CN102437889A - Multi-path voice broadcast system based on field programmable gate array (FPGA) and implementation method of multi-path voice broadcast system - Google Patents

Multi-path voice broadcast system based on field programmable gate array (FPGA) and implementation method of multi-path voice broadcast system Download PDF

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Publication number
CN102437889A
CN102437889A CN2011104046477A CN201110404647A CN102437889A CN 102437889 A CN102437889 A CN 102437889A CN 2011104046477 A CN2011104046477 A CN 2011104046477A CN 201110404647 A CN201110404647 A CN 201110404647A CN 102437889 A CN102437889 A CN 102437889A
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voice
class
little
signal
broadcast system
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赵其珍
彭建朝
许向众
于书举
孙燕英
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Beijing University of Technology
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Beijing University of Technology
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Abstract

The invention discloses a multi-path voice broadcast system based on a field programmable gate array (FPGA) and an implementation method of the multi-path voice broadcast system, and belongs to the field of voice systems. A voice digital signal can be processed by programming the FPGA. The voice digital signal is transmitted into a data transmission control module by analog to digital conversion of voice coding and decoding chips for buffer storage and local coding; under the action of a synchronization and sending time control module, 4B/5B coding and non return zero inverse (NRZI) coding are performed; a driving receiving circuit module sends a differential signal for intra-network real-time broadcasting; each heating and talking point converts a path of selected broadcast signal into voice, outputs the voice into an earphone, and can be in voice conversation with another hearing and talking point until a talking control signal is valid; and the conversation between the hearing and talking points can be heard by other hearing and talking points which select the heating and talking point and do not build a voice channel with the heating and talking point. In the invention, on the basis of guaranteeing the anti-interference capacity in the transmission process, the engineering simplicity and configuration flexibility of the system can be effectively improved.

Description

Multi-path voice broadcast system and implementation method based on FPGA
Technical field:
The present invention relates to the solution of a kind of digitlization multi-path voice broadcast singal system, be applicable to that especially selection paid attention to the class in the voice in many speakers system.
Background technology:
At present; Known voice system is made up of the MCVF multichannel voice frequency circuit; Wherein the control through single-chip microcomputer realizes that the digitized processing process of each road voice signal is its main method; The all built-in mould of employed audio coding/decoding chip/number and D/A switch circuit need not otherwise designed change-over circuit again.Because the interface resource of single-chip microcomputer is limited, and hardware resource customizes, therefore can big limitations application flexibility and versatility when organizing the multi-path voice broadcast system.It is thus clear that restriction mainly be:
1. each road voice broadcast service uses independently circuit for transmitting digital signals, so that the line engineering of voice receiving terminal is complicated, has limited the voice way of broadcast system.
2. through caching process, again through the netting twine transmission, the realization of total is very complicated, and certain delay is arranged, and takies the running time of CPU, influences data-handling capacity under the control of CPU for audio digital signals;
3. select or arbitrate voice channel through an intermediate equipment (like gating matrix), limited and respectively paid attention to the class active selectivity a little in the voice system;
4. as independently application system existence, lack the interface that embeds other system, limited the function design mutual with other system.
Summary of the invention:
In order to overcome above-mentioned deficiency, the present invention provides a kind of new design philosophy, and this thought is based on the digitized voice signal of field programmable gate array (FPGA) technology and handles.On the basis of this speech processes mode antijamming capability in guaranteeing transmission course, can effectively improve the engineering terseness and the configuration flexibility of system.
A kind of multi-path voice broadcast system based on FPGA is characterized in that: be the network transport infrastructure that is made up of netting twine; Have several to pay attention to the class a little in this topological structure, each construction module of paying attention to the class a little is followed successively by audio coding decoding chip, Data Transmission Controlling module and drives receiving circuit module; Select one and pay attention to the class the synchronous main website of a conduct, timed sending synchronization frame all the time, and remaining is paid attention to the class and a little is equipped with several selector buttons and a speech control button; It is characterized in that: adopt network transport infrastructure that audio digital signals is transmitted, make each pay attention to the class and a little all have the right to select him to want the voice messaging of listening to.
The implementation method of described multi-path voice broadcast system based on FPGA is characterized in that: as paying attention to the class a little of synchronous main website, timed sending synchronization frame all the time; But pay attention to the class a little all initiation session for wherein any one; The voice signal of transmitting terminal realizes that through the audio coding decoding chip voice signal arrives the conversion of digital signal; The digital signal that to change again later is input in the Data Transmission Controlling module, carries out buffer-stored and local address coding, and only paying attention to the class of initiation session is distributed an absolute address; Drive in the receiving circuit module this encoding digital signals being formed Frame and be input to synchronously and under the effect of transmitting time control module, form difference and send signal and in local area network (LAN), send in real time via netting twine;
The point selection reception of paying attention to the class in the speech network is paid attention to the class voice messaging a little from other; The differential received signal of receiving terminal converts data reception signal to and sends into the Data Transmission Controlling module through driving receiving circuit module; This Frame decoded obtain local address and speech data; Judge according to local address whether speech data is effective; Effective speech data is carried out buffer-stored, deliver to the audio coding decoding chip via the Data Transmission Controlling module and realize that digital signal arrives the conversion of voice signal, hears sound through earphone at last.
The concrete principle of the present invention is following:
1. set up TCM Data Transmission Controlling module and D&R driving receiving circuit module based on the FPGA technology; Through respectively paying attention to the class connection a little; Structure only has the full duplex multi-path voice digital data transmission platform of two pairs of transmission lines, as: utilize two pairs of twisted-pair feeders in the network cable of a 100Mbps speed can not have the voice digital signal of transmission 20 road 48kHz frequencies of compression.Can increase the number of paying attention to the class a little, it does not receive the restriction of geocoding and line engineering.
2. but any in the Network Transmission pays attention to the class a little all initiation session.The voice signal of transmitting terminal realizes that through the audio coding decoding chip voice signal arrives the conversion of digital signal.To change digital signal later again is input in the Data Transmission Controlling module; Carry out buffer-stored and local address coding; Drive in the receiving circuit module this encoding digital signals being formed Frame and be input to synchronously and under the effect of transmitting time control module, form difference and send signal and in local area network (LAN), send in real time via netting twine.
3. a little all optionally connected receipts of paying attention to the class in the speech network are paid attention to the class voice messaging a little from other.The differential received signal of receiving terminal converts data reception signal to and sends into the Data Transmission Controlling module through driving receiving circuit module; This Frame decoded obtain local address and speech data; Judge according to local address whether speech data is effective; Speech data carries out buffer-stored effectively, delivers to the audio coding decoding chip via the Data Transmission Controlling module to realize that digital signal arrives the conversion of voice signal, hears sound through earphone at last.
4. in the speech network structure, be provided with a synchronous main website, regularly send the synchronous transmission frame, paying attention to the class a little on all speech networks all can conflict free order be sent the voice data transmission frame of oneself according to the setting of TCM Data Transmission Controlling module.
5. core concept of the present invention is that the application hardware descriptive language carries out hardware programming to fpga chip, has set up the voice transfer network system that a kind of time-division sends, sign indicating number divides reception.Promptly this system can confirm a certainly pay attention to the class a little whether can send speech data through node address, when sends the timesharing of speech data and sends data problem; Simultaneously can select to confirm speech data that this node requires reception is from which node through the coding of node address is different.
6. through fpga chip is carried out hardware programming, can design the interface that is connected with any system bus or external bus, especially be connected simply with the programmable system on chip (SOPC) that designs based on FPGA, can a shared fpga chip.
7. need not programming, do not have postpone the running time of CPU, fully by hardware controls.
The invention has the beneficial effects as follows: can improve the data parallel disposal ability and the speed of service, for the flexible design of interface, the flexibility of enhanced system and versatility have better realized mutual with other system.It is easy to control in addition, and simple in structure, power consumption is little.
Description of drawings:
Below in conjunction with accompanying drawing and embodiment the present invention is further specified
Fig. 1 is the system block diagram of patent of the present invention
Fig. 2 is selected audio coder & decoder (codec) block diagram
Fig. 3 is the digital signal processing module of patent of the present invention
Fig. 4 is corresponding to the local coder requirement among Fig. 3
Fig. 5 is based on coding speech data frame format later
Embodiment
The present invention is a kind of multi-path voice broadcast system based on FPGA, and it is the network transport infrastructure that is made up of netting twine.Netting twine mainly contains coaxial cable, twisted-pair feeder, and optical fiber, what this example was selected for use is unshielded twisted pair.With the ring topology is example, has several to pay attention to the class a little in this topological structure, and each construction module of paying attention to the class a little is followed successively by audio coding decoding chip, Data Transmission Controlling module and drives receiving circuit module.Select one and pay attention to the class the synchronous main website of a conduct, timed sending synchronization frame all the time, and remaining is paid attention to the class and a little is equipped with several selector buttons and a speech control button.As shown in Figure 1.
This system is based on the voice communication of N between paying attention to the class a little, serves as the transmission that synchronization frame does not stop by the background music frame, pays attention to the class a little until some and goes on the air, and confirms the voice messaging of certain broadcast node of reception through selector button.All can set up voice channel between paying attention to the class a little for any two; Press the speech control button; They just can talk with like this, and remaining pays attention to the class and a little still can select to continue to listen to the node voice information of setting up voice channel, but cannot talk with between them.
Fig. 2 is an audio coding decoding chip block diagram.Conversion between voice signal and the digital signal all is to realize through the A/D of chip internal and D/A change-over circuit.There is a lot of audio coding decoding chips can supply to choose in the market; The TLV320AIC23B of TI company for example, WM8731 of WOLFSON company or the like, its onboard clock generator; Can produce the sample frequency of 8kHz~96kHz, support the processing of several data figure place.Adopt standard sample frequency 48kHz in this example, and 24 data processing.Because the WM8731 phonetic codec chip is built-in with power amplifier, can directly drive earphone, for this instance adopts.
Fig. 3 is that the nucleus module that the present invention relates to is the transmission control module (TCM) of digital signal and drives receiving circuit (D&R)., the voice digital signal of low frequency to carry out buffer-stored and this node address coding after being input to TCM from DOI pin (being connected with the DOUT pin of audio coding decoding chip) to voice digital signal; Reaching under the effect of transmitting time control module synchronously, encode.Coded system has a lot; Can adopt graceful Chester coding or 4B/5B coding or the like; Owing to adopt the Network Transmission of 100bMps,, again Frame carried out nrzi encoding so select the 4B/5B coding; TXCD pin through higher rate sends the voice transfer frame signal, forms differential signal through D&R again and realizes long line balance transmission.At receiving terminal, at first convert differential signal to single-ended signal, be transferred to TCM and carry out successively judging whether to meet the local node geocoding requirement of selecting node after NRZI and the 4B/5B decoding, meet and then deposit buffer in, otherwise abandon.Export from DIO numeral pin (being connected) according to the frequency requirement of audio frequency then with the DIN pin of audio coding decoding chip.
The effect that reaches the transmitting time control module synchronously just is meant that after synchronization frame sends each pays attention to the class Frame orderly transmission of timing under the setting of this module a little.
The characteristics of 4B/5B coding be exactly can not occur in each byte surpassing 3 continuous 0; Become through 1 of nrzi encoding, after 0 invariant feature, in each byte of data of transmission over networks, 5 consecutive identical data can not occur at last.
Fig. 4 is the sketch map corresponding to local coder requirement among Fig. 3.Adopt 8 coded systems, highest order is that broadcasting is a little paid attention to the class in i of 0 expression, and highest order is that all the other any k of 1 expressions are paid attention to the class a little to this i response of paying attention to the class a broadcasting.(system of wherein i ∩ k=0, and k≤i) only is 0 address for i the broadcast allocation highest order of paying attention to the class a little, and paying attention to the class a little then shared these highest orders for their k of response is 1 address code.
The coding speech data frame format later that is based on shown in Figure 5.With local coder+L channel data+right data: for example: buffer memory (the left and right acoustic channels data can be 16,24 or 32 s' data processing, because the ADC of WM8731 chip internal and the processing mode that DAC is 24, so the processing of this example employing 24 bit data) is carried out in the 8+24+24=56 position; Form in good time transmission order according to the synchronizing signal of real-time network, under the control of sending order with the speech data of buffer memory in regular turn four one group carry out the 4B/5B coding and transmitted: mark (5, complete 1); Initial code (10, JK=1110010001), address code (10); (as: 60: 30+30) of numeric data codes; Spacing (5, complete 0), totally 90 fixed-length datas.Accomplish nrzi encoding when sending, form tranmitting data register data-signal TXCD.Wherein initial code 0x391 is that the transfer of data of 4B/5B coding standard begins.The effect of address code is after the recipient receives Frame, and address code is carried out NRZI and 4B/5B decoding successively, judges that according to this address code these frame data are the data that this recipient wants.If then receive this Frame.Speech data inputs to phonetic codec chip through the DIN port and converts voice signal to through earphone output (as shown in Figure 2) through DAC in this pays attention to the class a little.If not then this pays attention to the class to name a person for a particular job and does not receive this Frame.In the design of this Frame, not needing check code, is can not influence auditory effect because speech data is lost some data on a small quantity in transmission, can save CRC check sign indicating number and processing procedure thereof.Because this Frame is the fixed-length data frame, so need not to increase end code
The background music module is served as the function of synchronous main website, the transmission of synchronization frame be the position let all nodes have identical clock synchronously.At such Fast Ethernet transmitting data,, need 100 so transmit a Frame, i.e. 100*10ns=1us according to 90 intervals that add 10 of length of data frame format; And the sampling period is 1/48kHz ≈ 20.8333us, hence one can see that can hold at most 20.8333/1=20.8333 promptly 20 Frames simultaneously in transmission over networks.

Claims (3)

1. the multi-path voice broadcast system based on FPGA is characterized in that: be the network transport infrastructure that is made up of netting twine; Have several to pay attention to the class a little in this topological structure, each construction module of paying attention to the class a little is followed successively by audio coding decoding chip, Data Transmission Controlling module and drives receiving circuit module; Select one and pay attention to the class the synchronous main website of a conduct, and remaining is paid attention to the class and a little is equipped with several selector buttons and a speech control button.
2. the implementation method of the multi-path voice broadcast system based on FPGA according to claim 1 is characterized in that: as paying attention to the class a little of synchronous main website, timed sending synchronization frame all the time; But pay attention to the class a little all initiation session for wherein any one; The voice signal of transmitting terminal realizes that through the audio coding decoding chip voice signal arrives the conversion of digital signal; The digital signal that to change again later is input in the Data Transmission Controlling module, carries out buffer-stored and local address coding, and only paying attention to the class of initiation session is distributed an absolute address; Drive in the receiving circuit module this encoding digital signals being formed Frame and be input to synchronously and under the effect of transmitting time control module, form difference and send signal and in local area network (LAN), send in real time via netting twine;
The point selection reception of paying attention to the class in the speech network is paid attention to the class voice messaging a little from other; The differential received signal of receiving terminal converts data reception signal to and sends into the Data Transmission Controlling module through driving receiving circuit module; This Frame decoded obtain local address and speech data; Judge according to local address whether speech data is effective; Effective speech data is carried out buffer-stored, deliver to the audio coding decoding chip via the Data Transmission Controlling module and realize that digital signal arrives the conversion of voice signal, hears sound through earphone at last.
3. method according to claim 2 is characterized in that: adopt 4B/5B and nrzi encoding.
CN2011104046477A 2011-12-07 2011-12-07 Multi-path voice broadcast system based on field programmable gate array (FPGA) and implementation method of multi-path voice broadcast system Pending CN102437889A (en)

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Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN200956590Y (en) * 2006-05-26 2007-10-03 曾海波 Bus type addressing digital public broadcasting device
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CN101547268A (en) * 2009-04-24 2009-09-30 北京飞利信科技股份有限公司 Digital voice transmission system based on LAN
CN101695023A (en) * 2009-10-19 2010-04-14 南京莱斯信息技术股份有限公司 Parallel expanded type multi-path audio exchange mixing system
CN102185702A (en) * 2011-04-27 2011-09-14 华东师范大学 Intelligent conference system terminal controller, and operating method and application thereof

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN101278559A (en) * 2005-08-02 2008-10-01 布拉德利·L·戈特弗里德 Conference system
CN200956590Y (en) * 2006-05-26 2007-10-03 曾海波 Bus type addressing digital public broadcasting device
CN101547268A (en) * 2009-04-24 2009-09-30 北京飞利信科技股份有限公司 Digital voice transmission system based on LAN
CN101695023A (en) * 2009-10-19 2010-04-14 南京莱斯信息技术股份有限公司 Parallel expanded type multi-path audio exchange mixing system
CN102185702A (en) * 2011-04-27 2011-09-14 华东师范大学 Intelligent conference system terminal controller, and operating method and application thereof

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Application publication date: 20120502