311results about "Broadcast information generation" patented technology

A collaborative cloud-based digital music composition, performance and production studio system network employing machine-intelligent digital audio workstation (DAW) systems that are supported by cloud-based automated music composition, performance, production and publishing services. Such services enable improved workflows and enhanced productivity, while ensuring that the music IP rights of all parties involved in the AI-assisted music creation process are respected and responsibly managed in the best economic interests of individual artists, performers, producers, publishers and consumers alike. The digital music studio system network can be realized in various ways supporting the DAW systems in ways suited to different user preferences and application environments.

A method comprising: at an importer of an IBOC digital radio broadcasting system: establishing a data link to an audio client configured to buffer audio from an audio source over a buffering duration, to produce buffered audio; receiving, from the audio client, an indication of the buffering duration; identifying a logical channel of an IBOC waveform assigned to the audio client and configured to transmit PDUs of the audio at a PDU rate; computing a quantized number of PDUs that is greater than one into which the buffered audio is to be divided at the audio client based on the buffering duration and the PDU rate; sending, to the audio client, an indication of the quantized number of the PDUs; and upon sending importer data requests to the audio client over the data link, receiving, from the audio client, client data responses each including the quantized number of the PDUs.

Techniques are disclosed herein for providing audio enhancement and optimization of an immersive audio experience. Examples may include generating an audio feature set for a transduced audio stream captured in an environment, inputting the audio feature set to a neural network model configured to generate an audio isolation mask associated with the transduced audio stream, and generating isolated audio for the transduced audio stream based at least in part on the audio isolation mask.

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.

The invention discloses railway train dispatching four-wire access equipment and an implementation method. The device comprises a first network transformer, a second network transformer, an audio coding and decoding chip, a modular connector, a PHY chip and a processor chip. The equipment is connected with a plurality of 450M analog train dispatching station radio stations through four-wire interfaces, and is connected with a digital train dispatching communication system central server through an Ethernet interface, so that the traditional 450M analog train dispatching station radio stations can be accessed to a digital train dispatching communication system, and dispatching communication and network management monitoring of the digital train dispatching system and the 450M analog train dispatching station radio stations are realized; a technical means is provided for upgrading transition compatibility of a railway wireless train dispatching system, and the method has important significance for guaranteeing railway train operation safety and improving transportation operation efficiency.

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 method comprises: receiving a first audio stream that conveys audio content and a second audio stream that conveys the audio content and is delayed relative to the first audio stream by a time delay; one-sided filtering first audio segments of the first audio stream to pass only positive frequencies of the first audio segments to first filtered audio segments; one-sided filtering second audio segments of the second audio stream to pass only positive frequencies of the second audio segments to second filtered audio segments; cross correlating the first filtered audio segments against corresponding ones of the second filtered audio segments, to produce cross-correlation results; detecting a peak indicated by the cross-correlation results; and estimating the time delay based on a position of the peak, to produce an estimated time delay.

Methods and systems for dynamically routing pulse-code modulated (PCM) audio subchannels to one or more sink devices using a software-based router. The PCM router may set a pre-conversion buffer in the conversion path to a sink device for a particular subchannel if the source PCM fragment size differs from the sink fragment size. It may also compare characteristics of the audio subchannel to the sink device to identify a mismatch and, on that basis, configure the conversion path to include a sample rate converter, a channel converter, or a format converter, as needed. Subchannels can be dynamically routed to two or more sink devices and routing may be changed during playback.

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.

The invention discloses a sound console digital audio signal processing method and system, a product and a medium, and relates to the field of voice or sound processing, and the method comprises the steps: obtaining a plurality of paths of audio input signals, carrying out the time-frequency domain conversion, analyzing the frequency distribution rule, extracting the fundamental frequency and overtone structure features, and building a tone feature fingerprint database. Then obtaining a preset standard audio signal, extracting a multi-dimensional feature vector, and matching the multi-dimensional feature vector with the fingerprint database to obtain a feature parameter group; on the basis, a coordination index system including the overall sound coordination degree, the sound level definition and the spatial sound image balance degree is constructed. A coordination index of an input signal is calculated in real time and compared with a standard signal, and an adjustment value is obtained. And finally, in combination with fingerprint database matching features, calculating a target audio processing parameter combination containing balanced, dynamic and spatial parameters. By implementing the method, the sound quality of multiple paths of audio signals can be improved.

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 invention discloses a digital sound console sound tuning data intelligent processing method and system based on an ARM processor, and belongs to the technical field of data processing. The method and the device are used for solving the technical problems that different tuning data of the digital sound console cannot be supervised and analyzed in different aspects, and subsequent implementation of different tuning cannot be dynamically optimized and managed in different aspects in an existing scheme. Monitoring, counting and processing the whole process of different tuning request processing in the operation process of the digital sound console to obtain tuning single supervision sequences corresponding to different tuning processing, and processing and analyzing local tuning processing states and local tuning support states in different aspects on the tuning single supervision sequences obtained through processing; according to all tone tuning single supervision processing analysis sequences obtained by target tone tuning history processing, processing analysis of an overall tone tuning processing state is carried out, investigation and verification of influence factors are carried out on a processing analysis result, and optimization management is carried out on subsequent different aspects of target tone tuning.

An audio mixer includes an operation panel that includes a plurality of physical controllers that control a value of a parameter. The audio mixer has a signal processor that processes an audio signal to be outputted from a plurality of input channels to a plurality of mixing buses according to the value of the parameter, and a processor that controls an operation of the signal processor. The plurality of physical controllers include a first physical controller that controls a value of a first parameter and a second physical controller that controls a value of a second parameter. The processor, in a first mode, divides the plurality of input channels into a first input channel of a first signal processing system and a second input channel of a second signal processing system different from the first signal processing system, divides the plurality of mixing buses into a first mixing bus of the first signal processing system and a second mixing bus of the second signal processing system, controls the signal processor to perform a first operation to process an audio signal to be outputted from the first input channel to the first mixing bus according to the value of the first parameter in the first signal processing system, and controls the signal processor to perform a second operation to process an audio signal to be outputted from the second input channel to the second mixing bus according to the value of the second parameter in the second signal processing system, in a second mode different from the first mode, controls, by using the plurality of input channels and the plurality of mixing buses as part of the same signal processing system, the signal processor to perform a third operation to process an audio signal to be outputted from the plurality of input channels to the plurality of mixing buses according to the value of the first parameter and the value of the second parameter in the same signal processing system.

To provide an audio signal processing device capable of shortening a setting time related to setting data for each channel.SOLUTION: In an audio system that has a mixing device and an audio signal processing device, an audio signal processing device 3 that is connected comprises: an operation acceptance unit 37 that accepts from a user a selection operation for selecting a channel of the mixing device; an audio signal input unit 32 that receives an audio signal to the channel accepted by the operation acceptance unit; a sound source identification unit 361 that analyzes the audio signal received by the audio signal input unit 32 to identify the type of a sound source; a setting data acquisition unit 362 that acquires setting data according to the type of the sound source identified by the sound source identification unit 361; and a setting data setting unit 363 that makes the mixing device set the setting data acquired by the setting data acquisition unit 362 to the accepted channel.SELECTED DRAWING: Figure 3

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 automate broadcast transmission monitoring of a television station.SOLUTION: Screens of a broadcast operation table monitor and a video monitoring monitor of an automatic program transmission system are captured in a PC (Personal computer). A program of the PC recognizes monitor images to monitor broadcast transmission. The program of the PC automatically monitors a broadcast execution situation such as whether or not a content is transmitted according to a broadcast operation table and whether or not a content to be transmitted next is prepared, by a tally, which displays a control result of the automatic program transmission system, determining whether or not display is performed on the monitoring monitor according to the broadcast operation table and determining whether or not a material address displayed on the operation table monitor agrees with a material address displayed on the video monitoring monitor .SELECTED DRAWING: Figure 4

A collaborative cloud-based digital music composition, performance and production studio system network employing machine-intelligent digital audio workstation (DAW) systems that are supported by cloud-based automated music composition, performance, production and publishing services. Such services enable improved workflows and enhanced productivity, while ensuring that the music IP rights of all parties involved in the Al-assisted music creation process are respected and responsibly managed in the best economic interests of individual artists, performers, producers, publishers and consumers alike. The digital music studio system network can be realized in various ways supporting the DAW systems in ways suited to different user preferences and application environments.

The utility model discloses a multi-channel digital sound console, which comprises a device shell and a mobile protection heat dissipation mechanism, various components and circuit boards in the device shell are borne and protected through the device shell, universal wheels are positioned at the bottom and are enabled to have mobility through fixed connection, and a user can easily move equipment to a required position. The top end of the interlayer is provided with multi-channel plug-in ports, the multi-channel plug-in ports are connected through screws and used for connecting external audio equipment and a sound console for communication, the ports and other electronic equipment process audio input and output signals, high-quality audio mixing and processing capacity is provided for a user, the upper side and the lower side of the handle are provided with ventilation holes, and a filter screen and a cooling fan are installed in the handle. The ventilation holes are tightly combined with the filter screen, dust and sundries are prevented from entering the equipment, meanwhile, air circulation is ensured, the cooling fan is connected to the filter screen through screws, the equipment is kept within the normal working temperature range by actively discharging internal heat, the structure is simple, and movement is convenient.

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

Methods and apparatuses for automatic analysis and optimization of an audio signal are described herein. An example method may comprise receiving, by an audio signal optimizer, a first indication to perform an audio signal optimization, receiving an audio signal from an input device, recording a sample of the audio signal, analyzing the sample of the audio signal for at least one audio parameter, and performing, based on an analysis of the sample of the audio signal, the audio signal optimization of the audio signal, wherein the audio signal optimization comprises a configuration of a gain level of the audio signal.

The utility model relates to the technical field of broadcast television equipment, and discloses an adjusting device for a broadcast signal transmitter, which comprises a fixed box, a movable cavity is arranged in the fixed box, one side of the outer wall of the fixed box is fixedly connected with a mounting box, and the inner wall of the mounting box is fixedly connected with a motor I; the output end of the first motor is fixedly connected with a transmission shaft, the transmission shaft is rotationally connected to the inner wall of the movable cavity, the outer wall of the transmission shaft is fixedly connected with a worm, the inner wall of the fixed box is rotationally connected with a rotating shaft, the outer wall of the rotating shaft is fixedly connected with a worm gear, and the upper surface of the worm gear is fixedly connected with a rotating block. According to the utility model, a motor I is started to drive a worm and a worm gear to be engaged, so that a rotating block above a rotating shaft rotates, and a signal transmitting machine above a fixed plate rotates at multiple angles, and then an electric push rod is started to drive a hinge block, so that a supporting rod drives the signal transmitting machine to further adjust the angle; and the practicability of the device is improved.

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.