and overtone purity, feeds the scores back to the coding adaptation module, and dynamically adjusts frequency weighting parameters of the psychological acoustic model. The music interface module outputs reconstructed audio and structured evaluation data. The system drives the coding adaptation module to adaptively optimize a compression strategy through score feedback of the evaluation feedback module, and the reconstruction module optimizes a generation process by using historical evaluation data to form a closed-loop data stream.

Inverters for direct current (DC) to alternating current (AC) conversion may comprise switching elements which produce harmonic overtones. Inductive elements may be added into the inverters, such as to attenuate effects of harmonic overtones. Methods and systems for a casing suitable for reduced potting over its inductive elements is described. One or more heat dispersing elements may be disposed in the casing. Related systems and methods are also described.