98 results about "High fidelity" patented technology

The invention discloses an intelligent sound effect processing system for an automobile DSP (Digital Signal Processor) power amplifier, particularly relates to the field of optimization of a vehicle-mounted sound system, is used for solving the problem of insufficient sound effect consistency and intelligent adaptation in a dynamic sound field, and is characterized in that acoustic data are acquired in real time through a microphone array, and a three-dimensional acoustic geometric model is constructed by combining ray tracing and an image source method; sound wave propagation paths and dynamic changes are accurately captured; frequency bands of low-frequency, intermediate-frequency and high-frequency signals are divided, spectrum distribution differences and gain fluctuation centers of all seats are evaluated, and a sound field consistency adjustment coefficient is generated and used for guiding design and optimization of a directional beam forming filter, adjusting amplitudes and phases of all sound channels, forming directional sound beams and suppressing noise interference; therefore, the method can dynamically adapt to the acoustic environment change in the vehicle, improves the sound effect balance and high fidelity, and enhances the auditory experience and satisfaction of passengers.

The invention discloses a real-time digital human generation system and method based on low-calculation-amount voice driving, and relates to the technical field of digital human interaction, and the system comprises an audio processing module which is configured to receive voice input in real time and extract an audio feature vector; the driving and rendering module is used for mapping the audio feature vectors into parameters representing mouth movement, generating a dynamic mouth image based on the preprocessed static face reference data and the parameters, and fusing the dynamic mouth image with the reference image; the synchronous control module is used for ensuring the synchronization of the audio features and the rendering video frames according to a timestamp alignment mechanism and a PID feedback control algorithm; and the dynamic scheduling module is used for monitoring a hardware resource load in real time and realizing dynamic allocation of computing resources through multi-thread parallel and task priority adjustment. According to the technical scheme provided by the invention, the breakthrough of low delay, high fidelity and low energy consumption of the digital human can be realized in mobile terminal, embedded and multi-platform scenes.

A computer system for upsampling decompressed audio data after lossy compression using specialized neural network techniques. The system processes compressed audio channels through an audio pre-processor that extracts spectral information, detects speech activity, segments audio, and normalizes input levels. A trained deep learning algorithm with multi-channel transformers using channel-wise and self-attention mechanisms recovers information lost during compression. The system further enhances audio quality through a time-frequency domain transformer applying Fourier transforms and Mel-scale frequency processing, while a perceptual quality assessor employing psychoacoustic models evaluates the output. This specialized audio processing approach significantly improves reconstructed audio quality by leveraging correlations between audio channels, addressing both spectral and temporal features, and optimizing for human perception characteristics, resulting in higher fidelity audio reproduction from compressed formats.

The invention provides a loudspeaker speech enhancement method and device, electronic equipment and a storage medium, and belongs to the technical field of speech signal processing, and the method comprises the steps: obtaining an initial spectrum signal and an initial power spectrum signal corresponding to an initial audio signal; inputting the initial power spectrum signal into a feedback elimination network model to obtain a target frequency domain signal; and obtaining a target audio signal based on the target frequency domain signal. The method comprises the following steps: performing mask adjustment on the frequency domain amplitude of an initial frequency spectrum signal by using a real number signal mask to generate a frequency domain estimation signal, and performing mask adjustment on the frequency domain amplitude and phase of the initial frequency spectrum signal by using a complex number signal mask to obtain a frequency domain enhancement signal. And finally, a target frequency domain signal is determined based on the frequency domain estimation signal and the frequency domain enhancement signal, so that the output target frequency domain signal has higher fidelity in the aspects of amplitude and phase, environmental noise and loudspeaker feedback signals can be removed to the greatest extent, and the squeal phenomenon is effectively inhibited.

The invention provides a hearing aid audio optimization method and system based on intelligent scene recognition, relates to the technical field of computers, and aims to realize high fidelity of signal amplification and ensure that the listening experience of a user in different environments is optimized at the same time, and the method comprises the following steps: obtaining initial audio data, classifying the initial audio data, and obtaining a classification result of the initial audio data; obtaining a target voice and background noise; performing first optimization processing on the target voice to obtain an optimized target voice; the definition degree of the optimized target voice is greater than that of the target voice; performing second optimization processing on the background noise to obtain environment voice; the energy of the environment voice is smaller than that of the optimized target voice; and determining an environment scene where the user is located based on the environment voice, and fusing the optimized target voice with the environment voice according to the environment scene to obtain an output voice.

Systems and methods are disclosed for retrieving information from legacy storage media with high fidelity and minimal contact. A read sensor with multiple read elements detects electromagnetic fields from tracks on a storage medium. Readings from different elements are combined to generate a composite interpretation, which can be used to, for example, generate digitized waveforms and determine if portions of tracks are damaged.

The invention relates to the technical field of voice recognition, and particularly discloses a high-fidelity AI voiceprint cloning method and system and a storage medium thereof.The method comprises the steps that user audio data are collected, de-noising processing is conducted on the audio data, the de-noised audio data are analyzed, and a de-noising processing quality evaluation result is generated; voiceprint feature extraction is carried out on the audio data meeting the denoising success result, denoising processing is carried out on the audio data not meeting the denoising success result again, parameters in the denoising processing process are adjusted, a cloned voice signal is generated, a cloned evaluation result of the cloned voice signal is collected, and the cloned voice signal is obtained. And storing the cloned voice signals meeting the voiceprint cloning success result, carrying out voiceprint feature extraction on the cloned voice signals not meeting the voiceprint cloning success result again, optimizing voiceprint feature extraction process parameters, and finally obtaining the high-fidelity AI cloned voice signals. And the problem of sound quality degradation caused by separation of de-noising processing and feature extraction links is solved.

The invention relates to the technical field of directional pickup, and discloses a directional pickup method and device, computer equipment and a medium, and the method comprises the steps: obtaining time domain voice signals collected by at least two microphones, and carrying out the time-frequency conversion; calculating an observation phase difference of each frequency point, and determining a target theoretical phase difference according to a target pickup direction specified by a user and the geometric parameters of the microphone; encoding the target theoretical phase difference into a query vector, encoding the observation phase difference into a key vector, and generating a space confidence mask through a geometric cross attention mechanism; and filtering the frequency domain signal by using the mask, and outputting enhanced voice in a target direction through inverse time-frequency transformation. According to the invention, through an attention mechanism guided by physical prior, the problem of phase winding is effectively solved, and high-robustness and high-fidelity speech pickup in any direction is realized in a strong reverberation and low signal-to-noise ratio environment.

The invention discloses a multi-modal guided high-fidelity image compression method and system and a medium, and the method comprises the steps: firstly extracting text and audio features aligned with image features through a pre-trained text encoder and a pre-trained audio encoder, and then inputting the text and audio features and the image features into a multi-modal feature fusion module for fusion. The multi-modal feature fusion module extracts and fuses multi-modal features from two dimensions of space and channel by using an affine transformation method. A multi-modal feature fusion mode is used for guiding at an encoder end, so that the feature extraction and expression capability is enhanced, and the distribution of potential features can be better predicted. In addition, a discriminator is designed to carry out multi-modal guided generative adversarial training, so that a high-fidelity image is obtained.

The invention discloses a digital human image generation method combining cold start driving and an active learning mechanism, and the method comprises the steps: processing input data through a preset few-sample emotional speech generation model which is derived from a cold start process and has high individuation and dynamic adaptability; according to the method, efficient personalized and multi-emotion voice batch generation can be realized on large-scale unlabeled input data, so that a target audio file with high fidelity and consistent emotion and personality expression is output; the preset few-sample emotional speech generation model is obtained by training a first qualified sample screened by a first cold start quality evaluator on the basis of candidate training samples, so that the model can be started under the condition of extremely little data, and automatic generation of a high-quality personalized digital human image under the condition of small samples is realized; besides, by inputting a target emotion embedding vector and a target character feature vector, synchronous expression generation under voice driving is realized, and the natural interaction capability and style consistency of the digital human are enhanced.

To provide an electrostatic speaker capable of maintaining high fidelity sound quality output even in a wide range of sound. [Solution] The electrostatic speaker includes an electrostatic sound generating diaphragm unit consisting of two parallel, spaced-apart electrode plates (1) and a diaphragm (2) between the electrode plates. The surface of the electrode plate facing the diaphragm is coated with a conductive layer, and both sides of the diaphragm are also coated with conductive layers. A voltage is applied to the conductive layers of the electrode plate and the diaphragm. Hollow sound-emitting holes are uniformly formed in the electrode plate, and an insulating buffer layer (3) is filled between the electrode plate and the diaphragm. Even if the applied high-voltage electric field voltage causes the diaphragm to distort due to excessive amplitude, the insulating buffer layer limits the diaphragm's amplitude, preventing distortion. The insulating buffer layer also prevents contact between the diaphragm and the electrode plate and maintains an appropriate minimum distance, preventing destructive discharges caused by too close a distance between the diaphragm and the electrode plate. This prevents the appearance of high-voltage discharge noise and prevents the diaphragm from being damaged by high-voltage discharge.

The invention relates to the technical field of data processing, in particular to an audio timbre conversion method and device, electronic equipment and a storage medium, and the method comprises the steps: determining a target language and a target timbre, obtaining the timbre feature of the target timbre, and obtaining the reference audio data based on the target language and recorded by adopting the reference timbre; based on the reference audio data, extracting content features, and based on the reference audio data, extracting corresponding fundamental frequency features and phonetic rhyme features; and reversely generating target audio data of the target timbre based on the timbre feature, the content feature, the fundamental frequency feature and the rhyme feature of the target timbre. Thus, in the process of reversely generating the target audio data, the pronunciation tone and rhythm in the reference audio data can be restored with high fidelity, the pronunciation effect based on the target language in the target audio data is guaranteed, and the generation quality of the audio data is improved.

The application discloses a kind of underground space optical cable high fidelity sound restoration method based on DAS system, comprising: the real-time signal collected to optical cable is carried out data demodulation, identify whether it contains energy impact signal generated by the optical cable after being padded and being hit, if it contains, the energy abnormal distribution position in energy impact signal is positioned, obtains positioning point;Continuous analysis time domain signal of positioning point, after filtering low-frequency interference signal, phase anomaly signal is rejected;The sound signals of a plurality of equivalent sensing nodes adjacent to the positioning point are superimposed and averaged to suppress additive noise, and then the sound signal is restored.The application can realize rapid positioning according to the spatial distribution mapping relationship of optical cable, and after filtering background noise from the collected signal, the highly correlated signals of multiple spatial sensing nodes near the positioning position are linearly superimposed to suppress additive random noise, and high-fidelity sound signal restoration is realized.

The utility model discloses a portable high-fidelity remote microphone, which belongs to the field of microphone technology; it includes a shell; a microphone array acquisition board is arranged in the shell; a plurality of identical electret microphones are arranged on the microphone array acquisition board, and the plurality of electret microphones are linearly and evenly distributed; the microphone array acquisition board is connected to an electronic circuit board, and the microphone array acquisition board can collect sound information and spatial position information of the sound, and convert it into an electrical signal and transmit it to the electronic circuit board. The utility model has a simple structure and is easy to use. The sound pickup distance can reach 30m. The electronic circuit board can process far-field audio signals to ensure high-fidelity audio output; the utility model can pick up high-fidelity and high-definition sound signals under far-field conditions.

The utility model discloses a USB high-fidelity decoder with a vibration unit, which comprises an integrated board. According to the utility model, the USB assembly, the system control assembly, the Bluetooth assembly, the first motor assembly, the first vibration driving assembly, the audio signal processor assembly, the loudspeaker assembly, the second motor assembly, the second vibration driving assembly, the earphone assembly and the high-fidelity digital-to-analog conversion assembly are integrated in the integrated board, so that the integrated board forms an integral module; the module can be attached to the surface of an object such as a mobile phone and can also be embedded into a mobile phone shell, the design that sound is experienced through the touch sense of the human body is achieved, a user can feel the sound in multiple dimensions on the basis of traditional auditory sense, the overall experience of entertainment equipment is enhanced, and meanwhile brand new understanding of music / sound is achieved.

The application discloses a kind of based on semantic drive's audio processing method and device;The method comprises: spatial decomposition is carried out to multi-channel audio signal, and forward target signal and backward environmental signal are extracted;Respectively carry out statistical noise reduction, obtain forward noise reduction gain and backward noise reduction gain;Respectively input acoustic semantic perception network, obtain forward and backward semantic probability vector;Based on semantic probability vector and semantic complementary weight matrix, the dynamic transmission weight of backward environmental signal is calculated;Weighted fusion is carried out using forward noise reduction gain, backward noise reduction gain and dynamic transmission weight, and enhanced audio signal is output;The present application transmits side rear important acoustic event on demand through semantic gating mechanism, retains environmental situation awareness capability while suppressing environmental noise, adopts statistical noise reduction and lightweight network heterogeneous deployment, meets the requirement of low delay and high fidelity of edge device, significantly improves the user experience of auditory enhancement device.

1. The name of the design product: high fidelity player (H41). 2. The use of the design product: the design product is used for high fidelity player. 3. The design points of the design product: in shape. 4. The picture or photo that best indicates the design points: perspective view 1.

This invention relates to a dual-channel audio control circuit, comprising a dual potentiometer circuit, a left channel driver circuit, a left channel output filter circuit, a right channel driver circuit, a right channel output filter circuit, a left channel speaker, and a right channel speaker. The dual potentiometer circuit receives and adjusts the dual-channel audio signals. The input terminal of the left channel driver circuit is connected to the left output terminal of the dual potentiometer circuit, and the input terminal of the right channel driver circuit is connected to the right output terminal of the dual potentiometer circuit. The left channel output filter circuit is connected between the left channel driver circuit and the left channel speaker, and the right channel output filter circuit is connected between the right channel driver circuit and the right channel speaker. This invention offers significant advantages in elevator voice systems, providing a more natural and immersive listening experience while meeting the requirements for high fidelity, flexible control, and system integration.

The application belongs to the technical field of electro-acoustic conversion equipment, and particularly relates to a replaceable-line high-fidelity noise-reducing in-ear earphone based on a floor iron moving unit, which comprises a 5-unit 4-frequency division hybrid driving module formed by the floor iron moving unit and a micro moving coil unit, and the two achieve signal cooperation through a phase synchronization circuit; full-frequency sound quality optimization and hybrid noise reduction are achieved through the phase synchronization circuit and a DSP+ANC chip, a sound cavity is matched with a shape memory alloy damping adjusting sheet, magnetic attraction and sealing of a 2-pin interface improve durability and sealing performance, a double-mode switching audio line and a three-in-one replaceable plug adapt to multiple devices, and touch control and a selectable Bluetooth kit realize convenient interaction. The application breaks through the sound quality limitation of a single unit through a double-unit cooperative driving structure, realizes dynamic adaptation of noise reduction and sound quality through an adaptive acoustic adjusting system, improves durability and personalized upgrade space through a reinforced replaceable line design, enriches use scenarios through an intelligent interaction module, and meets the multiple needs of high-end users for high-fidelity sound quality, flexible use and intelligent experience.

The invention provides a hearing aid device, method and device with a self-adaptive hearing compensation function, the hearing aid device comprises a wireless communication module, a main control chip and an audio output module, and a pre-training hearing compensation model is built in the main control chip. A full-band continuous personalized frequency response gain curve with gain amplitude positively correlated with hearing loss is output through model reasoning, an audio output module performs full-band frequency-point-by-frequency balanced linear gain compensation on an original digital audio signal played by the device based on the curve, the audio dynamic range is not compressed in the whole process, the original spectrum structure is not changed, and the gain amplitude of the original digital audio signal is not changed. And the high-fidelity tone quality is kept. Different from a traditional hearing aid and a conventional audio playing device, only for audio processing inside the device, pickup and noise reduction are not needed, hearing features of a hearing impaired user can be accurately adapted, clear and balanced listening can be achieved, tone quality distortion can be avoided, and compensation precision, use comfort and scene universality are considered.

To improve the encoding of scene-based audio data, after generating background components, foreground audio signals, and corresponding spatial components from scene-based audio data (such as high-order high-fidelity stereo HOA coefficients), correlation analysis is performed to determine the ordering pairs of correlation components from the foreground and background audio signals that will undergo stereo psychoacoustic audio encoding. On the decoding side, before reconstructing the scene-based audio data using the reordered correlation components, the encoded correlation coefficient pairs are decoded using stereo decoding and reordered using reordering information included in the bitstream.

The invention provides a training method of a music generation model, a music generation method, equipment and a medium, and relates to the technical field of artificial intelligence. Performing first up-sampling training on the training audio coding information based on a first generator in a vocoder in a music generation model to obtain intermediate training audio data of a first sampling rate, the intermediate training audio data is subjected to first up-sampling training based on a first generator in the vocoder, second up-sampling training is carried out on the intermediate training audio data based on a second generator in the vocoder to obtain training audio data of a second sampling rate, the second sampling rate is larger than the first sampling rate, and generative adversarial training is carried out on the vocoder based on the training audio data to obtain a trained music generation model. By adopting the method and the device, the dependence of model training on high-sampling-rate audio data can be reduced while the music generation model is ensured to output high-fidelity songs, so that the model training cost is controlled.

The utility model belongs to the technical field of mobile communication equipment, and particularly relates to a front cover for a loudspeaker, a loudspeaker module and a mobile terminal. The front cover for the loudspeaker comprises a metal sheet, the metal sheet is provided with a via hole, and the surface of the metal sheet is provided with a silica gel ring. According to the utility model, a better sound transmission effect of the loudspeaker can be realized, the sound transmission obstruction is small, the sound can clearly and naturally pass through the loudspeaker, the sound output of audio equipment can maintain high fidelity, and the loudspeaker can have stable performance under different frequencies; moreover, good damping characteristics are achieved, appropriate damping adjustment can be carried out on sound, reflection and scattering of the sound are reduced, distortion is reduced, and the sound quality is improved; the loudspeaker module also has excellent protection performance, realizes effective water and dust prevention, and can effectively prevent water, dust and other small particles from entering the loudspeaker module; and the anti-impact performance is higher, good protection is provided for internal parts, and damage caused by external force collision is reduced.

The invention provides an expression migration method and apparatus, an electronic device and a computer readable storage medium. The method comprises the steps of obtaining a source face image containing a first expression, a source audio and a target face model to be subjected to expression migration; determining expression feature data corresponding to the source face image, audio feature data corresponding to the source audio and identity feature data corresponding to the target face model; generating a face control parameter corresponding to the target face model according to the expression feature data, the audio feature data and the identity feature data; and according to the face control parameter, adjusting the position of a vertex in the target face model so as to generate an animation presenting a first expression with expression details matched with the emotion information for the target face model. Therefore, high fidelity of expression details can be kept in the expression migration process, the cost of expression migration is effectively reduced, and the flexibility and expandability of expression migration are improved.

The invention discloses a television cinema system with high-sensitivity, high-fidelity and high-resolution audio quality, and belongs to the electroacoustic field of electricity. A traditional television uniquely pursues the ultra-light and ultra-thin whole machine, and the sound effect of the whole machine is poor due to the fact that a loudspeaker of the traditional television lacks necessary air volume. The volume of the cavity between the display screen and the rear cover plate of the television is properly increased, the track-shaped full-band and bass patent loudspeaker is arranged in the television, and the left, right and rear surround sound boxes and the ceiling ceiling sound box which are mounted in a wall-mounted manner in a television broadcasting hall are matched; two-dimensional and three-dimensional cinema systems with internal sound boxes of 2.1, 3.1, 5.2. 4, 7.1. 4 and the like and wireless external sound boxes which are in cross complementation can be successfully established. The lowest lower limit frequency of bass is 76Hz, the highest upper limit frequency of high pitch is 30kHz, and the loudspeaker can be matched with LCD (liquid crystal display) or OLED (organic light emitting diode) televisions with various screen sizes to meet the urgent requirements of vast consumers.

A multi-driver multi-channel single enclosure Height-Channel (e.g., ATMOS™ or DTS-X®) enabled soundbar loudspeaker system 260 uses a novel signal processing system, driver mounting configuration (310L, 310R) and method to provide a high fidelity home theater listening experience, in a manner which relies on a new method for cancellation of unwanted direct (not ceiling-bounced) radiation of the Height-Channel (or virtual height envelopment) channel's sound 213DS.

The invention discloses a partition purification power supply management time sequencer for a high-fidelity sound system, and the time sequencer comprises a relay board, the relay board is provided with a first power supply purification circuit and a second power supply purification circuit, two differential mode capacitors in the second power supply purification circuit are in short circuit, and the second power supply purification circuit is used for supplying power to a post-stage power amplifier. The two differential mode capacitors in the first power supply purification circuit are electrically connected through the common mode inductor, and the first power supply purification circuit is used for supplying power to the access turntable, the multicast, the decoding or the pre-stage power amplifier. By optimizing the filter circuit, the requirements of other sound equipment except the post-stage power amplifier on the current purity can be met, and the phenomena of'soft ', 'weak' and sound lag of bass in the use process of the post-stage power amplifier can be avoided; therefore, the sound heard by a user of the high-fidelity sound system is pure in sound quality and vigorous and powerful in bass, and is rich in impact force and infectivity.

An apparatus includes a memory configured to store first audio data and second audio data. The device also includes one or more processors coupled to the memory and configured to determine priorities of audio sources of the audio scene. The one or more processors are further configured to render the first audio data using the object renderer to generate a first audio signal. The first audio data represents a first audio source associated with a first priority. The one or more processors are further configured to render the second audio data using the first Ambisonics renderer to generate a second audio signal. The second audio data represents a second audio source associated with a second priority.

The invention relates to a Wi-Fi streaming media high-fidelity audio real-time transmission system and a Wi-Fi streaming media high-fidelity audio real-time transmission method adopting dynamic code rate adaptation. The system comprises a transmitting end and a receiving end, wherein the transmitting end realizes layered encoding and packaging of audio signals through an audio acquisition module, a layered encoder, a forward complexity analyzer and a data packaging module; and a receiving end completes data analysis and low-power-consumption decoding through a data analyzer, a heterogeneous decoding processor and a power consumption scheduling manager. A dynamic code rate control mechanism based on a network state and a buffer area state is adopted, and a power consumption pre-scheduling strategy of forward complexity analysis is combined, so that collaborative optimization of code rate self-adaptive adjustment of a sending end and dynamic configuration of processing resources of a receiving end is realized. The technical problem that sound quality, delay and power consumption are difficult to consider in traditional wireless audio transmission is effectively solved, the power consumption of the receiving end is remarkably reduced while high-fidelity sound quality and low transmission delay are guaranteed, the method is particularly suitable for portable equipment such as earphones, and the endurance time of the portable equipment can be greatly prolonged.