21 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.

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.

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.

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.

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. 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 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 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.

The invention relates to the technical field of digital audio, and discloses a high-bit-rate audio stream adaptive playing method and system, a storage medium and equipment, and the method comprises the steps: initializing a playing system, building a multi-layer buffer architecture, and creating a connection pool of a network connection handle; adding a timestamp to each data packet, calculating the code rate of the current high-code-rate audio stream, and dynamically updating playing engine parameters; a connection handle is distributed for the network request, and request parameters are optimized and concurrency control is implemented according to the high code rate characteristic of the audio stream; establishing a cooperative working mechanism among all levels of buffer areas in the multi-layer buffer architecture, and dynamically adjusting the target depth and the data preloading strategy of the buffer areas based on the network condition and the playing state which are monitored in real time; according to the method, the system playing process is managed through the state machine, multi-level detection and recovery processing aiming at various abnormities is executed, the smooth high-quality and high-bit-rate audio playing experience is provided through the method, and the strict requirements of professional audio application are met.