42 results about "Professional audio" patented technology

The present invention provides a method of processing musical information. A user can perform a musical piece, using a real musical instrument (e.g. a guitar), which is received as audio input. The audio input is then assessed to determine whether it meets various quality standards—for example, whether the user played at the right pitch, at the right time, or at the right volume. If the audio input meets the standards, audio output is then provided of a professional performance of the musical piece, such that it sounds as if the user is performing the professional audio. The standards can be adjusted to different levels, depending on the user's skill on the musical instrument. At an easy level, low standards may be applied, such that even unskilled or beginner musicians can sound like a professional. For more skilled users, the standards may be more difficult to meet. If the user does not meet the quality standards, alternative audio output may be provided, indicating incorrect performance of the musical piece.

The invention relates to a loudspeaker mounting and adjustment system and method for installing and operating a professional audio system used in a stadium, concert hall or the like. The system comprises a vertical array of loudspeaker cabinets that can be suspended from a ceiling and that enables the horizontal and vertical angle of the sound dispersion field to be adjusted remotely and / or automatically while the system is suspended. Each loudspeaker cabinet is connected to a vertically adjacent loudspeaker cabinet via a pair of levers located on either side of the cabinet that control the angle between adjacent loudspeaker cabinets. A linear actuator is connected to each lever for controlling the lever position. Each loudspeaker cabinet also comprises a waveguide with at least one actuator for modifying the waveguide angle and thus the horizontal angle of the sound dispersion field.

A computer based system for selecting digital media frames is capable of predicting the frames that are to be subject to a subsequent action. The subsequent action could be the selection of the predicted frames for inclusion to create a new set of frames consisting of the selected frames; it could also be the selection of the predicted frames for exclusion to create a new set of frames consisting of the frames but now excluding the selected frames. Because the system automatically predicts the frames that are, for example, to be included or excluded in a new clip, this removes the need for the user to manually define start and end frames. Instead, the user merely has to accept the predicted frames or refine the predicted selection. This is far quicker and requires less complex user interaction; these are very important advantages for a system designed for ordinary consumers, as opposed to professional audio or video editors. The system hence finds particular application in consumer oriented devices such as laptop computers, mobile PDAs with wireless connectivity, mobile telephones, set-top boxes; hard-disc based personal video recorders (PVR).

The invention discloses a high-power professional audio power amplifier based on a Peltier effect radiation technology, comprising a box body, a left power amplification channel and a right power amplification channel. The left power amplification channel and the right power amplification channel are arranged in the box body and respectively provided with at least one power amplification chip, a semiconductor heat radiation sheet and a heat radiator which are fixed together from front to back, the semiconductor heat radiation sheet is provided with a hot end surface and a cold end surface, the hot end surface of the semiconductor heat radiation sheet is pasted on the heat radiator to from hot joint, the back of the power amplification chip is pasted with thermally conductive silicone and pasted on the cold end surface of the semiconductor heat radiation sheet, and outer heat radiation fans capable of taking away heat which is transferred to the heat radiator are respectively arranged in positions corresponding to the left power amplification channel and the right power amplification channel on the rear backboard of the box body. The invention has good heat radiation performance and high efficiency.

For use in professional audio amplifier applications, high current charge pump circuits including charge circuits and dynamic release circuits are connected between respective storage capacitors and the audio power amplifier output. The charge circuits charge their storage capacitors when an output signal swing of the power amplifier is below a corresponding predetermined positive and negative threshold. The dynamic release circuits release power from their charged storage capacitors to dynamically vary the voltage levels of their positive and negative power supply rails in direct relation to the output signal swing of the power amplifier, preferably at unity gain. Multiple positive and negative power supply rails at different voltage levels can be used with cascaded multi-stage dynamic release circuits to maximize system efficiency. The charge and dynamic release circuits can be connected between their storage capacitors and a buffer amplifier which also receives the power amplifier audio input.

The invention discloses a method for generating dynamic data for a dynamic seat based on audio editing. The method includes the steps of producing basic motion waveform database, importing a movie file through professional audio and video editing software, adding a motion rail for each freedom degree of the dynamic seat, editing the dynamic data as audio data, dragging and dropping basic motion waveforms on the motion data rail by a technician according to the needs of movie plots on the video rail, editing motion data with audio editing tools, playing selected editing sections, intercepting motion data from the audio data to control the movement of the platform through computer software, commissioning the motion effect, and storing motion data in the movie file. The method has the advantages of convenient editing, being direct, high efficiency, convenient commission and excellent synchronism, dynamic data producing time is substantially shortened, and the final experience effect of the dynamic seat is improved.

The invention relates to the technical field of communication equipment, and specifically relates to a multimedia dispatching desk. The multimedia dispatching desk comprises a base, a support and a host installed on the support; the host is provided with a high-definition display screen and interphones arranged at the two sides of the high-definition display screen; and a central console is integrally arranged below the host, and a high-performance control panel is arranged inside the central console. The multimedia dispatching desk provided by the invention is highly integrated with equipmentsuch as a capacitive touch control operation screen, a professional handle, a professional loudspeaker, a high-definition camera and the like, and can be perfectly combined with a large-screen display device, sound equipment, central control equipment and so on deployed in a command center, so that a professional audio and video command and dispatch operation can be easily realized; meanwhile, the multimedia dispatching desk has a professional voice bus design, and supports multiple voice devices to access a system at the same time; besides, concise and efficient control can be realized through a control button; and the multimedia dispatching desk also has a noise reduction function, and can achieve multiple forms of dispatching functions, such as push-to-talk, secret talk and loudspeaking and so on.

A dipole radiating dynamic speaker design comprises a diaphragm (32) providing a surface for radiating acoustic output, a voice coil assembly (36) for driving the diaphragm, and a magnet system for providing magnetic flux across the voice coil assembly. The speaker incorporates a vented pole flux conductor (42) and a vent adjacent to the diaphragm permitting the rearward acoustic output to propagate away from the diaphragm producing acoustic output as a dipole radiator.A method of utilizing the dipole radiating dynamic speaker design for tweeter and mid-range speaker applications used in home, automotive, and professional audio systems is disclosed. In addition a novel enclosure design is disclosed that utilizes a front panel perimeter vent providing improvement of audio reproduction quality for mid-range speaker applications.

The invention relates to a professional audio recording-playing device which adopts a CF card as storage medium, comprises an embedded CPU and is characterized in that the embedded CPU is connected with the CF card; the embedded CPU is also connected with a filed programmable gate array; and the filed programmable gate array is connected with an input-output port through an audio circuit. The device has the advantages of completely getting rid of the open-reel drive which has large volume and is inconvenient for use; a recorder of the CF card has small volume, can reach high audio index and can completely satisfy the requirements of the professional audio recording and playing; furthermore, the CF card has small size, thus being easy to store.

The invention discloses a system for test the intensity of terminal audio signal, including terminal, audio signal acquisition device and audio signal processing device. Among them, the terminal is used for broadcasting audio signals. Audio signal acquisition device collects the electric signals between the load of power amplifier and outside amplifier of terminal, and sends to the audio signal processing device, which gets the strength of the terminal audio signal according to the electric signals. The invention also discloses a method of testing the intensity of terminal audio signal by audio signal acquisition device and audio signal processing device. The application of this invention can avoid the subjective judgment error due to the difference of people's hearing when using hearing to test the terminal audio signals, thus it's more accurate for testing the audio intensity of terminal, raises the efficiency of test, and greatly reduces the testing cost compared with the professional test instrument.

The invention discloses a three-in-one audio transmission control system, mainly comprising an audio set controller, a network adapter, a network wireless terminal, an audio set and a wireless host. The three-in-one audio transmission control system is characterized in that a stereo sound source and microphone speech level are mixed and transmitted by the audio set controller to the network adapter in the transmission mode of an analog signal, the network adapter carries out A / D(Analog to Digital) conversion on the analog signal, a digital level signal is subjected to TCP / IP (transport control protocol / internet protocol) communication protocol compiling, and a compiled Packet is broadcasted to a whole LAN (local area network) through a network connecting wire. According to the three-in-one audio set transmission control system, a digital network technology, a radio technology, and a professional audio set and public broadcasting technology are unprecedentedly integrated, and all demands of users on the audio set are solved to the maximum extent.

Current microphone designs are still using concepts developed in the early 20th century. A transducer followed by a simple buffer with high impedance, low power, and low gain. Even when tubes were replaced by solid state devices the same practice continued, with the low gain triode being replaced by an even smaller power, lower current FET buffer. The invention offers a microphone system intended for professional audio applications that dispenses with these, low power concepts and incorporates elements of a microphone, microphone pre amplifier, and high power mixing console / pre amp line drivers into a single housing. By combining transducer, high fidelity pre amp circuit, and high power line driving stage with a high voltage, high current external power supply, the capabilities of microphones can be greatly increased in many ways. Our approach can be implemented in a variety of circuit topologies including discrete transistors, tubes and integrated circuits, our concept allows microphones to connect directly to analog or digital recording devices without the need for external mixers or pre amplifiers. The design not only allows for unprecedented simplicity in the work flow of the modern recording studio, which combines the use of stand-alone analog records, standalone digital recorder, and computer workstation based recording options, but also eliminates redundant circuits that are present in microphones, microphone pre amps, and mixing consoles, while offering the customer increased value.

A system for auditing performance data in commercial venues having one or more computing devices located in venues where live or pre-recorded music is performed. The computing devices capture and store content in each venue using at least one audio input configured for professional audio performances. The computing devices store the content as a content data record with associated identifiers for date, time, and venue location. The content data records are transferred from each venue to a central server. Encryption keys encrypt the content data records from each venue to verify the authenticity of the content transmitted from the venues to the server. The server includes a processor for opening, reviewing, and separating the content data records from each venue into distinct files, each file comprising an individual work performed in the venue. The system further includes an interface for providing access to the archived content data records at the server.

The invention relates to the technical field of power amplifying circuits, in particular to a professional audio amplifier. The professional audio amplifier comprises a power preamplifier and a power post-amplifier, wherein a built-in equalizer is connected between the power preamplifier and the power post-amplifier; the built-in equalizer is provided with a signal-through electronic switch; the professional audio amplifier further comprises a music signal detection and analysis circuit, and an electronic switch controller for controlling the signal-through electronic switch. During the use, the professional audio amplifier can modify the sound quality through the built-in equalizer according to characteristics of a loudspeaker box so as to achieve an effect that the professional audio amplifier is perfectly matched with the loudspeaker box; moreover, through the music signal detection and analysis circuit, the type of an input signal can be detected for conduction of corresponding control, so that an equilibrium curve cannot be measured with a conventional testing method,and a commercial information enciphering purpose of hiding the equilibrium curve is achieved; meanwhile, all indexes are in accordance with national and international standards; furthermore, the professional audio amplifier is provided with a variety of protection circuits, so that the professional audio amplifier has a perfect protecting function.

The invention relates to a method for the recording of an audio signal constituted by multiple tracks and one or more channels, in a digital data format on a single track of a recording support of digital information; as well as to a method for the reproduction of such a format which also allows the manipulation of the plurality of tracks which form the audio signal recorded in such a format on a single track of said recording support. A field of application is in the professional audio sector, for the disk jockeys (DJs) engaged in the mixing of songs in dance halls and discotheques, facilitating and making the performances of these professionals more creative, through the invention allowing them to manipulate simultaneously different digital tracks of instruments and / or vocals of a song independently and in real time. It is also applicable in the domestic audio environment (karaoke, etc.).

The invention discloses a supreme vision correction technology. The supreme vision correction technology comprises the following steps: (1) carrying out psychological counseling; (2) under the condition that a vision corrector closes eyes, guiding the vision corrector to continuously recall objects established during growth and processes for gradually mastering each skill and continuously overcoming bottlenecks during learning, and establishing an eyeglasses pick-off belief for the vision corrector; and (3) creating Alpha waveband audio frequency by using a special audio-frequency apparatus. According to the supreme vision correction technology, a vision problem is caused by eyeball strain, ametropia can disappear by using an eyeball easing method, thus, eyeglasses pick-off is achieved, and a betz theory for recuperating vision and a full-brain technology for omni-directional brain development are combined, so that the vision correction of an adolescent is achieved, drugs, medical appliances and the like are not required, excessive eye use is not required, vision is corrected in an easy and safe manner, the vision correction is efficient and stable, and thus, the supreme vision correction technology is applicable to extensive popularization and use.

The pretransmitted specification provides a device and equipment for amplifying an audio signal. The device comprises an amplification unit, a feedback unit, a detection unit and a voltage control unit, wherein the amplification unit is used for amplifying an audio input signal and outputting an audio output signal; the amplification unit comprises a first input end and a second input end; the feedback unit is used for amplifying the audio output signal based on a preset amplification parameter and outputting a generated feedback gain signal to the detection unit; the detection unit is used for carrying out detection processing on the feedback gain signal to obtain an envelope signal; and the voltage control unit is used for adjusting input resistance of the second input end of the amplification unit according to the amplitude of the envelope signal, so as to adjust the amplification gain of the amplification unit. According to the device for amplifying the audio signal, the amplification gain of the amplification unit can be automatically and rapidly adjusted, the control principle is simple, the circuit is stable, the cost is low, and the device is very suitable for being appliedto non-professional audio experience occasions such as vehicle-mounted handsets and handheld interphones.

The invention discloses an audio-video error source checking method based on a GStreamer frame. During audio and video file playback, by configuring the dump control switch, real-time downloading of audio and video files being played is realized, when there is an abnormal playback phenomenon during the playback of audio and video files, and then the film source downloaded from the dump is copyed to the PC terminal for playing with the mainstream player, if the playback effect is the same as the product playback effect, that is to say,blurred screen occurs, you can preliminarily determine thatthis is a problem with the film source itself, the audio and video source can be analyzed by professional audio and video analysis tool software, thereby further confirming whether the playback problem is caused by the source problem. The method of the invention can help the developer of the player to quickly locate the root of the problem such as playing the blurred screen and the like, avoid the problem of erroneous analysis direction when the developer analyzes the root of the problem, cause the waste of time, and effectively improve the efficiency of solving the problem of playing the audio and video files.

An audio frequency variable-impedance instrument-level to professional-level audio interface with balanced DC power, comprising a unique arrangement of electronic elements, consisting of the musical instrument-level signal splitter with professional-level audio output interface 10, the professional-level signal splitter to instrument-level interface 12 and the power supply with regulation, innovative new DC balancing and power distribution 14. The innovation of balanced DC power effectively cancels the unwanted power supply noise on professional audio electronics giving greatly improved distortion and lower noise specifications over previous designs.

The invention relates to a field broadcasting system. The field broadcasting system comprises a mobile phone APP module, wherein a user logs in a main page of the mobile phone APP module through scanning, and different playback information is displayed according to activity types through playback classification of the main page; a user PC end module having the same interface and function as the mobile phone APP module, wherein a user can synchronously watch videos on the user PC end module and the mobile phone APP module; and an operation background module providing an information interface for the mobile phone APP module and the user PC terminal module, wherein the operation background module generates a look-back two-dimensional code, sets whether a user can download the two-dimensionalcode, sets video content time limit, prints the generated corresponding two-dimensional code on an admission ticket or projects the generated corresponding two-dimensional code on a large screen released on site. According to the invention, the problems of selfie behavior, poor video picture effect and poor tone quality effect existing in live broadcast of visitors are solved, and professional audio and video records at the optimal shooting position are provided.

The invention belongs to the technical field of professional audios and videos, particularly relates to a hybrid networking system and method for multi-rate network port distributed nodes, and aims to solve the problem of how to send and receive video streams in a system in which the multi-rate network port distributed nodes coexist. The system comprises a management server, a receiving node, a sending node, a display device and a video source, wherein the management server manages the sending node to open / close multicast sending, configures a multicast address, sets bandwidth occupied by a video stream and configures the receiving node forming the same large screen; the sending node configures a preset bandwidth reached by the compressed video data, pre-compresses the video data according to the preset bandwidth through a compression algorithm, evaluates the compressed data volume, and selects a corresponding compression parameter to compress the video data to obtain the compressed data volume; according to the invention, a plurality of receiving nodes and sending nodes with different network port types can normally send and receive the video streams in a distributed splicing system, packet loss caused by the fact that the video streams exceed the network port capability of the nodes is prevented, and the quality of the video streams received by the receiving nodes is ensured to be the best as much as possible.

A digital stethoscope has an earpiece microphone (4) connected acoustically with flexible tubing to a Precordial Hockey-Puck sensor (1) and connected electronically to a Pro-audio Computer (8) with real-time software to display, control monitoring and record vital signs of a patient undergoing medical procedures. The patient is acoustically isolated from the computer. The user interface can select or de-select optional functional real-time code blocks.

The invention relates to a sound box control technology, in particular to a calibration method for a stage, film and television professional audio system. This method solves the multi-layer adjustable problem at the system level by using active speakers with array layer DSP parameters and introducing speaker models to virtualize multi-layer DSP parameters. The speaker model is adjusted synchronously. All in all, this technology can not only simplify the difficulty of audio system debugging and calibration, reduce the workload, but also ensure better sound effects during system debugging.

The invention discloses an echo noise reduction method through the multi-dimension technology. The realization process of the echo noise reduction method comprises the steps that the sound source azimuth is located according to the number of audio acquisition equipment, wherein the audio acquisition equipment refers to at least two acquisition devices; the transmission process of sound waves is simulated according to the geometrical characteristics of a multidimensional space, and echo data transmitted in the delayed way are processed by using the multi-dimension technology; clutter signals are intelligently filtered through the geometrical characteristics, and synthetic processing of audio signals is accurately controlled; and the synthesized audio data are exported. Compared with the methods in the prior art, the special requirements for the acquisition equipment are not required in the whole noise reduction process so that the echo noise reduction method through the multi-dimension technology has the advantages of being low in cost, easy and convenient to arrange and low in the requirement for the professional level of users in comparison with professional audio acquisition equipment; and the method is high in practicality and wide in range of application so that the technology can be widely applied to VoIP products of various platforms and plays an important role in enhancing conversation quality and service quality and thus the method is easy to popularize.