CN102307271A - Network digital integrated talkback system and transmission method thereof - Google Patents
Network digital integrated talkback system and transmission method thereof Download PDFInfo
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- CN102307271A CN102307271A CN201110114531A CN201110114531A CN102307271A CN 102307271 A CN102307271 A CN 102307271A CN 201110114531 A CN201110114531 A CN 201110114531A CN 201110114531 A CN201110114531 A CN 201110114531A CN 102307271 A CN102307271 A CN 102307271A
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Abstract
The invention discloses a network digital integrated talkback system and a transmission method thereof and is characterized by overcoming restraint of echo interference during transmission and digitalizing analog voice signals. A working process of the network digital integrated talkback system is that: a radio frequency (RF) signal of an analog voice signal which is picked by a microphone is filtered out by a differential circuit, and the analog voice signal and an analog voice signal which is played by a horn and picked by the microphone are sent into a digital signal processing (DSP) unit inside an FM1182 chip, the two analog voice signals are subjected to comparison operation inside the chip and output after interference and restraint echoes are filtered out; the processed analog voice signals are subjected to anti-aliasing filtering and sent into an analog/digital (A/D) converter for A/D conversion to form digital voice signals, wherein the digital voice signals are sent into the DSP unit for conversion and then sent into a network interface so as to be transmitted to remote equipment through a network; the remote analog voice signals are digitalized and sent into the DSP unit through the network interface for D/A conversion so as to be smoothly filtered and output to the DSP unit; the remote analog voice signals are processed and sent into a power amplifier to push a loudspeaker. The network digital integrated talkback system provides clear communication, is low in cost and reliable in performance.
Description
Technical field
The present invention relates to a kind of mechanics of communication, particularly disclose a kind of network digital one intercom system and transmission method thereof, be 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 through PC network realization sound, this kind mode not only increases cost but also brings bigger noise jamming.
Summary of the invention
The objective of the invention is to address the deficiencies of the prior art, adopt widely used at present DSP technology, disclose a kind of network digital one intercom system and transmission method thereof, it is short to have remedied the prior art products speech range, and echo, shortcoming that noise is big.
The present invention is achieved in that a kind of transmission method of network digital one intercom system, it is characterized in that: overcome the inhibition of the echo interference in the communication process and the digitlization of analog voice signal;
The inhibition that a, echo are 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 through 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; Quantize in the real process is in cataloged procedure, to accomplish simultaneously; 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 pass through coding Q level conversion become corresponding binary digit voice signal, represent Q=2 with k position binary system code character
k
A kind of network digital one intercom system is characterized in that: comprise that 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 course of work of system is: the analog voice signal that picks up by microphone through difference channel filtering RF signal after; Play the DSP that the analog voice signal that is picked up by microphone is again sent into the FM1182 chip internal with loudspeaker, export after the echo with suppressing thereby carry out computing comparison filtering interfering 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 through network interface after the digitlization earlier, sends into power amplifier more after treatment and promotes loud speaker.
The invention has the beneficial effects as follows: take all factors into consideration cost and performance demands, take the solution of DSP (TMS320C5402)+FM1182.The present invention removes has low preparation cost, dependable performance, 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 interference suppress partly to become 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 echo of the present invention is suppressed structured flowchart.
Fig. 2 is an analog voice signal digitlization block diagram of the present invention.
Fig. 3 is a network digital one intercom system block diagram of the present invention.
Embodiment
System of the present invention totally is divided into the two large divisions on forming structure, i.e. analog part and digital processing part.The invention solves 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 present invention 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 loudspeaker sound played together through the A/D conversion; Principle is with following voice digitization) relatively middle, through 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 to 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 through 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 in cataloged procedure, to accomplish simultaneously, 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 pass through coding Q level conversion become corresponding binary digit voice signal, promptly representes (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, network digital one intercom system block diagram of the present invention.
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: convert analog voice signal the process of audio digital signals into, 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, lower-powered analog voice signal is amplified promote loud speaker so will pass through the one-level power amplification circuit.
Claims (3)
1. the transmission method of a network digital one intercom system is characterized in that: overcome the inhibition of the echo interference in the communication process and the digitlization of analog voice signal;
The inhibition that a, echo are disturbed: adopt the FM1182 chip; Utilize its SAM technology that adopts to have the ability that the pickup bundle forms; Communication clearly is provided; 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 sound played 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 through the audio digital signals of network interface after at last digitlization.
2. according to the transmission method of the described network digital one of claim 1 intercom system; It is characterized in that: 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, specifies Q level, and sample value is represented with immediate level, is called quantized value; Described coding is to represent quantized value with the binary system code character; With " 0 " and " 1 " expression level value; Quantize in the real process is in cataloged procedure, to accomplish simultaneously; The digital speech quantized signal of output is a kind of many level digitals voice signal, level numerical digit Q, generally gets Q=256; This many level digitals voice signal is as directly transmitting; Interference free performance is very poor, therefore will pass through coding Q level conversion become corresponding binary digit voice signal, representes Q=2 with k position binary system code character
k
3. network digital one intercom system that is generated by the described transmission method of claim 1 is characterized in that: comprise that 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 course of work 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 at chip internal and export after the echo with suppressing; 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 through network interface after the digitlization earlier, sends into power amplifier more after treatment and promotes loud speaker.
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Cited By (7)
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| CN104734740A (en) * | 2013-12-20 | 2015-06-24 | 国民技术股份有限公司 | Wireless audio data communication method and device |
| CN105338501A (en) * | 2014-08-08 | 2016-02-17 | 中兴通讯股份有限公司 | Information transmission method and device as well as information obtaining method and device during call and terminal |
| CN109005485A (en) * | 2018-07-28 | 2018-12-14 | 贵州航天天马机电科技有限公司 | A kind of voice power amplifier data processing circuit |
| CN111009243A (en) * | 2019-11-20 | 2020-04-14 | 厦门立林科技有限公司 | Voice recognition control method and system for building control system and storage medium |
| CN112637438A (en) * | 2020-12-17 | 2021-04-09 | 中科上声（苏州）电子有限公司 | Single-line transmission-based entrance guard double-end intercom method and system |
| CN114567706A (en) * | 2022-04-29 | 2022-05-31 | 易联科技(深圳)有限公司 | Public network talkback equipment jitter removal method and public network talkback system |
| CN116628481A (en) * | 2023-07-21 | 2023-08-22 | 江西红声技术有限公司 | Electronic countermeasure information source identification method, system, computer and readable storage medium |
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Cited By (8)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN104734740A (en) * | 2013-12-20 | 2015-06-24 | 国民技术股份有限公司 | Wireless audio data communication method and device |
| CN104734740B (en) * | 2013-12-20 | 2018-08-14 | 国民技术股份有限公司 | A kind of wireless audio data communications method and device |
| CN105338501A (en) * | 2014-08-08 | 2016-02-17 | 中兴通讯股份有限公司 | Information transmission method and device as well as information obtaining method and device during call and terminal |
| CN109005485A (en) * | 2018-07-28 | 2018-12-14 | 贵州航天天马机电科技有限公司 | A kind of voice power amplifier data processing circuit |
| CN111009243A (en) * | 2019-11-20 | 2020-04-14 | 厦门立林科技有限公司 | Voice recognition control method and system for building control system and storage medium |
| CN112637438A (en) * | 2020-12-17 | 2021-04-09 | 中科上声（苏州）电子有限公司 | Single-line transmission-based entrance guard double-end intercom method and system |
| CN114567706A (en) * | 2022-04-29 | 2022-05-31 | 易联科技(深圳)有限公司 | Public network talkback equipment jitter removal method and public network talkback system |
| CN116628481A (en) * | 2023-07-21 | 2023-08-22 | 江西红声技术有限公司 | Electronic countermeasure information source identification method, system, computer and readable storage medium |
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Application publication date: 20120104 |