155results about "Broadcast information generation" patented technology

Systems and methods are disclosed for emulating an environment created by the outputs of a plurality of devices. The system receives device control data for a device in a first venue. The control of the outputs of said devices according to the device control data creates an environment within the first venue. The system retrieves profile data for devices within a second venue. The system associates a device in the second venue with a device from the first venue, both devices having a similar output type. The system then generates control information adapted from the associated device of the first venue for the device in the second venue. The system controls the outputs of each device in the second plurality of devices according to the generated control information to emulate the environment within the first venue in the second venue.

Methods are provided of three dimensionally synthetizing a digital audio signal depending on spatial commands and filter instructions. Such methods comprise: performing, depending on the spatial commands, a spatialization method at audio rate on the digital audio signal to spatialize it and therefore produce a spatialized audio signal whose reproduction implements, in a three dimensional, 3D, audible space, spatialized virtual sound sources defined by spatial parameters; performing, depending on the filter instructions, digital filtering on the spatialized audio signal to produce an evolved or synthetised audio signal whose reproduction implements, in the 3D audible space, evolved virtual sound sources; and providing the evolved audio signal to a plurality of loudspeakers or headphones to reproduce the evolved audio signal and, therefore, implement the evolved virtual sound sources in the 3D audible space. Systems, computer systems and computer programs that are suitable for performing such methods are also provided.

A transmission system includes encoding and modulation means for encoding and mapping a plurality of frames. Each frame encapsulates a set of information bits of the data traffic, and is associated with a first modulation order and a coding rate, to a plurality of frames of encoded and modulated symbols; physical layer framing means arranged for inserting synchronization symbols taken from a second modulation order into at least one frame of encoded and modulated symbols of said plurality, to obtain a plurality of physical layer frames. The transmission system has expansion means to replace, if the second modulation order is smaller than the first modulation order, at least a subset of the synchronization symbols inserted by the physical layer framing means by symbols from the same constellation as the modulated symbols in the frames of encoded and modulated symbols.

The invention relates to an automatic debugging method, device and equipment for a multimedia sound console and a medium, and the method comprises the steps: obtaining sound field space data, matching a pre-cached basic cloud template according to the sound field space data, obtaining an input audio signal of multiple sound channels, and transmitting the input audio signal to the multimedia sound console; determining a spectrum characteristic and a dynamic range characteristic of the input audio signal; carrying out equalization analysis on the spectrum features, and determining a corresponding equalization parameter set; determining a corresponding gain parameter set according to the dynamic range characteristics; according to the reverberation parameter, the equalization parameter set, the gain parameter set and the filtering parameter set, a corresponding local cache template is established, a cloud template matching technology based on sound field space data is introduced, the limitation that traditional tuning depends on fixed parameters or manual adjustment is overcome, the automation degree of tuning is improved, manual intervention is reduced, and the tuning efficiency is improved. And meanwhile, the stability of audio output quality is ensured.

Examples relate to generating an editable representation of a DJ performance. A computing system accesses at least one data stream representing the DJ performance and analyzes it to identify audio elements within the performance. The system generates a digital representation of the audio elements editable in a digital audio workstation, wherein the digital representation comprises a structured data format including information related to tracks, effects, and timing. The digital representation may include track objects for tracks played, clip objects representing segments of tracks, processor objects representing effects applied to tracks, and tempo objects representing tempo information. The system stores the digital representation, which can be loaded into a digital audio workstation application for editing using tools specific for DJ editing workflows.

The present invention provides a method for processing music audio data, comprising the steps of providing input audio data representing a first piece of music containing a mixture of predetermined musical timbres, decomposing the input audio data to generate at least a first audio track representing a first musical timbre selected from the predetermined musical timbres, and a second audio track representing a second musical timbre selected from the predetermined musical timbres, applying a predetermined first audio effect to the first audio track, applying no audio effect or a predetermined second audio effect, which is different from the first audio effect, to the second audio track, and obtaining recombined audio data by recombining the first audio track with the second audio track.

The sound console comprises a lower shell and an upper cover, the rear side of the top of the lower shell is hinged to the rear side of the bottom of the upper cover, the rear side of the top of the lower shell is provided with a lower connecting piece, the rear side of the bottom of the upper cover is provided with an upper connecting piece corresponding to the lower connecting piece, and the upper connecting piece is connected with the lower connecting piece. The lower connecting piece is provided with a hinge groove with an opening in the top, the upper connecting piece is provided with a hinge column corresponding to the hinge groove, the hinge column is clamped in the hinge groove and rotationally connected with the hinge groove, a notch is formed in the bottom of the hinge column, the upper connecting piece is provided with a limiting plate connected with the hinge column, and the limiting plate is connected with the hinge column. The lower connecting piece is provided with a limiting groove corresponding to the connecting plate. The utility model has the advantages that the structure is stable, the carrying and the use are convenient, the upper cover and the lower shell are firmly assembled together, the upper cover rotates smoothly, the upper cover can be assembled and disassembled at any time through the rotation angle without tools and bolts, the assembly and disassembly are convenient, and the use effect is good.

The invention relates to a mechanical workstation device (1) configured to allow alternation between a first configuration and a second configuration, the device (1) comprising a support (10) having a first group of electronic components fixed on its first main face, the support being suitable for mounting on a support frame (15), a rotation mechanism (13) configured to drive the support (10) in rotation about a main axis (X13) relative to the support frame (15), the device (1) comprising, fixed on its second main face (102), substantially parallel to its first main face, a second group of electronic components (12), such that the rotation of the support (10) about the main axis (X13) allows selective access to one or the other group of electronic components.

The utility model discloses a DJ sound console. The DJ sound console comprises a shell, a control panel, a channel mixer, a mainboard and a meter head, the number of the control panels is multiple, the control panels are vertically arranged in the shell along the length direction of the shell, the control panels are parallel to each other, and the upper end of each control panel is provided with a knob, a toggle switch, an indicator light and a button which are used for adjusting the frequency band and optimizing the tone; the channel mixer is arranged on the inner wall of the front end of the shell, and multiple input / output channels are formed in the channel mixer and used for being connected with external equipment; the main board is arranged in the shell, and the control panels and the channel mixer are electrically connected with the main board; the meter head is arranged on the top surface of the shell, and the meter head is electrically connected with the mainboard.

An electronic device comprising circuity configured to determine, for a track selected from a set of tracks, a perceptual information loss related to removing the selected track from the set of tracks.

Disclosed herein is a method of determining a plurality of inter-track auditory masking measurements between a plurality of audio tracks, the method comprising: dividing each audio track into a first set of frequency bands; and determining, for each frequency band of each audio track, an inter-track auditory masking measurement that is a measure of the masking between the frequency band of the audio track and a combination of the corresponding frequency bands of all of the other audio tracks; wherein each inter-track auditory masking measurement is determined in dependence on a psychoacoustic model.

To provide an audio mixer which can be easily used when using one mixer by a plurality persons.SOLUTION: An audio mixer contains an operation panel having a first division containing a first operator for operating a first parameter and a second division containing a second operator for operating a second parameter. A control part divides a plurality of input channels into a first input channel and a second input channel, and divides a mixing bus into a first mixing bus 303A and a second mixing bus 303B. A processing part is controlled so as to be operated separately in a first signal processing system and a second signal processing system. In the first signal processing system, a sound signal output from the first input channel to the first mixing bus in accordance with the first parameter is processed. In the second signal processing system, a sound signal outputting from the second input channel to the second mixing bus is processed in accordance with the second parameter.SELECTED DRAWING: Figure 7

To provide a spot production data generation device that generates spot production data capable of replacing a part of a voice of a spot.SOLUTION: A spot production data generation device 1 includes an audio data coupling portion 10 that inputs one or more pieces of audio data corresponding to each predetermined time interval of a spot for each time interval, adjusts the audio data to the same number of channels in all time intervals, and generates combined audio data that is combined in the time direction, an audio metadata generation portion 20 that inputs a list of audio data for each time interval for each time interval and generates audio metadata that specifies the playback position of the audio data in the combined audio data, and a data integration portion 30 that integrates the combined audio data and acoustic metadata into a single file to generate data for spot production.SELECTED DRAWING: Figure 1

The utility model discloses an elastic compression portable sound console, and belongs to the field of sound console equipment. Comprising a base, a bottom rotating groove is formed in the rear surface of the base, a connecting bottom block is arranged in the bottom rotating groove, bottom rotating columns are rotationally connected to the two ends of the connecting bottom block and the inner side wall of the bottom rotating groove respectively, a multifunctional panel is arranged above the base, and a plurality of tuning sliding grooves are formed in the upper surface of the base; the interior of the tuning chute is slidably connected with a push control block, and the front surface of the multifunctional panel is fixedly connected with a plurality of adjusting knobs. The device has excellent portability, and can fully meet diversified use scene requirements of users. Due to the unique foldable design, the multifunctional panel can rotate upwards around the bottom rotating column to the inclination angle facilitating operation and can rotate downwards to be attached to the base after being used, and the overall size is greatly reduced.

To efficiently and accurately exclude a specific slave station belonging to a group from group broadcasting targets in broadcast communication.SOLUTION: A broadcast communication device 40 is an information processing device of a master station for performing broadcast communication on a group 50 including one or more slave stations 51, and includes an exclusion target slave station for excluding each of slave stations 51 belonging to the group 50 from broadcast communication targets for the group 50 on the basis of, inappropriateness information 410 representing inappropriateness of performing the broadcast communication on an individual slave station 51, a determination unit 41 for determining either a non-exclusion target slave station that is not excluded from the broadcast communication targets for the group, and a communication unit 42 for performing broadcast communication on a slave station 51 which is a non-exclusion target belonging to the group 50.SELECTED DRAWING: Figure 9

A system includes Memory, Display and Processor. Memory stores application with executable instructions. Processor performs operations: retrieving schedule information for channels to be transmitted simultaneously by a broadcaster, the information includes start and end times for types of content to be transmitted; identifying, from the information, first time segments in which a first content type is to be transmitted and second time segments in which the first type is not to be transmitted; displaying a plot of the information with content types transmitted over time for each of the channels, the first segments corresponding to the first type are displayed as a first color and the second segments are displayed as a second color; identifying, from the second segments for each channel, transition windows comprising durations within which all of channels are scheduled to not transmit the first type; and overlaying on the plot an indicator of the windows.

To provide an information processing device suitable for switching and distributing videos from a plurality of video sources including a network camera via HTTP streaming.SOLUTION: An information processing device 100 for performing information processing for distributing a video content using, as raw materials, videos of a plurality of video sources including a network camera 1500 includes video source thumbnail display means for generating and displaying a video source thumbnail based on a video signal transferred by a transfer method in which a node for transmitting the video source transfers data without confirming reception by a receiving node, and a switching means, and further includes play list generating means for generating a play list based on information regarding which video source was switched at what timing, and information processing means for distributing the video content to a viewer terminal 1600 by a transfer method in which the same data is repeatedly re-transmitted until a reception notification of the receiving node is confirmed.SELECTED DRAWING: Figure 4

To provide a digital mixer capable of displaying / operating / changing a plurality of channels and a plurality of parameters at the same time.SOLUTION: A digital mixer 10 comprises: an input interface 12; a processor 14; an output interface 18; at least a first display 20 and a second display 21; and an operator 22. The operator 22 contains a mode switching button for switching between a first mode and a second mode. The processor 14 displays parameter setting screens of audio signals in different channels on the first display 20 and the second display 21 respectively when the mode switching button is set to the first mode, and displays parameter setting screens of the different audio signals in the same channel on the first display 20 and the second display 21 when the mode switching button is set to the second mode.SELECTED DRAWING: Figure 1

The method uses an automatic system of electrical / electronic equipment to synchronize voices, instruments, videos, and light effects. It consists of four phases that significantly reduce the need for sound, video, and lighting technicians, allowing venue owners, presenters, and performers to independently manage events and shows. Through a system of references, artists can customize the sound by amplifying or excluding instruments of interest, adding prerecorded sound effects, and connecting live instruments. Additionally, a pitch recognition system supports the lead voice, optimizing it for perfection. The method is also ideal for amateur singers, allowing them to perform with a live band and improve their performance with images and choreographic effects, further enhancing the overall show experience.

To be capable of preferably performing a series of media access controls on a reception side.SOLUTION: A container of a predetermined format including a media stream is transmitted. A predetermined number of sets of media access information associated with a series of media access controls are successively inserted in a layer of the media stream or a layer of the container. For example, the media access information includes identification information for making a distinction between other media access information, and identification information for making an association with the other media access information.SELECTED DRAWING: Figure 16

The present invention provides an audio processing method in which an operator can perform audio processing of a distribution source in the same conditions / environment as other terminals receiving content distribution. One embodiment relates to an audio processing method in which a distributor terminal used by a distributor of content is connected to an audio device that performs audio processing related to the content, the distributor terminal performs distribution of the content to one or more listener terminals, the distributor terminal specifies an operation-use listener terminal that performs a change operation of an audio processing parameter of the audio device among the one or more listener terminals, the operation-use listener terminal accepts the change of the audio processing parameter, the operation-use listener terminal transmits the changed audio processing parameter of the audio device to the audio device via the distributor terminal, and the audio device performs audio processing based on the audio processing parameter received from the operation-use listener terminal.

The invention relates to a mixing console device with at least two input channels, at least one output channel, and a signal processing unit configured for processing signals in the input channels and outputting the processed signal in the output channel. The mixing console device has a selection area in which at least two of the input channels can be user-selected, and an interface area associated with the selection area, which has at least one control element for influencing the selected input channel and which is variable depending on the selected input channel.

Systems and methods for efficient and reliable cross-platform impression availability across linear and digital streaming content platforms is provided. Specifically, impressions estimates from both linear content platforms (102) and digital content streaming platforms (104) are used to identify relative capacities for satisfying a campaign of placement of the supplemental content. The relative capacities may be used to balance a portion of the capacities across the platforms that are used to satisfy the campaign.

An exemplary method includes a processor executing instructions stored in a memory, the instructions for generating an electronic count, binding the electronic count to a first performance to generate a master clock, and transmitting a first performance by a first musician and first timing information to a network cache, storage, timing, and mixing module. The first performance by the first musician can be recorded locally at full resolution and can be transmitted to a full resolution media server, and the first timing information can be transmitted to the master clock. The first performance by the first musician is transmitted to a sound device of a second musician, and the second musician composes a second performance, which and the second timing information are transmitted to the network cache, storage, timing, and mixing module. The first performance and the second performance are mixed together with the first timing information and the second timing information to generate a first mixed audio, which can be transmitted to the sound device of a third musician.

The present invention relates to a transmission system arranged for generating data traffic to be transmitted. The transmission system comprises : - encoding and modulation means for encoding and mapping a plurality of frames whereby in each frame a set of information bits of the data traffic is encapsulated, each frame being associated with a first modulation order and a coding rate, to a plurality of frames of encoded and modulated symbols, - physical layer framing means arranged for inserting synchronization symbols taken from a second modulation order into at least one frame of encoded and modulated symbols of said plurality, so obtaining a plurality of physical layer frames. The transmission system is characterized by comprising expansion means to replace, if the second modulation order is smaller than the first modulation order, at least a subset of the synchronization symbols inserted by the physical layer framing means by symbols from the same constellation as the modulated symbols in the frames of encoded and modulated symbols.

The present invention relates to a transmission system arranged for generating data traffic to be transmitted in a satellite communication system. The transmission system comprises : - encoding and modulation means for encoding and mapping a plurality of baseband frames of information bits, each baseband frame associated with a modulation and coding type, to a plurality of frames of encoded and modulated symbols, - generator means to generate one or more values to determine a superframe length, - framing means arranged for inserting synchronization symbols and preambles into frames of the plurality of frames of encoded and modulated symbols, so obtaining a plurality of superframes, each preamble indicating the start of a superframe, - spreading means arranged to receive the plurality of superframes (or a version thereof), and to spread each symbol in the plurality of superframes with a spreading sequence to a chip sequence, so obtaining a plurality of spread superframes, wherein the generated one or more values determine for a current superframe a frame length different from the frame length of the preceding superframe.

An information processing device suitable for switching between images from a plurality of image sources and outputting image content is provided. [Solution] An information processing device that performs information processing to output video content using video from multiple video sources including at least one tilt-controllable camera as material has a tilt control means that, based on the "size" of the image of the video source that is the video of the switched video source (PGM) and the length from the top of the person's head to the top of the screen, tilts the camera so that the length from the top of the person's head to the top of the screen is shorter than that of the PGM when the "size" set for the tilt-controllable camera is tighter than the "size" of the PGM, and tilts the camera so that the length from the top of the person's head to the top of the screen is longer than that of the PGM when the "size" set for the tilt-controllable camera is looser than the "size" of the PGM.

The invention relates to a public broadcasting method and system. The method is applied to a remote control forwarding terminal of the public broadcasting system, the public broadcasting system further comprises at least one remote controller, playing equipment and a broadcasting server, the playing equipment and the broadcasting server are connected through a network, and the remote control forwarding terminal is in communication connection with the server through an intranet. The method comprises the following steps: receiving a remote control signal sent by a remote controller completing frequency matching; and generating and sending a key signal corresponding to the remote control signal to a broadcast server, so that the broadcast server compares the key signal with a pre-stored instruction, and drives a playing device to play broadcast information corresponding to the key signal. According to the invention, public broadcasting can be rapidly realized, the inherent defect that the infrared remote control technology is limited by short operating distance can be broken through, and the method is especially suitable for scenes needing wide area coverage, such as large stadiums and campuses, so that the timely response speed of public broadcasting in large-scale places can be improved.

The invention discloses a balance adjustment method and system for a sound console volume controller, and belongs to the field of sound console control. The method comprises the steps that audio signals and environment noise are collected and preprocessed; audio state perception and noise identification are carried out by using a deep learning model combining a time sequence convolutional neural network and a long short-term memory network; a volume control and frequency equalization dual-channel parallel computing architecture is adopted, and controlled variables are generated based on a first-order active-disturbance-rejection control algorithm and a second-order active-disturbance-rejection control algorithm; using an improved particle swarm optimization algorithm to carry out collaborative optimization on the dual-channel control parameters; and finally synthesizing and outputting a high-quality audio signal. The system correspondingly comprises an audio acquisition module, a deep learning reasoning module, a dual-channel control module, a parameter optimization module and the like, and can be integrated with a hardware acceleration unit. The method effectively solves the problem of coupling of volume and frequency balance control, and has the advantages of high environmental noise adaptive capacity, intelligent parameter adjustment, high processing real-time performance and the like.