31 results about "Professional audio" patented technology

The invention relates to the technical field of browser-side high-performance computing, in particular to a WebAssembly-based high-performance front-end audio and video processing method and system, and the method comprises the following steps: designing a modularized Wasm runtime container, constructing a zero-overhead memory interaction mechanism, and establishing a hardware adaptive scheduling engine; the method has the beneficial effects that dynamic loading of an audio and video algorithm is realized by designing a modularized Wasm runtime container, serialization overhead of communication between threads is eliminated by utilizing a shared memory mechanism, a hardware self-adaptive dynamic scheduling system is created, and finally three core objectives are achieved: execution efficiency close to native codes is provided in a pure browser environment; establishing a standardized processing pipeline without dependence of a third party; intelligent scheduling and elastic expansion and contraction of terminal heterogeneous computing resources are realized, and Web end infrastructure support is provided for professional audio and video applications.

The invention relates to the technical field of wireless communication, in particular to a wireless audio transmission method, device and system, electronic equipment and a storage medium. In the present invention, a central device can control two peripheral devices at the same time by using the same first connection isochronous flow link protocol data unit, and allows one link to dynamically occupy two time slots to perform continuous retransmission when needed through the setting of a link transmission enabling mark. The flexible time slot resource sharing mechanism effectively and greatly improves the probability that the central equipment correctly receives the second connection isochronous flow link protocol data unit, thereby improving the transmission reliability of the ultra-low delay wireless double-microphone audio. In addition, due to the independent design of the transceiving time sequence, error propagation is avoided, and the overall robustness of the system is enhanced. Finally, according to the scheme, high reliability, high efficiency and millisecond-level ultra-low delay performance required by professional audio application are achieved with lower signaling overhead.

This invention relates to the fields of animal behavior, animal welfare, and mental health conditioning, specifically a music conditioning system and method for pets and laboratory animals. It is a pure PC-based software system, including an audio editing and playback control module, a music library management module, and a music playback module. The output of the audio editing and playback control module is connected to the input of the music playback module, and the music library management module and the audio editing and playback control module have a bidirectional data connection. This invention precisely matches different animals through three parameters: pitch shifting, speed adjustment, and volume adjustment, significantly improving audio compatibility. The effects of emotional stabilization, anxiety relief, and sleep promotion are significantly better than general music. One-click templates are suitable for quick home use. Customizable visual editing is suitable for pet hospitals, laboratory animal centers, and research institutions to establish standardized programs. Waveform graphs, spectrograms, and acoustic spectrograms are displayed synchronously in real time, allowing users to intuitively judge the audio structure. Adjustment accuracy is improved by more than 80%, requiring no professional audio knowledge.

An AI-powered multimedia workstation integrates a digital audio workstation (DAW), visual graphics editor, and office productivity tools within a single platform. Operating on an AI / quantum-inspired software system, it performs real-time audio, visual, and data processing. A central AI / Quantum Content DNA Engine enables structural, stylistic, and semantic analysis, separation, and manipulation of multimedia content, including audio, video, images, and text. The workstation features a hybrid control surface with dynamic haptic feedback faders, high-resolution touch displays, and professional audio input / output interfaces. AI functions include automated music composition, intelligent mixing and mastering, cross-modal content generation, and context-aware workflow assistance. Quantum-inspired algorithms accelerate rendering, simulation, and large-scale data analysis. The system supports real-time global collaboration via high-speed network protocols and cross-platform compatibility with external devices and software. By combining AI-assisted and quantum-inspired processing, the workstation enhances creative production, performance, and productivity, providing an integrated environment for professional audio, visual, and office applications.

The invention discloses a test method, device and equipment of an audio subsystem and a medium, and relates to the technical field of airborne avionics equipment testing, an initialization configuration file is compiled according to channel information, channel connection can be realized according to the initialization configuration file, and thus connection with tested equipment is realized. And based on different test steps, switching of different channels is realized, and smooth transmission of various data is ensured. The test sequence set is constructed according to the test procedure of the tested equipment, and the test host can automatically complete the test of each function according to the test steps contained in each test sequence, so that the high efficiency, comprehensiveness and accuracy of the test are ensured. The test results are verified according to the expected results contained in the test sequences, whether the test succeeds or not can be judged, rapid processing and analysis of the test results are achieved, and test errors are reduced. And the whole test process reduces the dependence on a professional audio test instrument, so that the test cost is remarkably reduced.

This application discloses an AI-based method and system for intelligent control of digital speaker sound field parameters, relating to the field of digital professional audio equipment technology. The method includes: acquiring environmental acoustic parameters and user-played content, and identifying the playback scene to obtain the playback scene; performing auditory demand analysis to obtain an auditory demand solution; obtaining an initial sound field control target, and conducting an environmental impact assessment based on the environmental acoustic parameters to obtain an optimized sound field control target; iteratively optimizing the control parameters of the digital speaker in conjunction with the playback scene to obtain an optimized control scheme, and intelligently controlling the digital speaker. This solves the technical problem that existing digital speaker sound field parameter control methods cannot dynamically adapt to complex environmental changes and personalized auditory needs, resulting in poor sound quality and a poor user experience.

1. The name of the design product: power amplifier. 2. The use of the design product: the design product is used for power amplifier of high-fidelity sound system, home theater, professional audio system, etc. 3. The design points of the design product: in shape. 4. The picture or photo that can best show the design points: perspective view.

The invention provides an intangible cultural heritage digital protection system and method, and relates to the technical field of digital protection. The intangible cultural heritage digital protection system comprises a digital recording and storage module, a digital platform and database module, an interactive digital display module, a cultural inheritance and education module, an artificial intelligence and big data analysis module and a cultural heritage propagation and protection module. And the digital recording and storage module comprises a video and audio recording unit which is used for recording a traditional artistic expression form through high-definition camera shooting and high-quality recording equipment. According to the invention, the details of intangible cultural heritage can be accurately recorded by using high-precision digital acquisition equipment such as a high-definition camera and professional audio recording equipment, and new possibility is provided for inheritance of the cultural heritage through high-precision recording and storage, an innovative display and interaction mode and a globalized transmission means.

The invention relates to a 12-channel XLR audio playing system based on an SOC chip platform and an audio and video separation output method, and belongs to the technical field of audio and video processing. A dual-path output architecture is used; wherein the video path is from the player to MediaCodec decoding to Surface rendering to HDMI (High Definition Multimedia Interface) output; the audio path is from MediaCodec decoding to a user-defined audio module to a JNI interface to ALSA direct control to 12-channel XLR output. The method comprises the following steps: 1) initializing and configuring; 2) preparing bottom layer equipment; (3) double-thread parallel processing is carried out; 4) customizing audio routing and playing; according to the technical scheme, 12-channel XLR professional audio output is supported, the multi-channel, high-fidelity and low-delay audio output capacity is achieved, in the aspect of audio and video physical separation output, videos are output through HDMI, audios are output through XLR, the professional audio and video system integration requirement is met, in the aspect of low-delay audio processing, Audio Track and HAL layers are bypassed, ALSA is directly controlled, and the audio delay is smaller than 20 ms.

The present invention relates to the field of artificial intelligence and speech synthesis technology, and discloses an emotional speech synthesis method, apparatus, computer equipment, and readable storage medium, comprising: obtaining a target text to be speech synthesized, identifying an emotional vector representation of the target text; predicting the emotional style of the emotional vector representation to obtain an emotional style weight; determining the phoneme embedding sequence of the target text based on a pre-trained acoustic model, and controlling the acoustic model to adjust the phoneme embedding sequence using the emotional style weight so that the acoustic model generates the emotional speech of the target text using the adjusted phoneme embedding sequence. Even in the absence of professional audio material or emotional annotation, speech output with high emotional expressiveness can be obtained, thereby improving the generalization ability of the model.

This application discloses a multi-protocol compatible TDM multi-channel allocation method, relating to the field of audio data transmission technology. By using a protocol discrimination strategy based on signal characteristic parameters, compatibility and switching efficiency are improved. Then, based on the discriminated protocol type, the audio data to be allocated is encapsulated into dynamically reconstructed TDM frames, which allows multiple output modules to share a single TDM data line, reducing material consumption and cost. The channel allocation strategy is then obtained, and the default channel allocation strategy is adjusted according to the channel allocation strategy. Only a software update strategy is needed, and the flexibility of channel expansion and adjustment far exceeds that of existing static allocation schemes, which can adapt to diverse dynamic scenario requirements such as professional audio and in-vehicle audio.

The invention relates to the field of digital professional sound equipment, in particular to a sound power amplifier which comprises a power amplifier body and a machine shell, a containing cavity used for containing the power amplifier body is formed in the machine shell, the machine shell comprises a bottom shell and an upper cover, and a positioning frame used for containing the power amplifier body is installed in the bottom shell. The positioning frame is elastically connected with the bottom shell through the damping assembly, the power amplifier body is fixedly installed on the positioning frame through the locking assembly, the rear end of the upper cover is rotationally installed at the rear end of the bottom shell through a hinge, an operation opening is formed in the front end of the upper cover, and an operation cover is rotationally installed on the operation opening. A junction box is fixedly installed on the rear side of the bottom shell, an opening in one side of the junction box communicates with an opening in the rear side of the bottom shell, an opening in the other side of the junction box is sealed through a sealing cover, and the power amplifier body can be protected and placed in a machine shell and is prevented from being damaged in the use process in a complex environment.

1. The name of the design product: multifunctional effects device (Xrecer R1). 2. The use of the design product: used in professional audio scene, can provide a variety of timbre effects and stable performance. 3. The design points of the design product: in shape. 4. The picture or photo that can best show the design points: perspective view 1.

The utility model belongs to the technical field of building design and professional audio-video system accompanying cable lifting, and particularly relates to a loop chain type lifting point take-up device lifting system which comprises a control box. A main body is arranged on one side of the control box, and a take-up and pay-off mechanism is fixedly mounted in the main body; wherein the main body comprises a shell, a top plate is fixedly mounted at the top of the shell, a lifting lug is arranged at the top of the top plate, the bottom of the shell is fixedly connected with a wire holder through a connecting wire, and a limiting assembly is arranged outside the connecting wire. According to the utility model, through the take-up and pay-off mechanism installed in the housing, the connecting line and the wire holder can synchronously lift along with the audio-video equipment, and meanwhile, the main body and the take-up and pay-off mechanism are matched to hide the collected line, so that the aesthetic property of the equipment is improved; and potential safety hazards and maintenance inconvenience caused by stacking of various lines can be avoided.

The invention provides a method of an ultra-high-definition IP multimode gateway system based on an x86 architecture ST2110 standard, and relates to the technical field of broadcast television and professional audio and video transmission. By configuring ST 2110 IP stream input and output parameters, bidirectional conversion between SDI signals and video, audio and auxiliary data IP streams is realized. According to the method, separation, packaging and parameter matching are carried out on the SDI data, audio and video time sequence synchronization is carried out, and then output is carried out, so that the flexibility and stability of media transmission and the multiplexing capability of existing IT infrastructures are effectively improved.

1. The name of this design product: XLR connector. 2. Purpose of this design product: used to connect professional audio equipment. 3. The key point of the design of this product lies in its shape. 4. The picture or photo that best illustrates the design points: Stereoscopic drawing 1. 5. Other situations that need explanation Other explanations: The double-dotted line in this application indicates no length limit.

The present application discloses a standard compliance test and verification system for high-performance network audio streams, including a test detection system, and a variety of sound source distribution systems, audio jump systems, multi-form high-performance network audio stream input systems, multi-form high-performance network audio stream processing systems, multi-form high-performance network audio stream output systems, and multi-brand network switching systems that are respectively connected thereto. According to the embodiments of the present application, problems such as non-interoperability and non-openness between various high-performance network audio stream transmission protocols in the current professional audio industry field are solved, and the test and verification of standard compliance indicators such as electrical performance, network transmission, and control performance are supported for multi-brand, multi-type, and multi-form high-performance network audio stream products, improving the test efficiency.

The invention discloses an audio transmission method based on an FPGA (Field Programmable Gate Array) chip, which is characterized in that the FPGA chip of host equipment adopts a parallel processing mechanism, channel data transmitted by FPGA chips of multiple slave equipment can be simultaneously received, each channel is independently and synchronously processed, the waiting time of sequential processing is avoided, and the audio transmission real-time performance is greatly improved; the requirements of professional audio processing and the like on time-sensitive scenes are met; secondly, the FPGA chip realizes audio transmission processing based on hardware logic, the execution speed is high, the delay is low, data can be directly processed and forwarded by hardware in an RGMII network port communication path, the time loss is reduced, and the real-time performance is further guaranteed; besides, a communication path is constructed based on an RGMII network port protocol, the mature and stable characteristics of the communication path are utilized, high-speed transmission is supported, the anti-interference capability is high, data can be ensured to be complete and accurate and audio signals can be stably transmitted in a long-distance and complex electromagnetic environment, and the audio transmission requirements in scenes of different geographic positions, complex industrial environments and the like are met.

1. The name of the design product: monitoring sound (ST MS300). 2. The use of the design product: applied to professional audio field such as recording studio, radio station, film and television production, music mixing, etc. 3. The design points of the design product: the combination of shape and pattern. 4. The picture or photo that best indicates the design points: perspective view 2.

1. The name of the product of this design: audio cable. 2. Purpose of the product of this design: used to connect musical instruments, audio equipment and professional audio to transmit audio signals. 3. The design key point of this appearance design product lies in the shape. 4. The picture or photo that best illustrates the design points: Stereoscopic diagram 1.

The present disclosure discloses a NiMo-based nanomaterial earphone diaphragm sand a preparation method thereof. The diaphragm uses a paper base film as the basis, and a uniform distribution of a NiMo-based nanomaterial coating is formed on a surface of the base film by plasma modification and electrostatic spraying technology, forming a concentric ring structure and using the NiMo-based nanomaterial as connecting lines to enhance the stability and conductive performance of the overall structure. This design not only significantly improves the audio quality performance of the earphones, including clarity, detail restoration, and dynamic range, but also greatly improves the durability and environmental adaptability of the diaphragm. The present disclosure provides an innovative solution for high-end earphone manufacturing, and is especially suitable for music lovers who pursue the ultimate audio experience and professional audio fields.

1. Name of the product in this design: Smart Screen (Sound, Light and Electricity Version). 2. Application of this design: This design is used for real-time traffic guidance, safety warning and dynamic information dissemination and broadcasting of professional audio on highways and urban expressways. 3. The key design feature of this product is its shape. 4. The image or photograph that best illustrates the design's key points: a 3D model.

The invention discloses a low-power-consumption portable noise monitoring device, and relates to the technical field of environment monitoring. Comprising a main control module, a noise sensing module, a communication module, a power management module and an alarm module which are highly integrated in a compact shell. An event-driven asynchronous working mechanism and a one-master multi-slave power supply design are adopted, and ultra-low power consumption is realized through a special low-power-consumption function and a redundant hardware closing strategy; the noise sensing module adopts a software and hardware collaborative dynamic gain adjustment scheme and is combined with a professional audio ADC chip and software control to ensure wide-range and high-precision measurement; the communication module adopts an LPWAN technology and supports a deep power saving mode; the device is awakened through RTC timing or noise signal triggering, Leq calculation and data uploading are rapidly completed, and then the device returns to sleep. The system is small in size and flexible in deployment, the sleep current is as low as 45 [mu] A or below, noise data can be monitored and reported in real time, and the real-time performance, accuracy and emergency response capability of noise monitoring are improved.

1. Name of the product in this design: Recording and Broadcasting Host. 2. Application of this design: This design is a professional audio and video recording and processing device, which is used in fields that require efficient retention and dissemination of audio and video content. 3. The key design feature of this product is its shape. 4. The picture or photo that best illustrates the key design points: Design 1 3D diagram 1. 5. Design 1 is designated as the basic design.

1. Name of the designed product: professional plug. 2. Use of the designed product: professional audio field. 3. Design points of the designed product: combination of shape and pattern. 4. Picture or photo that best shows the design points: perspective view.

The invention discloses a method for realizing a giant professional audio pointer level meter by adopting an embedded type and a stepping motor. The method comprises an audio signal processing module, an ESP32 acquisition control system module and a stepping motor driving system, the audio signal processing module is used for completing the technical links of amplification, rectification, calibration, charging and discharging time constant control and noise control of audio signals; the ESP32 acquisition control system module is used for realizing signal acquisition and control by adopting an embedded development platform ESP32S3; the function of the stepping motor driving system is that a stepping motor scheme is adopted, the stepping motor driving system is comprehensively matched with an ESP32 supporting platform, accurate pointer driving is achieved, and the stepping motor driving system is matched with a communication mechanism of a motor. According to the technical scheme, the productization of a large-scale audio pointer level meter is realized, and the indicating precision is higher and cannot be realized by a traditional mechanical indicating gauge head.

1. Name of the Design Product: Professional Sound Card (MiCaster). 2. Use of the Design Product: Equipment for professional audio processing. 3. Design Key Points of the Design Product: Lies in the shape of the product. 4. Picture or Photograph that Best Illustrates the Design Key Points: Perspective View 1.

1. Name of the product in this design: Professional Audio Power Amplifier. 2. Intended use of this design: for use in an audio power amplifier. 3. The key design features of this product are the combination of shape and pattern. 4. The image or photograph that best illustrates the design's key points: a 3D model. 5. Other situations requiring explanation: The location indicated by A in the view is the extended power cord and plug, which is a common design and is omitted here as it does not affect the overall appearance protection.

The invention discloses a video content structured processing method and system fusing a voice large model and a visual semantic large model, and relates to the technical field of video processing. Comprising the following steps of: 1, taking an original video file as input, and separating an audio stream from the original video file by adopting a professional audio and video processing tool; performing standardization processing on the audio stream to obtain a standardized audio file; and if the original video file has no audio track, triggering a preset exception handling mechanism and outputting an error prompt. Through an automatic chapter division mechanism based on semantic understanding, original unstructured video content can be converted into structured data with a hierarchical semantic structure, so that the subsequent content retrieval, annotation and reuse process is greatly simplified. Compared with a traditional method depending on manual annotation or fixed time slicing, the method can dynamically generate the structured nodes according to the actual semantic change of the video content, and the organization efficiency and the reutilization rate of video resources are remarkably improved.

The present application relates to an audio transmission method and apparatus, a device, a computer-readable storage medium, and a vehicle. The method is applied to a vehicle, avoids the high cost of use of a professional audio DSP signal processor, and only uses a small portion of the functions of the audio DSP signal processor, thereby reducing the cost of controlling a vehicle motor to produce sound.