CN105207788A - Multifunctional audio system based on broadband IP - Google Patents
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Abstract
The invention discloses a multifunctional audio system based on broadband IP, which relates to an intelligent audio playing system, wherein advanced communication and voice digital signal processing technologies are utilized to carry out modular design on voice processing and transmission, the existing network and cable are utilized, the current network switch is improved, the large-area professional audio management and control and flexible function configuration are realized, the system can be configured into a building broadcasting system, a KTV system, a professional cinema audio system, a conference system, a background music system and audio alarm systems in various linkage modes through software change, and the system has all characteristics of related systems and professional data processing capability.
Description
Technical field
The present invention relates to integrated circuit, circuit hardware design and Embedded Software Design, PC software and algorithm design, Communications And Computer integration field, utilize advanced communication and digital speech signal processing technology, speech processes and transmission are carried out modularized design, utilize existing network and cable, by transforming the current network switch, the network switch increases professional audio process module, on computers by software simulating professional audio in Intranet and the large area management and control of outer net and the flexible configuration of function, system can be configured to building broadcast system by software change, KTV system, specialty theater audio system, conference system, audio alarm system under background music system and various linkage pattern, the language data process mode of system under different functions and functional requirement different, the phonetic feature of dynamic-configuration different spaces can be required according to user, and pass through the different stereo spatial of dynamic-configuration loudspeaker structure.In professional audio process, use advanced algorithm and the great potential by current chip, adopt modularity mode in systems in which integrated time delay, equilibrium etc., space function is different, and corresponding process is different with the combination of algorithm.
Background technology
Current building broadcast and audio system adopt centralized power amplifier substantially, the mode of distributed mode speaker, professional audio cable is adopted to transmit between audio-frequency power amplifier and loud speaker, the cable type that different functions needs and number change are very large, can not share the algorithm of comprehensive wiring and Digital Signal Processing between various different function.
Current audio droadcasting system, the equipment such as general employing centralized configuration source of sound, matrix, amplifier, decentralized configuration loud speaker.Source of sound generally adopts multiple, as CD player, computer etc., by matrix, the audio frequency of needs is sent to the cable of specifying, and drives loud speaker to play.Current system as shown in Figure 1.
Annexation is as follows: CD player etc. is connected with matrix as source of sound, connect power amplifier after matrix, after amplifier, access loud speaker, because the signal amplitude of amplifier is general very high, reach 70 ~ 100VAC, therefore the access interface of each loud speaker needs the transformer of a matched impedance.
The course of work is as follows: the output of the sources of sound such as player can be simulation or digital signal, be generally analog signal, analog signal is through matrix, by the audio repeating of needs on corresponding amplifier, then be sent to corresponding professional audio cable, a cable generally connects 20 ~ 100 loud speakers, and the high pressure audio signal on cable is after transformer, be converted to the drive singal in loud speaker specification, make loud speaker sounding.
Although current building audio system achieves centralized control, incompatible with the communication technology of advanced person, make the cost of construction higher, and function singleness, provide the added value of extra-service almost not have.
Use professional power amplifier and gating matrix equipment.Costly, and function singleness, arranging and using needs professional to carry out to professional equipment, and general user cannot operate.
Use professional audio cable: cable, according to bearing capacity, mounts different loudspeaker loads and quantity; When load is more, use many voice-frequency cables; Voice-frequency cable is routed to loudspeaker from machine room; Omnidistance analog signal transmission, is easily disturbed; Signal transmission distance is far away, and power loss is larger; Every root cable is due to differences such as the physical location of manufacturing firm, laying and structures, and the decay of its frequency domain to audio signal is different, and the decay of this part frequency domain cannot compensate.
Loudspeaker need to use transformer to carry out load matched.The loud speaker of general manufacture has fixing power and impedance, when using high-voltage signal transmitting energy, must increase transformer carry out impedance matching at receiving terminal, and directly mounting needs additional customized loud speaker.
70 ~ 100V analog signal transmission: the security performance of direct-burried is lower, once cable cuticular breakdown, will cause the accidents such as electric leakage to occur, even cause electric shock accidents; Due to adopt analog signal transmission, the faults such as the short circuit of circuit cannot accurately detect, or testing cost too high and cannot economization batch adopt, cause system protection function imperfect.
Cable fault, then follow-up loudspeaker no longer work; Safety is guaranteed in the output must cutting off corresponding cable in maintenance, causes the service in overhauling all to be interrupted.
Incompatible with current IP network.Current mainstream technology is broadband IP and distributed power source, and the input due to substantial contribution makes the price of relevant device decline to a great extent, and is popularized.Audio system only has and IP technological incorporation, can be developed.
Summary of the invention
The present invention realizes by the following method.
Be illustrated in figure 2 IP-based multi-function audio system, annexation is: all loud speakers are directly connected with Node Controller, each Node Controller can connect 1 ~ 16 loud speaker (loudspeaker), Node Controller provides 24VDC to power by distributed power source, the other interface of Node Controller is connected with various transducer, as Smoke Sensor, infrared sensor etc.Node Controller is connected with data switching exchane by netting twine, and switch is connected with electronic lock, ONU, PC computer etc. by netting twine, ONU is connected with multiple switch by netting twine, and ONU can be connected with local side apparatus by optical cable or remote device connects.
According to service needed, system supports two kinds of mode of operations.
Local area network (LAN) interior decay.By any Node Controller of computer access, the audio signal of needs is sent to Node Controller according to the IP form of computer network, Node Controller is according to the requirement of user, corresponding voice data is processed, then be sent to the loud speaker of specifying, complete audio frequency process and playing function in the space of specifying.
Web mode of operation.User, through authorizing, can pass through computer access gateway, then enter internal network.Enter the associative operation after internal network, consistent with local area network (LAN) interior decay.
Advantage of the present invention is:
Polymera multiple target region orientation is play.Using PC or server as program source, the quantity of broadcasting channel is different from the data rate requirement of network speed and each program; In native system, the data communication default configuration of PC and server is 100MBPS/1GBPS adaptive interface, when the professional audio of 24 bits, 48K sampling is play, supports 10/100 broadcasting channel.Spatial distributions is divided into different regions by native system support, and the audio frequency of specifying is not received the impact of change in other regions in the space broadcast of specifying.Native system can divide broadcast area automatically according to sensing data, as in airport lounge, when passenger uses RFID boarding card, different spaces in boarding lounge can read the RFID data of passenger, as passenger, away from boarding gate, now boarding program starts, then system can judge according to the RFID data of up-to-date reading the region that passenger is detained, and dynamically multiple set of speakers of sub-region are synthesized interim broadcast area, audio plays prompt passengers check-ins to the boarding gate of specifying; After passenger's boarding, this dynamic combined automatically terminates combination.
Any audio format file is optional.Installation file format conversion module on PC, can be the form being suitable for system plays by the file transform of arbitrary format, be sent to the playback terminal of specifying.File broadcast format is acquiescence conversion, and when played file being dragged and dropped into appointment playback equipment, format conversion is carried out automatically.
Other audio-frequency playing terminal professional and digital signal processing capability.The professional audio broadcast request that Node Controller and drive circuit product support 24 bits, 48K/96K/192K to sample.The EQ of each sound channel default is configured to 10 sections, and each Node Controller can configure 4 sound channel signals; The frequency-adjustable of every section of EQ (PC of the heart is in the controlling arranged); 4 sound channel signal cascade configuration can be 1 road sound channel signal by each Node Controller, and now the EQ of each sound channel is configured to 40 sections, every section of EQ frequency-adjustable (PC of the heart is in the controlling arranged); The execution of all parameters realizes in distributed node controller.
Dynamic area is combined, and dynamic-configuration is stereo, supports the different stereo spatial audio configuration requirements required such as 2.1/5.1/7.1/10.2/22.2.An audio frequency of Node Controller is play module and is supported 2.1 stereo modes, and each Node Controller configures 4 audio frequency and plays module, and user can utilize 1 Node Controller to realize the stereo of 7.1 sound channels; When the stereo spatial of the higher levels such as laying 10.2 and 22.2, only need the loud speaker laid Node Controller as required and match with it in designated space.When building stereo environment, can for each channel configuration EQ, volume and time delay, to reach the object of optimizing stereophonic sound field.
Fusion seamless with other system, supports that fire-fighting and other warning system priority configure and linkage function.Broadcast system can realize linking with fire-fighting system, and the detection signals such as Smoke Sensor can direct access node controller, then by being wirelessly transmitted to the host computer of specifying; Be delivered to Node Controller by host computer by netting twine to the control command of water spray valve, the corresponding valve of Node Controller instruction operates; In the process, broadcast system can obtain the warning message of transducer, in appointment or according to preset program, in corresponding space broadcast warning message and relief information.Ru 10th floors breaking out of fires, broadcast system can according to preset program, first 9th floors, 11st floors warning message broadcast is triggered, prompting personnel withdraw from corresponding passage, then ordering protocol is to 12nd floors, 13rd floors etc., till until between whole designation area broadcast message being sent to this building in predetermined time section; Broadcast system can with security protection video system interlink, camera captured be detained article or article lose time, by broadcasting, information can be sent between designation area, or information is sent to neighbouring Security Personnel.
Based on the best price/performance ratio solution of broadband and ultra broadband IP framework.Channel quantity and broadcasting-quality are directly relevant to IP network.The transmission of voice between computer and Node Controller, all by IP network, completes the conversion of network to I2S and analog-driven in Node Controller inside, adopts Double-strand transmission between Node Controller and loud speaker.The audio frequency characteristics of twisted-pair feeder itself needs to be calibrated to professional audio pattern, calibrates and is realized by preset EQ parameter.Native system does not need the audio program equipment of specialty and the voice-frequency cable of specialty, and merging completely with current IP network, is the solution that cost performance is the highest.
Decentralized configuration, other sense and control technique interfaces of reserved wisdom building.Node Controller is the equipment of configuration of IP address, can be addressed after network is connected; Node Controller and loud speaker support the use, and can be laid in the space of needs; Node Controller and loud speaker adopt decentralized configuration, and the configuration of each point is carried out as required, selects corresponding distributed power source module simultaneously; Node Controller possesses multiple interfaces, as USB interface, RJ45 network interface, universal serial bus RS485 interface, CAN interface and other bus interface, various interface can directly with corresponding equipment connection, plug and play pattern is taked to access, as USB interface can plug USB flash disk, in USB flash disk, data can be read, and related content are write in USB flash disk.USB interface and RJ45 interface etc. can directly be connected with WiFi module, thus realize normal connection and the data transmission of system by means of broadband wireless communication network.
accompanying drawing explanation.
Fig. 1: current building broadcast system
Fig. 2: IP-based multi-function audio system block diagram
Fig. 3: Node Controller schematic diagram
Fig. 4 A:EQ(is balanced) modules in parallel combination schematic diagram
Fig. 4 B:EQ(is balanced) module tandem compound schematic diagram
Fig. 5: sound channel delay process
Fig. 6: the dynamic combined of different spaces function
Fig. 7: the signal processing flow of conference system
Fig. 8: the audio system configuration of digital camera.
Embodiment
Below in conjunction with drawings and Examples, content of the present invention is described in further detail.
Node Controller schematic diagram as shown in Figure 3, annexation is: TCP/IPMAC/PHY chip 4/8/16 port module external interface is RJ45 network interface, by equipment connections such as netting twine and computer, network microphone, servers, inside is connected with MCU by data wire, MCU connects digital audio process chip, digital audio processing chip exports 4 ~ 8 channel audio signal and bass PWM exports and bass modulating output, 4 ~ 8 channel audio signal are connected with Class-D audio driven respectively, and then each driving is connected with 1 ~ 3 loud speaker; Digital audio processing chip connects memory cell, carries out the process such as storage to the data accepted and send; MCU embeds interface with memory cell, RS485 interface chip, CAN interface chip, USB interface and transducer simultaneously and is connected; Microphone unit provides audio frequency to input for this locality, and power supply unit provides power supply for module.
Operation principle is: computer, server, voice data and control command are sent to the MAC/PHY interface of Node Controller by network microphone etc. by netting twine, after unpacking, instruction is sent to MCU, the voice data confirmed is sent to digital audio processing chip, digital audio processing chip is according to instruction scheduling memory cell and computing unit, time delay is carried out to voice data, audio mixing, the various process such as equilibrium, bass part by bass PWM output port or bass modulating output port to external equipment, other audio frequency output to Class-D audio driven module respectively according to 2.1 configurations, then loud speaker sounding is driven.
Node Controller according to instruction, can directly read corresponding voice data by USB port, or the voice data of direct reading cells inside is sent to digital audio processing chip.
Node Controller can detect the operating state of the loud speaker of each module inner and outside mounting in real time, after being packed, Monitoring Data is uploaded to computer process and display by MCU.
Node Controller has other interfaces, as transducer embeds interface, RS485 bus interface and CAN interface etc., can access different control commands and data, realizes uploading and assigning of data and instruction.As received the data of infrared sensor, data being reported to computer, according to the pattern that computer is pre-configured, detecting that people carrys out the rear corresponding audio frequency of OnNow and exports or judge that people leaves the corresponding audio frequency of rear closedown and exports; Or according to the data of Smoke Sensor, start voice alarm function etc.
EQ(is balanced as shown in Figure 4 A) modules in parallel combination schematic diagram, connection relation is: CH [1/2/3] module drives module to form by partition holding [1/2/3], multichannel EQ, 4 Class-D respectively.4 digital input port SDI [1/2/3/4] of each multichannel EQ module connect respectively at partition holding, 4 output interface SDO [1/2/3/4] drive module to be connected with 4 Class-D respectively, and each Class-D drives module to connect 1 ~ 3 loud speaker.Partition holding is connected with Leased line and address bus respectively.
Operation principle is: MCU, according to instruction, calculates the initial address of write data, then according to sequential, data write partition holding respectively; MCU instruction simultaneously multichannel EQ module, according to the preset initial address of each sound channel, obtains the data of each sound channel from partition holding.Multichannel EQ is after obtaining data by interface SDI [1/2/3/4], according to preset parameter, voice data is carried out to the equilibrium treatment of 0 ~ 10 section of Parameter adjustable, then data are sent to SDO [1/2/3/4], Class-D drives the data received through EQ process, is converted to power signal Direct driver loud speaker.
In paralleling model, each multichannel EQ module has 1 ~ 4 constrained input.The EQ of each sound channel can be divided into 10 sections, every section of Parameter adjustable.
EQ(is balanced as shown in Figure 4 A) module tandem compound schematic diagram, connection relation is: CH [1/2/3] module drives module to form by partition holding [1/2/3], multichannel EQ, 1 Class-D respectively.In 4 digital input ports of each multichannel EQ module, SDI1 is connected with partition holding, export SDO1 access input SDI2 and realize cascade, export SDO2 access input SDI3 and realize cascade, export SDO3 access input SDI4 and realize level, SDO4 and Class-D drives module to connect, and each Class-D drives module to connect 1 ~ 3 loud speaker.Partition holding is connected with Leased line and address bus respectively.
Operation principle is: MCU, according to instruction, calculates the initial address of write data, then according to sequential, data write partition holding respectively; MCU instruction simultaneously multichannel EQ module, according to the preset initial address of each sound channel, obtains the data of each sound channel from partition holding.Multichannel EQ is after obtaining data by interface SDI1, according to preset parameter, voice data is carried out to the equilibrium treatment of 0 ~ 10 section of Parameter adjustable, then SDI2 is accessed after data being sent to SDO1, according to preset parameter, voice data is carried out to the equilibrium treatment of 0 ~ 10 section of Parameter adjustable, then SDI3 is accessed after data being sent to SDO2, according to preset parameter, voice data is carried out to the equilibrium treatment of 0 ~ 10 section of Parameter adjustable, then SDI4 is accessed after data being sent to SDO3, according to preset parameter, voice data is carried out to the equilibrium treatment of 0 ~ 10 section of Parameter adjustable, then data are sent to SDO4, Class-D drives the data received through EQ process, be converted to power signal Direct driver loud speaker.
In series model, each multichannel EQ module has 1 constrained input.The EQ of each sound channel can be divided into 40 sections, every section of Parameter adjustable.
Sound channel delay process as shown in Figure 5, connection relation is: CH [1/2/3] process write data initial address, partition holding [1/2/3] is write after the address pointer of adjustment write, CH [1/2/3] read data initial address is all from the minimum of segmented system or maximum address, enter data format after digital independent and aim at module, be then sent to follow-up module by SDOx.
Operation principle is: the data reading rate of each sound channel is fixing, the read-write of data calculates according to 1Byte (8bits) at every turn, the voice data data reading rate in processes of sampling as 24 bit 48K is 24 bit * 48K=144KB/s, and the voice data data reading rate in processes that 24 bit 192K sample is 24 bit * 48K=576KB/s.
Be carry out by increasing or by subtracting order according to memory space during read data, with by increasing to example, every 3 read operations just can splice the 24 bit audio data completed.Suppose that the starting segment address of read data is 0, and the starting segment address writing data is X, then, after time delay [(X-0)/degrees of data speed], can enter real digital audio processing function.Different sound channels adopts different writing data initial address and ensure that identical read data divides segment base, just can obtain different delay process functions.
As under 24 bit 48K sampling configurations, sound channel 1 time delay 0.5s, sound channel 2 time delay 0.6s, then calculate according to above, write data initial address=0.5*144K=72K, write data initial address=0.6*144K=86.4K in the segmentation of sound channel 2 in the segmentation of sound channel 1.The delay parameter of such sound channel 1 and sound channel 2 can configure difference, and sound channel 2 is than sound channel 1 multiple-time delay 0.1s.
Adopt the time delay of this kind of method, have the error of smallest particles degree.Sound channel 1 delay parameter is 0.2458s, then the data initial address of writing obtained is 35395.2, and address must be integer, and therefore error is 6/1000000ths, i.e. 1.4us.This round-off error can be reduced by the out of phase read data synchronizing signal of clock signal fast, or multiple velocity repeats to read data and reduces error.
The dynamic combined of different spaces function as shown in Figure 6, multi-function audio system can the function in each space of dynamic combined, and as space 1 is configured to meeting room function, space 2 is configured to KTV function, and space 3 is configured to working region; After corresponding function completes, space function can be resetted, be set as other function as requested.
Space function is different, needs the module of the audio frequency process of configuration different.Corresponding configuration is realized by computer software.
Regardless of space function, can both monitor the running status of equipment, the feedback data of equipment.
The signal processing flow of conference system as shown in Figure 7.When system configuration is conference system, need to carry out independent configuration to data etc.In meeting room, spokesman needs configuration network paging microphone, and is carried out mating (convention needs to mate in advance) in the IP address of ICQ microphone and riding position, participant position by computer.
Owing to introducing microphone at meeting room, voice can be got back in microphone after loud speaker segmentation, and in certain space, specific frequency domain can be formed and utter long and high-pitched sounds.System adopts computer adaptive algorithm to suppress uttering long and high-pitched sounds.
Audio data transmission path in conference system: spokesman's ICQ microphone-Node Controller (accessing IP network by network interface)-switch (realizes exchanges data according to the IP address of setting, data upload)-computer (reception data, data are processed, according to the setting of conference system, the parameters such as the Node Controller address of interpolation broadcasting end and corresponding audio processing mode)-switch (data down transmission)-Node Controller (voice data that reception issues, carry out the process of specifying, the loud speaker of specifying will be sent to after data processing) (the generation of-loud speaker, allow the clear language hearing spokesman of participant).
The process of computer to voice is mainly two kinds of patterns: shift frequency and digital filtering.Shift frequency is exactly the plus-minus that the voice frequency domain of uttering long and high-pitched sounds can be caused to carry out frequency domain, and the 142Hz+ uttered long and high-pitched sounds as caused/-2.2Hz voice, transfer to 148Hz+/-2.2Hz, thus avoids the generation of uttering long and high-pitched sounds.Digital filtering to cause the 142Hz+/-2.2Hz voice of uttering long and high-pitched sounds directly remove from data or be reduced to very little degree, thus be unlikely to cause and utter long and high-pitched sounds.
On Node Controller, increase pick-up module exactly for the ancillary method of uttering long and high-pitched sounds, the data upload of pick-up is processed in real time to computer, thus monitor in meeting generation of whether uttering long and high-pitched sounds.By the process to trigger data, obtain causing the frequency domain components of uttering long and high-pitched sounds, in a computer real-time audio frequency or filtering process are carried out to respective component.
The audio system configuration of digital camera as shown in Figure 8, connection relation is: optical networking mouth switch is connected with server, ONU respectively by optical fiber, ONU is connected respectively at Node Controller by netting twine, external 1 ~ 8 loud speaker of each Node Controller, is driven by bass CLASS-D simultaneously and is connected with super low voice box.
Operation principle is: corresponding video is separated with audio frequency by the movie theatre database of server internal, and specify the play area of audio frequency (different Node Controller groups and corresponding loud speaker), data are sent to after optical fiber network interface cuts open switch, ONU is sent to according to the pattern of group broadcast according to the IP address of specifying, Node Controller is sent to after being converted to the signal of telecommunication, Node Controller carries out the process of time delay, equilibrium etc. to each sound channel according to order parameter, audio signal is sent to loud speaker sounding according to stereo mode, forms the audio system of stereophonic cinema.
Each Node Controller can complete the stereo dynamic-configuration of 7.1 sound channels, and 4 Node Controllers can complete the stereo sound configuration of 22.2.When stereo sound configuration precision improves, need to use multiple Node Controller.
Claims (11)
1. the present invention relates to a kind of multi-function audio system based on treating IP with leniency, utilize advanced communication and digital speech signal processing technology, speech processes and transmission are carried out modularized design, utilize existing network and cable, by transforming the current network switch, the network switch increases professional audio process module, pass through the flexible configuration of the large area management and control of software simulating professional audio and function on computers, system can be configured to building broadcast system by software change, KTV system, specialty theater audio system, conference system, audio alarm system under background music system and various linkage pattern, system possesses all features and the expert data disposal ability of related system.
2. system mainly comprises with lower module: computer and software, IP network, Node Controller, power supply, loud speaker.
3. the professional audio processing capacity according to claims 1, is characterized by: the professional audio process comprising various function, as time delay, equilibrium, audio mixing, chauvent's criterion etc.; Different functions is integrated in Node Controller and computer software according to modularity respectively; System is when installation space function, and corresponding audio frequency processing capacity module can be followed automatically.
4. the large area management and control function constructed according to claims 1, is characterized by: can mount loud speaker as required in system, number of loudspeakers is not limit; Each loud speaker can play different audio frequency at the same time, or as requested multiple set of speakers is woven to a specific function region and plays; After multiple loud speaker forms a play area, the broadcasting that can set this region is stereo or speakerphone mode; Support multiple stereo mode: 2.1 patterns, 5.1 patterns, 7.1 patterns, 22.2 patterns etc.; The operating state of each loud speaker can be detected and record; System, by extra high-power audio amplifier, supports the powerful loud speaker of extra mounting.
5. the flexible configuration function constructed according to claims 1, is characterized by: except the space function of default configuration, and other function that user needs can configure on computers, or is realized by the parameter of the correlation function module of setting Node Controller; Functional module group can carry out switch, bypass and setting parameter; Other function that oneself can like by user is run in systems in which according to plug-in unit pattern.
6. the IP network according to claims 1, is characterized by: identical real-time Transmission and the process carrying out voice based on IPV6; In the region having IP network, as long as bandwidth is enough, do not need to carry out secondary cable wiring manufacture; Lower-powered loud speaker can be driven by netting twine and not need professional audio cable, and the frequency domain characteristic of netting twine is compensated by EQ; System can according to different space functions, the IP network parameter that setting is different automatically; System supports Inside and outside network operation and transfer of data.
7. the Node Controller according to claims 1, is characterized by: Node Controller adds professional audio process and audio frequency Class-D power amplifier module on common data switching exchane basis; Node Controller has the multiple interfaces such as RS485 bus, CAN, USB simultaneously; The normal data translation-protocol of often kind of interface is built-in, or carries out long-range agreement change according to the content of sensor manufacturer; Node Controller possesses unique IP address.
8. the power supply module according to claims 1, is characterized by: the power of power supply module is appraised and decided according to the gross output of the loud speaker of mounting; Power supply module adopts distributed placement in physical layout, and as far as possible near the larger loud speaker of power or set of speakers.
9. the audio alarm system under the linkage pattern according to claims 1, is characterized by: the transducer that linkage pattern needs can be directly connected by bus with Node Controller, also directly can equal computer by netting twine and connect; Computer automatically can identify the classification of warning and find the position of warning according to generalized information system, automatically triggers relevant audio alarm; Audio alarm and corresponding paging system, can support that input text goes out acoustic pattern; Audio database can be multilingual, and supports successively to play according to the languages order of setting.
10. the delay function in the professional audio process according to claims 2, is characterized by: system constructs different time delays by not coexisting of the read-write initial address of voice data in memory block; The initial address of pronunciation audio data is identical, is singly that writing address is different according to time delay difference; Writing address and time delay are linear relationships, and linear dimensions is the cycle of read data, namely write data initial address=read data initial address+time delay/read data cycle; Data writing operation can multiple sound channel be carried out simultaneously, also can carry out fast one by one.
Equalization function in 11. professional audio process according to claims 2, is characterized by: Operation system setting 3 class is balanced: diagram equilibrium, equalization parameter, phase equalization; System has 4 dual tracks, arrange 10 equalizers, and each parametric equalizer is adjustable in each sound channel; The input and output of 4 sound channels can cascade, and namely go up the input of output as next sound channel of a passage, therefore system only can retain 1 dual track, but the equilibrium of each sound channel has 40, and the Parameter adjustable of each equilibrium.
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| CN (1) | CN105207788A (en) |
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| CN107257335A (en) * | 2017-06-09 | 2017-10-17 | 上海天跃科技股份有限公司 | Audio information transmissions system |
| CN116153321A (en) * | 2023-04-19 | 2023-05-23 | 中国传媒大学 | Sound processing computing server and sound processing computing method suitable for audio system |
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