CN103441821B - A kind of digital subaudio frequency weak signal processing method - Google Patents

A kind of digital subaudio frequency weak signal processing method Download PDF

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CN103441821B
CN103441821B CN201310394336.6A CN201310394336A CN103441821B CN 103441821 B CN103441821 B CN 103441821B CN 201310394336 A CN201310394336 A CN 201310394336A CN 103441821 B CN103441821 B CN 103441821B
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郭长勇
张键
李辉辉
张财元
时勇
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Tianjin 712 Communication and Broadcasting Co Ltd
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Abstract

The present invention relates to digital mobile radio communication technology, particularly to a kind of digital subaudio frequency weak signal processing method.First the method carries out over-sampling to numeral sub-audio signal, adds up each sampled point obtained, if cumulative sampled point to symbol lengths, completes bit synchronization by bit-synchronization algorithm;Use aggregation algorithms that the level value of each symbol is classified subsequently, and from set, calculate positive and negative level standard value, calculate decision threshold;At the optimum sampling point of each symbol of digital subaudio frequency signal, it is sampled judgement, and adds average algorithm in the calculating of each symbol sampling judgement, obtain digital subaudio frequency code element information, i.e. complete the extraction of digital subaudio frequency code.The present invention compares traditional method, is not affected by hardware sampling rate error, and under the communication environment of low signal-to-noise ratio, can obtain accurate optimum sampling point and decision level, and then effectively reduce the bit error rate that digital subaudio frequency weak signal receives.Applied widely, there is promotional value.

Description

A kind of digital subaudio frequency weak signal processing method
Technical field
The present invention relates to digital mobile radio communication technology, particularly relate to a kind of digital subaudio frequency weak signal process side Method.
Background technology
Digital mobile radio communication system (locomotive station, transmitter receiver) is audio signal modulation when, often less than sound The frequency range of frequency part adds a sub-audio signal, at receiving terminal by determining whether that sub-audio signal decides whether to open Noise elimination.Add subaudio frequency and can improve the capacity of resisting disturbance of digital mobile radio communication system, it is achieved the choosings such as different groups exhales, generally call Calling function.
Subaudio frequency is divided into simulation subaudio frequency CTCSS and digital subaudio frequency DCS, CTCSS and DCS broadly falls into international signaling Standard, wherein CTCSS has 38 standard code, and DCS code character is generally 83 or 104.Optional code character quantity due to DCS Flexible more than CTCSS, DCS signal digital processing, and Golay coding has error correction, so DCS has the most excellent compared with CTCSS Gesture.
The standard baud rate of digital subaudio frequency DCS is 134.3Hz, and frequency is relatively low, and connects 1 and even 0 owing to having in DCS code, The waveform frequency of DCS code is relatively low, about 25Hz~60Hz.For the signal that frequency is the lowest, if comprising direct current in signal, then this is straight Stream because the electric capacity of series winding is while filtering direct current, can not can cause DCS to believe by filtering at hardware circuit series capacitance Number distortion is serious, so this direct current signal needs to filter in digital signal processing.
Owing to digital subaudio frequency DCS having been only required baud rate, and sub-audio signal frequency modulation is not limited, So the demodulation output subaudio frequency amplitude of receiver is not fixed, and owing to the inconsistent of hardware circuit may cause signal to be modulated The change of frequency deviation, then the signal amplitude of receiving terminal demodulation output is it may happen that change, so digital subaudio frequency DCS of demodulation output Wave-shape amplitude is the most uncertain, and owing to direct current signal cannot filter by hardware, needs the most accurately to calculate DCS code Decision threshold.
In the processing procedure of digital signal, bit synchronization is one of key of process.Owing to the coding of digital subaudio frequency DCS is advised Synchronizing sequence is not reserved in then, so receiving DCS signal when, it is impossible to reach position by processing synchronizing sequence Synchronize.The bit synchronization of traditional DCS signal acceptance method is to be realized by the synchronized sampling clock that adjustment is local, and adjustment is adopted The sample rate that sample clock realizes receiving terminal to 134.3Hz is consistent with DCS code sign rate.Traditional DCS method of reseptance exists such as Lower shortcoming:
1) due to hardware differences, the sample rate of 134.3Hz there may be error, and sampling will cause error to be tired out for a long time Long-pending.
2) synchronized sampling cannot ensure the optimum sampling point sampling at each symbol, the most affected by noise when weak signal.
3), during weak signal sampling, sampled point noise may cause code value mistake in judgment.
4) DC component of signal may cause code value mistake in judgment.
For reception digital subaudio frequency DCS code that can be correct when weak signal, need a kind of signal processing method of design.
Summary of the invention
The problem existed in view of above-mentioned prior art, it is an object of the invention to design a kind of digital subaudio frequency weak signal and processes Method.With it, achieve DCS equalisation of over-sampled signals, door algorithm the latest is then utilized to realize bit synchronization, followed by poly- The level value of each symbol is classified by set algorithm, and calculates positive and negative level standard value from set, calculates decision threshold, and often The calculating of individual symbol sampling judgement adds average algorithm, is finally completed the process that DCS code receives.
The present invention adopts the technical scheme that: a kind of digital subaudio frequency weak signal processing method, it is characterised in that the method First numeral sub-audio signal is carried out over-sampling, each sampled point obtained is added up, if cumulative sampled point To a symbol lengths, complete bit synchronization by bit-synchronization algorithm;Use aggregation algorithms that the level value of each symbol is divided subsequently Class, and from set, calculate positive and negative level standard value, calculate decision threshold;Each symbol optimal at digital subaudio frequency signal Sample point is sampled judgement, and adds average algorithm in the calculating of each symbol sampling judgement, obtains digital subaudio frequency Code element information, i.e. completes the extraction of digital subaudio frequency code.
The invention have the advantages that: the present invention compares traditional method, not by the shadow of hardware sampling rate error Ring, and under the communication environment of low signal-to-noise ratio, accurate optimum sampling point and decision level can be obtained, and then it is sub-effectively to reduce numeral The bit error rate of audio frequency weak signal reception.Applied widely, there is promotional value.
Accompanying drawing explanation
Fig. 1 is overall flow figure of the present invention;
Fig. 2 is that bit-synchronization algorithm completes bit synchronization flow chart;
Fig. 3 is that aggregation algorithms calculates decision threshold flow chart;
Fig. 4 is that sampling judgement obtains digital subaudio frequency code element information flow chart;
Fig. 5 is that the present invention is applied to locomotive station system block diagram;
Fig. 6 is locomotive station subaudio frequency partial schematic diagram.
Detailed description of the invention
In order to be more clearly understood that the present invention, describe in detail in conjunction with the accompanying drawings and embodiments: with reference to Fig. 1, first this method is selected Select and numeral sub-audio signal is carried out over-sampling under the sampling rate of 1.2KHz, each sampled point obtained is tired out Add, if cumulative sampled point to symbol lengths, complete bit synchronization by bit-synchronization algorithm;Use aggregation algorithms pair subsequently The level value classification of each symbol, and from set, calculate positive and negative level standard value, calculate decision threshold;Believe in digital subaudio frequency Number each symbol optimum sampling point at be sampled judgement, and in the calculating of each symbol sampling judgement, add average calculation Method, obtains digital subaudio frequency code element information, i.e. completes the extraction of digital subaudio frequency code.
This method utilizes over-sampling to obtain the Sample buffer of DCS waveform, then averages waveform, and rough estimate is current The DC component of DCS waveform.Then utilize early late door algorithm to complete bit synchronous operation, calculate current sampling point corresponding morning door and The integration of door late, relative error adjusts sampling point position, calculates the signal energy of whole symbol lengths corresponding to current sampling point Value, and calculate error threshold with this.When door error is less than this error threshold the latest, it is believed that present bit synchronously completes.
After bit synchronization completes, obtain the optimum sampling point of each symbol, find the maximum in all symbols and minima, And on the basis of once, set up positive and negative two set, the optimum sampling point of each symbol is classified, if this level point value and just The distance of reference value is less than thresholding, then put into and just gather, if the distance in like manner with negative reference value is less than thresholding, then puts into negative collection Close.After traveling through whole DCS signal 23 bit, from the central point just gathering and bearing set of computations respectively set, as just The standard value of negative level, and calculate direct current with this, this direct current is set to decision threshold.
Owing to the DC component before bit synchronization is by averaging calculated to whole DCS waveform, its direct current Estimation is inaccurate, and has a larger fluctuation, and the direct current that aggregation algorithms calculates is the most accurate, but the premise of aggregation algorithms be position with Step completes.So before synchronously completing in place, bit-synchronization algorithm uses the direct current of algorithm rough estimate of averaging, and works as bit synchronization After completing, this direct current is replaced with the D. C. value that aggregation algorithms calculates by bit-synchronization algorithm, after D. C. value can make the most accurately Continuous bit-synchronization algorithm is the most accurate.
After completing bit-synchronization algorithm and aggregation algorithms calculating decision threshold, algorithm enters at the optimum sampling point of each symbol Line sampling is adjudicated, in order to reduce the noise interference to sampled point under weak signal, if about the average optimum sampling point of algorithms selection The level value done, compares with this value and decision threshold, finally gives DCS code.
With reference to Fig. 2, bit-synchronization algorithm completes bit synchronization and has the following steps:
Step 201: numeral sub-audio signal is carried out over-sampling under the sampling rate of 1.2KHz.Digital subaudio frequency signal Symbol rate be 134.3Hz, and owing to needs carry out the operation such as bit synchronization, need dynamic adjustment, so needing signal mistake Sampling, selects in this method to sample numeral sub-audio signal under the sampling rate of 1.2KHz, and the most each symbol is 9 Individual sampled point.
Step 202: if previous symbol period completes bit synchronization, then use previous symbol period aggregation algorithms accurate The DC component estimated.Owing to the core of bit-synchronization algorithm is to use door synchronous method the latest, and the method judges current adopting Sampling point needs reach or rear shifting is to compare early door integration and the relation of slow door integration, and this relation is the quickest to DC component Sense, so needing to remove the DC component of signal.Often collecting 9 sampled points, reach the length of a symbol period, bit synchronization is calculated Method all can perform once to have attempted bit synchronization, if previous symbol period completes bit synchronization, then algorithm is at bit synchronous base Obtain the optimum sampling point of symbol on plinth, and utilize aggregation algorithms to obtain level value accurately, and calculate accurately with this DC component.Then this step uses by the calculated DC component accurately of aggregation algorithms.
Step 203: if previous symbol period does not complete bit synchronization, then whole digital subaudio frequency signal is calculated average Value, obtains rough estimate DC component.Do not complete bit synchronous in the case of, current symbol sampling determination point be not On the optimum sampling point of symbol, so the DC component that current aggregation algorithms calculates is without reference to value, this step is to whole DCS signal calculates meansigma methods and obtains rough estimate DC component.Computing formula is as follows:
In formula, CT is rough estimate DC component, dnFor the n-th sampled point of DCS signal, N is adopting of whole DCS signal Sampling point number, under the sampling rate of 1.2KHz, N=207(mono-group of DCS signal totally 23 symbols, 9 sampled points of each symbol).
Step 204: if synchronization control number of times is without departing from threshold value, then on the basis of current sampled point, sets early door Scope and slow door scope, in the range of all sampled points remove DC component, and calculate early door range sample point and late respectively The integrated value of door range sample point.Early the length of door and slow door is chosen as being slightly larger thanSymbol lengths, for this sampling speed Rate, the length of door is chosen as 5 sampled points.The door too short meeting of length causes bit synchronization estimation difference relatively big, and the long meeting of door length Operand is caused to strengthen.Computing formula is as follows:
In formula, SE is the integrated value of early door, SLFor the integrated value of slow door, L is the length of door, this step L=5, dnBelieve for DCS Number the n-th sampled point.
Step 205: calculate the energy value of current sign length waveform, calculates error threshold according to this energy value.Due to defeated The amplitude entering signal is uncertain, so the error threshold of door is also uncertain the latest, this step selects by calculating one The energy of individual symbol lengths waveform calculates error threshold.Computing formula is as follows:
E in formulaSBeing the energy of a symbol lengths waveform, S is the sampled point number of a symbol, this step S=9, dnFor N-th sampled point of DCS signal.
The error threshold of door is chosen as energy the latest.Computing formula is as follows:
T in formulaEError threshold for door the latest.
Step 206: calculate the error of the early integrated value of door range sample point and slow door range sample point.Calculate door the latest to calculate The error of the integrated value of two doors, i.e. S in methodE-SL, the waveform removing DC component is performed door algorithm the latest when, it is Can be by judging that the symbol of the error of two doors determines the offset direction of sampled point.
Step 207: if the error of the integrated value of door is just and to be more than error threshold the latest, then it represents that sampled point is advanced, This sampling location is moved to left a point, then performs subsequent cycle bit-synchronization algorithm, until exceed synchronization control frequency threshold value or Person completes bit synchronization.
Step 208: if the error of the integrated value of door is more than error threshold for negative and absolute value the latest, then it represents that sampled point is stagnant Rear, this sampling location being moved to right a point, then performing subsequent cycle bit-synchronization algorithm, until exceeding synchronization control number of times threshold Value or complete bit synchronization.
Step 209: if the absolute value of the current error of the integrated value of door the latest is less than error threshold, then it is assumed that complete position same Step, current sampled point is i.e. it is believed that the optimum sampling point of symbol.
With reference to Fig. 3, aggregation algorithms calculates decision threshold and has the following steps:
Step 301: search level maximum and the minima of the digital subaudio frequency signal received.I.e. search DCS signal The level maximum L of 23 symbolsmaxWith minima Lmin
Step 302: with level maximum LmaxOn the basis of set up and just gather, with level minima LminOn the basis of set up negative collection Close, and by level maximum LmaxWith minima LminDistance 1/2 be set to distance decision threshold, define level maximum LmaxFor positive benchmark, level value minima LminFor negative benchmark.
Step 303: if all symbols (23 symbols) search does not completes, then calculate current sign level and positive benchmark away from From.Computing formula is as follows:
In formulaFor nth symbol level SnWith positive benchmark LmaxDistance.
Step 304: if the distance of current sign level and positive benchmark is less than the distance decision threshold arranged, then this is accorded with Number it is included into and just to gather.
Step 305: if the distance of current sign level and positive benchmark is more than the distance decision threshold arranged, then calculate current Symbol level and the distance of negative benchmark.Computing formula is as follows:
In formulaFor nth symbol level SnWith negative benchmark LminDistance.
Step 306: if the distance of current sign level and negative benchmark is less than the distance decision threshold arranged, then this is accorded with Number it is included into negative set.
Step 307: return step 303, calculating following symbol to positive benchmark and the distance of negative benchmark, and put into the collection of correspondence Close.I.e. travel through 23 symbols of DCS signal.
Step 308: if all symbols (23 symbols) have been searched for, then calculate in just gathering each symbol to other symbol Distance close.Computing formula is as follows:
In formulaFor positive set symbol level SmTo just gathering other symbol level SnDistance close, M is positive set symbol Number.
Step 309: find and just gather the symbol S that middle-range clutch is minimumm, by the level S of this symbolmIt is defined as positive level mark Quasi-value LP
Step 310: calculate the distance of each symbol and other symbol in negative set and close.Computing formula is as follows:
In formulaFor negative set symbol level SmTo negative set other symbol SnDistance close, M be negative set symbol Number.
Step 311: find the symbol S that negative set middle-range clutch is minimumm, by the level S of this symbolmIt is defined as negative level mark Quasi-value LN
Step 312: calculate DC component by positive level standard value and negative level standard value, this DC component is set to The decision threshold of DCS sampling judgement.Computing formula is as follows:
C in formulaRFor DC component exact value, this value is set to the decision threshold of DCS sampling judgement.
With reference to Fig. 4, sampling judgement obtains digital subaudio frequency code element information and has the following steps:
Step 401: calculate the optimum sampling point position of current sign.Under 1.2KHz sampling rate, each symbol has 9 to adopt Sampling point, so symbol optimum sampling point position offsets 9 points backward before the optimum sampling point position of each symbol is.
Step 402: calculate the average of point near optimum sampling point.Owing under weak signal, DCS Signal-to-Noise is less, noise Disturb the optimum sampling point position level fluctuation by causing each symbol relatively big, and then likely mistake in judgment, so this step Select point near optimum sampling point is taken average, thus reduce the noise fluctuations impact on judgement.Computing formula is as follows:
A in formulamFor the symbol average of corresponding optimum sampling point position m, SnFor DCS signal the n-th sampled point, E is for taking average Scope, this step selects comprise the most each two points and average, so E=2.
Step 403: if the average of point is more than the decision threshold of digital subaudio frequency sampling judgement near optimum sampling point, then Judgement current sign is 1.
Step 404: if the average of point is less than the decision threshold of digital subaudio frequency sampling judgement near optimum sampling point, then Judgement current sign is 0.
By each symbol of DCS signal of input is sampled judgement, obtains DCS digital signal, digital signal is entered Row decoding operation, is finally completed DCS code and receives process.
It is that the present invention is applied to locomotive station system block diagram with reference to Fig. 5, Fig. 5.Radio band signal through antenna receive after, Down coversion and demodulation operation is completed by channel device.Waveform after demodulation is divided into audio-frequency unit and subsonics by baseband filtering circuit Frequently part, then supplied with digital signal processes the digital processing of the complete pair signals of circuit, is converted into the operations such as voice.Signal subsequently Be converted to simulated sound by voicefrequency circuit, be then converted to, by loudspeaker, the sound that human ear is heard.
With reference to Fig. 6, the particular hardware of the DCS signal processing giving locomotive station in figure is constituted.Network in figure Label sub-tone is sub-audio signal input, and the signal of channel device demodulation signal after subaudio frequency low pass filter exists Input at this network label.Electric capacity C129 is capacitance, filters the DC component of sub-audio signal.Device N15A is that computing is put Big device chip 2904, this chip and resistance R178, resistance R177 form reverse amplification circuit, for putting sub-audio signal Greatly, electric capacity C130 is used for the noise section that rejection frequency is higher, and the power supply of 5V is defeated by 8 feet of operational amplifier chip 2904 Entering, electric capacity C135 and electric capacity C132 is for filtering the noise that 5V powers.The voltage of 5V by resistance R176 and resistance R175 dividing potential drop, and Inputted, as the reference voltage of operational amplifier chip 2904 by 3 feet of operational amplifier chip 2904.Electric capacity C131 is used for Filter the DC component of sign-changing amplifier output.3.3V is powered decile by resistance R186 and R185, at sub-audio signal One positive fixed-bias transistor circuit of superposition, is converted to positive voltage by whole sub-audio signal, and convenient ADC below gathers.Electric capacity C140 is for filtering the high fdrequency component in sub-audio signal.
Digital signal processing circuit is chosen as the mixture control 56F8322 that Freescale company produces, and this chip has The operational capability of DSP and the function of microcontroller, this chip has 6 ADC input pins, and this programme selects to use 23 feet ADC input pin ANA4 is as input, and the sub-audio signal that input demodulation obtains, chip 56F8322 passes through on-chip timer Gather DCS signal with the speed of 1.2KHz, in sheet, then complete the computing of algorithm, be finally completed the reception of DCS code.
According to the above description, the solution of the present invention can be realized in conjunction with art technology.

Claims (4)

1. a digital subaudio frequency weak signal processing method, it is characterised in that first numeral sub-audio signal is carried out by the method Over-sampling, adds up to each sampled point obtained, if cumulative sampled point to symbol lengths, passes through bit synchronization Algorithm completes bit synchronization;Use aggregation algorithms that the level value of each symbol is classified subsequently, and calculate positive and negative level from set Standard value, calculates decision threshold;It is sampled adjudicating at the optimum sampling point of each symbol of digital subaudio frequency signal, and The calculating of each symbol sampling judgement adds average algorithm, obtains digital subaudio frequency code element information, i.e. complete digital subaudio frequency The extraction of code.
A kind of digital subaudio frequency weak signal processing method the most according to claim 1, it is characterised in that bit-synchronization algorithm is complete Bit synchronization is become to have the following steps:
Step 201: numeral sub-audio signal is carried out over-sampling;
Step 202: if previous symbol period completes bit synchronization, then use previous symbol period aggregation algorithms accurately to estimate DC component;
Step 203: if previous symbol period does not complete bit synchronization, then whole digital subaudio frequency signal is calculated meansigma methods, Obtain rough estimate DC component;
Step 204: if synchronization control number of times is without departing from threshold value, then on the basis of current sampled point, sets early door scope With slow door scope, in the range of all sampled points remove DC component, and calculate early door range sample point and slow door model respectively Enclose the integrated value of sampled point;
Step 205: calculate the energy value of current sign length waveform, calculates error threshold according to this energy value;
Step 206: calculate the error of the early integrated value of door range sample point and slow door range sample point;
Step 207: if the error of the integrated value of door is just and to be more than error threshold the latest, then this sampling location is moved to left one Point, then performs subsequent cycle bit-synchronization algorithm, until exceeding synchronization control frequency threshold value or completing bit synchronization;
Step 208: if the error of the integrated value of door is more than error threshold for negative and absolute value the latest, then this sampling location is moved to right One point, then performs subsequent cycle bit-synchronization algorithm, until exceeding synchronization control frequency threshold value or completing bit synchronization;
Step 209: if the absolute value of the current error of the integrated value of door the latest is less than error threshold, then it is assumed that complete bit synchronization, Current sampled point is i.e. it is believed that the optimum sampling point of symbol.
A kind of digital subaudio frequency weak signal processing method the most according to claim 1, it is characterised in that use aggregation algorithms Calculating decision threshold has the following steps:
Step 301: search level maximum and the minima of the digital subaudio frequency signal received;
Step 302: set up on the basis of level maximum and just gather, sets up negative set on the basis of level minima, and by electricity The 1/2 of the distance of flat maximum and minima is set to distance decision threshold, and definition level maximum is positive benchmark, and level value is Little value is negative benchmark;
Step 303: if all symbol search do not complete, then calculate current sign level and the distance of positive benchmark;
Step 304: if the distance of current sign level and positive benchmark is less than the distance decision threshold arranged, then this symbol is returned Enter and just gather;
Step 305: if the distance of current sign level and positive benchmark is more than the distance decision threshold arranged, then calculate current sign Level and the distance of negative benchmark;
Step 306: if the distance of current sign level and negative benchmark is less than the distance decision threshold arranged, then this symbol is returned Enter to bear set;
Step 307: return step 303, calculating following symbol to positive benchmark and the distance of negative benchmark, and put into the set of correspondence;
Step 308: if all symbol search complete, then calculate each symbol in just gathering and close to the distance of other symbol;
Step 309: find and just gather the symbol that middle-range clutch is minimum, the level of this symbol is defined as positive level standard value;
Step 310: in the negative set of calculating, each symbol closes to the distance of other symbol;
Step 311: find the symbol that negative set middle-range clutch is minimum, the level of this symbol is defined as negative level standard value;
Step 312: calculate DC component by positive level standard value and negative level standard value, this DC component is set to numeral The decision threshold of subaudio frequency sampling judgement.
A kind of digital subaudio frequency weak signal processing method the most according to claim 1, it is characterised in that sampling judgement obtains Digital subaudio frequency code element information has the following steps:
Step 401: calculate the optimum sampling point position of current sign;
Step 402: calculate the average of point near optimum sampling point;
Step 403: if the average of point is more than the decision threshold of digital subaudio frequency sampling judgement near optimum sampling point, then adjudicate Current sign is 1;
Step 404: if the average of point is less than the decision threshold of digital subaudio frequency sampling judgement near optimum sampling point, then adjudicate Current sign is 0.
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