CN202004920U - Network-digital integrated door phone system - Google Patents
Network-digital integrated door phone system Download PDFInfo
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- CN202004920U CN202004920U CN2011201387225U CN201120138722U CN202004920U CN 202004920 U CN202004920 U CN 202004920U CN 2011201387225 U CN2011201387225 U CN 2011201387225U CN 201120138722 U CN201120138722 U CN 201120138722U CN 202004920 U CN202004920 U CN 202004920U
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Abstract
The utility model relates to a network-digital integrated door phone system which is characterized in that: the network-digital integrated door phone system comprises an RF (radio frequency) anti-jamming module, a DSP (digital signal processor) processing module of an FM1182 (frequency modulation) chip, an anti-aliasing filter module, A/D (analog to digital) conversion module, a DSP processing module, a network interface module, a D/A (digital to analog) conversion module, a smoothing filtering module, a DSP processing module, a power amplifying module and a loudspeaker output module; the signal transmitting path of a system is as follows: after RF signals of an analog speech signal picked up by a microphone is filtered by a difference circuit, the analog speech signal and an analog speech signal played by a loudspeaker and picked up by the microphone are calculated and compared in DSP inside the FM1182 chip, and then are output after interference is filtered and echo is suppressed; the output audio-frequency signal is fed into the A/D to carry out (analog to digital conversion to form a digital speech signal after anti-aliasing filtering is carried out, and then the digital speech signal is fed into the DSP to carry our transformation processing to transmit the digital speech signal to a network interface, and finally, the digital speech signal is transmitted to the remote through the network. In the utility model, clear communication is provided, the cost is low and the performance is reliable.
Description
Technical field
The utility model relates to a kind of mechanics of communication, particularly discloses a kind of network digital one intercom system and transmission method thereof, is applied to the occasion that bank ATM needs two-way intercommunication from walk help, campus, subway station, prison etc.
Background technology
Intercom roughly has wireless and wired dual mode in the market, and this dual mode all to have significant disadvantages be that speech range is restricted.Also have the part talk-back host to adopt analog voice signal is imported the transmission of PC by PC network realization sound, this kind mode not only increases cost but also brings bigger noise jamming.
Summary of the invention
The purpose of this utility model is to address the deficiencies of the prior art, and adopts widely used at present DSP technology, discloses a kind of network digital one intercom system, and it is short to have remedied the prior art products speech range, and echo, shortcoming that noise is big.
The utility model is achieved in that a kind of network digital one intercom system, it is characterized in that comprising with lower module: RF is anti-interference, the DSP processing of FM1182 chip, anti-aliasing filter, A/D conversion, DSP processing, network interface, D/A conversion, smothing filtering, DSP processing, power amplification and loud speaker output; The signal transmission path of system is: the analog voice signal that is picked up by microphone is sent into the DSP of FM1182 chip internal after through difference channel filtering RF signal, thus with play the analog voice signal that is picked up by microphone again by loudspeaker and carry out computing comparison filtering interfering and suppress exporting after the echo at chip internal; Behind anti-aliasing filter, send into A/D through the audio signal after interference and the echo inhibition and carry out analog to digital conversion formation audio digital signals, audio digital signals is sent into DSP and is carried out after a series of conversion process audio digital signals being sent into network interface, finally is sent to long-range by network; Long-range to need the analog voice signal of transmission also be same, and smothing filtering exports DSP to after sending into DSP to carry out the D/A digital to analog conversion by network interface after the digitlization earlier, sends into power amplifier more after treatment and promotes loud speaker.
The transmission method of network digital one intercom system comprises the inhibition of the echo interference that overcomes in the communication process and the digitlization of analog voice signal;
A, the inhibition that echo is disturbed: the FM1182 chip that adopts U.S. Fu Di Science and Technology Ltd. to produce, utilize its SAM technology that adopts to have the ability that the pickup bundle forms, provide clear and do not have the communication of noise, the talker almost not influence of pickup bundle to being positioned at the pickup bundle, thereby eliminate the outer ambient noise of pickup bundle, the noise inhibiting ability of its enhancing, when eliminating steady-state noise and non-stationary noise, also support the acoustic echo of 60dB to eliminate, analog voice signal by the microphone input compares through sending into together in the DSP module after the A/D conversion and by the sound that the loudspeaker broadcast is picked up by the microphone input again, by speech processing algorithm echo is effectively suppressed and the voice output of will speaking normally;
The digitlization of b, analog voice signal: the analog voice signal to input at first is with limit filtering, sample then, quantize and encode, analog voice signal is transformed into digital bit stream, sends in the network by the audio digital signals of network interface after at last digitlization.
Described sampling is that analog voice signal is periodically scanned, analog voice signal continuous in time is become upward discrete analog voice signal of time, discrete analog voice signal through oversampling comprises all information of source analog voice signal, can recover former simulation voice signal, the lower limit of sampling rate should meet Nyquist sampling theorem--f undistortedly
s〉=2f
H, f wherein
sBe sampling frequency, f
HBe the analog voice signal highest frequency; Described quantification is that the amplitude of sample value is carried out discretization, promptly specifies Q level, and sample value is represented with immediate level, is called quantized value; Described coding is to represent quantized value (i.e. " 0 " and " 1 " two kinds of level values) with the binary system code character, quantizing in the real process is to finish simultaneously in cataloged procedure, the digital speech quantized signal of output is a kind of many level digitals voice signal, level numerical digit Q, generally get Q=256, this many level voice digital signal is as directly transmitting, interference free performance is very poor, therefore to Q level conversion be become corresponding binary digit voice signal through coding, represent Q=2 with k position binary system code character
k
The beneficial effects of the utility model are: take all factors into consideration cost and performance demands, take the solution of DSP (TMS320C5402)+FM1182.The utility model removes has cheap, the dependable performance of cost of manufacture, during standby outside the advantage such as low power operation, has also remedied shortcomings such as the echo that is had in the like product, noise be big.On the structure, take the theory of modularized design, it is promptly independent unified again to make digital processing part and echo disturb suppressing portion to be divided into two modules, can select module according to actual needs, two module segmentations can be used for other occasion again, not only increase flexibility, also increase practicality.
Description of drawings
Fig. 1 is that the utility model echo is suppressed structured flowchart.
Fig. 2 is the utility model analog voice signal digitlization block diagram.
Fig. 3 is the utility model network digital one intercom system block diagram.
Embodiment
The utility model system totally is divided into the two large divisions, i.e. simulation part and digital processing part on forming structure.The utility model has solved two big difficult point: a of prior art, the inhibition that echo is disturbed; The digitlization of b, analog voice signal.
With reference to the accompanying drawings 1, be effective containment echo, the utility model adopts the FM1182 chip of U.S. Fu Di Science and Technology Ltd., and the SAM technology that it adopts has the pickup bundle and forms ability, can provide clear and does not have the communication of noise.The talker almost not influence of pickup bundle to being positioned at the pickup bundle, and eliminate the outer ambient noise of pickup bundle; The noise inhibiting ability of its enhancing also supports the acoustic echo of 60dB to eliminate when eliminating steady-state noise and non-stationary noise.Analog voice signal by microphone input 1 input is sent into DSP module (similar Digital Signal Processing by microphone input 1 sound that picks up again afterwards and by the sound that loudspeaker are play together through the A/D conversion, principle is with following voice digitization) relatively middle, by speech processing algorithm echo is effectively suppressed and the voice output of will speaking normally.
The SAM technology: promptly small array microphone (Small Array Microphone) technology is a kind of voice processing technology with unique acoustic algorithms, and it can run on software, also may operate on this money chip.In this money chip, the SAM algorithm has been solidificated in the chip, have very high integrated level, without any need for peripheral components, and in this money pronounciation processing chip, adopted the technology that is similar to DSP, optimize at voice, have higher efficient.
With reference to the accompanying drawings 2, analog voice signal is carried out digitlization (DSP processing).Analog voice signal to input at first is with limit filtering, samples then, quantizes and encode, and analog voice signal is transformed into digital bit stream, sends in the network by the audio digital signals of network interface after with digitlization at last.
For realizing the digitlization of analog voice signal, select the TMS320C5402 chip of American TI Company, this chip is a low-power consumption, high performance fixed-point DSP chip, has the advantages such as CPU structure of fast operation, optimization.
Sampling: be that analog voice signal is periodically scanned, analog voice signal continuous in time is become upward discrete analog voice signal of time.The discrete analog voice signal of the sampling of process comprises all information of source analog voice signal, can recover former simulation voice signal undistortedly.The lower limit of sampling rate should meet Nyquist sampling theorem--f
s〉=2f
H(f
sBe sampling frequency, f
HBe the analog voice signal highest frequency).
Quantize: the amplitude of sample value is carried out discretization, promptly specify Q level, sample value is represented (calling quantized value) with immediate level.
Coding: represent quantized value with the binary system code character, (i.e. " 0 " and " 1 " two kinds of level values), quantizing in the real process is to finish simultaneously in cataloged procedure, and the digital speech quantized signal of output is a kind of many level digitals voice signal, level numerical digit Q (generally getting Q=256).This many level digitals voice signal is as directly transmitting, and interference free performance is very poor, therefore will Q level conversion be become corresponding binary digit voice signal through coding, promptly represents (Q=2 with k position binary system code character
k).
Network interface: for digitized audio digital signals is sent to network, select the 10M Ethernet chip RTL8109AS of Realtek Semiconductor company for use, this chip is supported the automatic detection mode of pnp, is supported ethernet ii and IEEE802.3 10Base5,10Base2,10BaseT, support UTP, AUI and BNC detect, support the multiple functions such as auto polarity correction of 10BaseT automatically.
With reference to the accompanying drawings 3, the utility model network digital one intercom system block diagram.
RF is anti-interference: adopt the difference channel design, thereby utilize the difference channel principle to suppress the RF radiofrequency signal that is produced by circuit and environment.
DSP handles: mainly to the analog voice signal processing that performs mathematical calculations, as do convolution, effect such as add up, multiply each other.
Anti-aliasing filter: converting analog voice signal to audio digital signals need sample to analog voice signal, and sampling will be satisfied Nyquist's theorem--f
s〉=2f
H(f
sBe sampling frequency, f
HBe the signal highest frequency), otherwise will produce aliased distortion, anti-aliasingly to eliminate this distortion exactly.
The A/D conversion: analog voice signal is converted to the process of audio digital signals, and process is referring to above-mentioned sampling, quantification, coding.
Network interface: for digitized signal is sent to network, select the 10M Ethernet chip RTL8109AS of Realtek Semiconductor company for use, this chip is supported the automatic procuratorial organ of pnp formula, is supported ethernet ii and IEEE802.3 10Base5,10Base2,10BaseT, support UTP, AUI and BNC detect, support the multiple functions such as auto polarity correction of 10BaseT automatically.
The D/A conversion: converting audio digital signals to analog voice signal, is the inverse process of A/D conversion, and process is referring to above-mentioned sampling, quantification, coding.
Smothing filtering: the time after the D/A variation is gone up discrete analog voice signal, after filtering, become analog voice signal continuous in time.
Power amplification: the continuous analog voice signal after The disposal of gentle filter, also be not enough to promote the loud speaker sounding because of signal is weak, power is less, promote loud speaker so will lower-powered analog voice signal be amplified through the one-level power amplification circuit.
Claims (1)
1. network digital one intercom system is characterized in that comprising with lower module: RF is anti-interference, the DSP processing of FM1182 chip, anti-aliasing filter, A/D conversion, DSP processing, network interface, D/A conversion, smothing filtering, DSP processing, power amplification and loud speaker output; The signal transmission path of system is: the analog voice signal that is picked up by microphone is sent into the DSP of FM1182 chip internal after through difference channel filtering RF signal, thus with play the analog voice signal that is picked up by microphone again by loudspeaker and carry out computing comparison filtering interfering and suppress exporting after the echo at chip internal; Behind anti-aliasing filter, send into A/D through the audio signal after interference and the echo inhibition and carry out analog to digital conversion formation audio digital signals, audio digital signals is sent into DSP and is carried out after a series of conversion process audio digital signals being sent into network interface, finally is sent to long-range by network; Long-range to need the analog voice signal of transmission also be same, and smothing filtering exports DSP to after sending into DSP to carry out the D/A digital to analog conversion by network interface after the digitlization earlier, sends into power amplifier more after treatment and promotes loud speaker.
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| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN2011201387225U CN202004920U (en) | 2011-05-05 | 2011-05-05 | Network-digital integrated door phone system |
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|---|---|---|---|
| CN2011201387225U CN202004920U (en) | 2011-05-05 | 2011-05-05 | Network-digital integrated door phone system |
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Cited By (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN103384352A (en) * | 2013-07-30 | 2013-11-06 | 深圳市汇川技术股份有限公司 | Elevator remote talkback system and access device |
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2011
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Cited By (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN103384352A (en) * | 2013-07-30 | 2013-11-06 | 深圳市汇川技术股份有限公司 | Elevator remote talkback system and access device |
| CN103384352B (en) * | 2013-07-30 | 2016-06-22 | 深圳市汇川技术股份有限公司 | Elevator remote talkback system and access device |
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Granted publication date: 20111005 Termination date: 20190505 |
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