13 results about "High fidelity" patented technology

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 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.

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 utility model belongs to the audio signal transmission technical field, concretely relates to a kind of vehicle-mounted multi-channel wireless sound system and vehicle, it include: multiple loudspeaker, and: control module, control module includes main control control module and controlled control module;UWB component, including main control UWB component and controlled UWB component, main control UWB component is electrically connected with main control control module, controlled UWB component is electrically connected with controlled control module, each controlled control module is electrically connected with one or more loudspeaker.The utility model solves the problem that vehicle internal high fidelity audio signal transmission is limited by wiring harness cost, utilizes UWB transmission technology while high fidelity transmission audio signal, avoids complex audio line wiring, and single control loudspeaker can be formed multiple sound fields in vehicle.

The utility model discloses a HI -FI micro system sound, including face shell, top shell, wood top bottom plate and bottom shell, bottom shell is fixed on wood top bottom plate, top shell covers the top of wood top bottom plate, and the both sides of top shell are locked on bottom shell, and the both sides of bottom shell all are equipped with tray support, and the fixed mainboard support of tray support has the inside slide connection of core, and the front end of top shell is equipped with the screen PCB, and face shell covers the outside of screen PCB, and the both sides of top shell are provided with wood sound box, and the front end of wood sound box is equipped with wood box front plate, and the wood box front plate is covered with the sound box cloth net frame. This HI -FI micro system sound is used to play the similar sound height of original, and receives CD and enters the core, releases the sound through the wood sound box of both sides, greatly restores the sound, and the high fidelity effect is good.

The application discloses a sound signal processing method and device, a virtual reality device and a storage medium. The method comprises the following steps: obtaining a gaze track of a user and sound body information in a field of view of the user, wherein the sound body information comprises position information and acoustic classification information of each sound body; obtaining attention information of the user to each sound body according to the gaze track and the position information of each sound body; and processing a sound signal emitted by each sound body according to the attention information of the user to each sound body and the acoustic classification information of each sound body. The technical scheme of the application can design a sound processing scheme for a sound body according to the attention degree of the user to the sound body, so that the processed environmental sound meets the real needs of the user, the user can obtain the environmental sound that he / she pays attention to, and the hearing effect provided by the virtual reality device has high fidelity.

This application discloses an audio processing method, device, and storage medium, belonging to the audio field. The method includes: filtering the left and right channel signals of the original audio using a first low-pass filter to obtain low-frequency left and right channel signals; the first low-pass filter is a finite impulse response low-pass filter with minimum phase property; filtering the left and right channel signals of the original audio using a first high-pass filter to obtain high-frequency left and right channel signals; and mixing the high-frequency left and right channel signals with the low-frequency left and right channel signals after sound field adjustment to obtain a stereo output signal. This solution achieves flexible sound field adjustment while completely solving the problem of blurred sound image positioning caused by nonlinear phase distortion in traditional solutions, as well as the problem of auditory imbalance caused by changes in low-frequency energy distribution, thereby achieving a high-fidelity and high-reproducibility stereo sound field processing effect.

The application discloses a high-fidelity AB class audio power amplifier. According to an embodiment of the audio power amplifier, the main architecture is a two-stage operational amplifier adopting a gain boosting technology, and according to a circuit topology structure and working principle, the audio power amplifier can be divided into five parts, i.e. an input stage, a gain boosting stage, an output stage, a bias circuit of a floating voltage source and a frequency compensation circuit. By setting the gain boosting stage, the output impedance can be increased without additionally increasing the common-source and common-gate device, and then the gain is increased. The application can make the audio power amplifier keep high gain to reduce nonlinearity when driving a small resistance earphone load, and a complex frequency compensation network is not needed to be designed.

The application discloses a sound replication method and related device, and relates to the technical field of speech synthesis. In one aspect, a hybrid expert sound replication model is adopted, each expert sub-model corresponds to only one acoustic category, and all the expert sub-models adopt a lightweight structure. Therefore, the end-side device only needs to activate a single expert sub-model matched with target acoustic category information for reasoning, which greatly reduces the model parameter quantity and computing cost, and thus successfully runs on the low-resource end-side device. In another aspect, the target speaker personalized acoustic features pre-delivered by the cloud and locally stored are directly used in the replication process. The features accurately depict the details of the speaker's timbre, accent, prosody, etc., and make up for the insufficient expression ability of the lightweight model. Therefore, the scheme can generate a highly similar replicated voice to the target speaker on the low-resource end-side device without relying on real-time calculation of the cloud large model, and takes into account the feasibility of end-side deployment and the high fidelity of the synthesized sound quality.

This invention discloses a method, system, medium, and robot for speech extraction during robot interaction, designed in the field of artificial intelligence technology. The method includes: acquiring the registered audio and mixed speech of the target speaker; extracting frame-level voiceprint features from the registered audio and pooling them into sentence-level voiceprint features; extracting sub-band embedding sequences of multiple different sub-bands from the mixed speech; inputting the fused features obtained by fusing the sub-band embedding sequences and sentence-level voiceprint features into a speaker extraction module; the speaker extraction module dividing the spectrogram into multiple sub-spectral maps according to predefined frequency bands; mapping each sub-spectral map to a sub-band embedding sequence; inputting all sub-band embedding sequences into a hybrid expert model; obtaining the separated target speaker features; and decoding the target speaker features to obtain the target speech of the target speaker. This method achieves robust, stable, and high-fidelity extraction of the target speaker's speech.