CN102831893A - Method for rapidly destroying broadcast audio file viruses - Google Patents
Method for rapidly destroying broadcast audio file viruses Download PDFInfo
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- CN102831893A CN102831893A CN2012101385178A CN201210138517A CN102831893A CN 102831893 A CN102831893 A CN 102831893A CN 2012101385178 A CN2012101385178 A CN 2012101385178A CN 201210138517 A CN201210138517 A CN 201210138517A CN 102831893 A CN102831893 A CN 102831893A
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Abstract
The invention provides a method for rapidly destroying broadcast audio file viruses. The method comprises adjusting quantization bits of audio frames in a coding stream one by one, and rewriting audio frames subjected to re-quantization coding into the coding stream; firstly selecting one channel sub-band from all channel sub-bands which are subjected to the current audio frame coding to carry out the adjustment that 1 is subtracted from bit distribution; and further selecting one or more channel sub-bands to carry out the adjustment that 1 is added into the bit distribution. According to the method, an audio does not need to be decoded into a PCM (pulse-code modulation) mode, but samples of a minority of channel sub-bands are simply processed at a compression domain, the calculation complexity is low, and audio files can be changed in a large scale, thereby guaranteeing that viruses hidden in the audio files can be destroyed.
Description
Technical field
The invention belongs to the digital audio processing technology, particularly the broadcast audio file processing technology.
Background technology
China's radio station network is through the development of decades, and progressively having formed with broadcast net (Intranet) and office net (outer net) is the network system of core.For the safety that guarantees that the radio station is professional and program broadcasts, the Intranet in SARFT(The State Administration of Radio and Television) regulation broadcasting station must be isolated with outer net, with attack and the viral infection that prevents malice.Yet on the other hand, a large amount of material information such as music program, interview recording, Press release etc. need exchange between intranet and extranet, and the Intranet of sealing will have a strong impact on the work efficiency and the program quality in radio station.For addressing this problem, safe isolation gap product arises at the historic moment and fast development, has obtained in the radio station at home using widely.
The security threat that comes from the internet mainly contains two kinds of assault and virus infectionses.Gateway provides the preventive effect to preceding a kind of assault, but powerless to a kind of virus infections in back.Virus is that carrier is propagated usually with the file; A lot of gateway equipment are taken precautions against file virus through adopting inspection file extension, the first-class form examination of file means; But these methods all can't fundamentally prevent virus infections, because virus possibly concealed in file data (like voice data).
The industry standard of the audio file formats that China broadcasting station is existing is the MPEG-I second layer form (being called for short mp2) in the ISO11172-3 standard.This formatted file is that (Moving Pictures Expert Group, a kind of voice data compressed encoding second layer standard method of MPEG) formulating is compressed original voice data and produced through adopting Motion Picture Experts Group.The mp2 audio file is made up of several audio frames, and the frame information of each audio frame comprises 6 parts such as frame head, position distribution, zoom factor selection, zoom factor, passage sub-band samples code word and auxiliary data, and is as shown in Figure 1.Wherein, passage sub-band samples code word has partly comprised the coded message of 32 frequency subband samples of left and right sides passage, and quantizes the position distribution portion decision of the used bit number of each passage sub-band samples by the front.The value that the position is distributed is big more, and the quantization bit position of distributing to the respective channel subband is just many more, and the expression of this passage sub-band samples value is just accurate more.Passage subband described herein is meant the subband of the respective channel of together being confirmed by channel number and subband numbering, as the passage subband (i, j) being expressed as channel number is i, subband is numbered the corresponding subband of j.
In standard, 36 samples of a passage subband have been divided into 12 joints (granule), and every joint comprises 3 continuous samples.Know that by Fig. 1 the sample code word of each passage subband is alternately arranged by joint in audio frame, i.e. the 1st joint of each passage subband of row earlier, the 2nd exclusive then joint is until the 12nd last joint.Passage sub-band samples code word is the main part of audio frame, and this part does not have verification scheme, can be distorted arbitrarily and is not realized.Therefore, audio file can become the excellent carrier of virus.The virus of concealing in audio file can utilize the leak of playout software to realize the attack to the Intranet machine, and this virus attack is to use the form checking method to take precautions against.The method that prevents the file virus invasion is at present divided two types substantially: detect virus and break virus.
The method that detects virus is to realize defence through detecting virus.Typical method has relative method and anti-tamper method.Relative method is found virus through file data and virus characteristic are compared.Be similar to and look into malicious software, this method need be brought in constant renewal in virus characteristic; Whether anti-tamper method is to add authentication or check code in advance hereof, distorted through the detection file to judge whether infective virus, and this method need deal with in the source that audio frequency is made, and uses also very inconvenient.
The method of break virus is a kind of defence method of active preferably, and its target is direct break virus.These class methods can be divided into the uncompressed domain processing and compression domain is handled two types.The uncompressed domain processing at first becomes the original pulse coded modulation with the audio decoder that compresses, and (Pulse Code Modulation, PCM) data change the PCM data through certain mode then, are reduced into compressed file through coding at last.This method has experienced decoding and cataloged procedure again, and computation complexity is higher; Compression domain is handled then direct bit stream to compressed encoding and is changed, as adding watermark, changing zoom factor or coding codeword etc.Though the compression domain disposal route is simple, often tonequality is had bigger damage.In a word, the break virus method mainly is faced with two technological difficulties at present, promptly how to reduce computation complexity and how in break virus, to guarantee tonequality as much as possible.These two problems are not well solved in existing method.
Summary of the invention
Technical matters to be solved by this invention provides a kind of new, fast and can guarantee the virus damage method of broadcast audio file tonequality.
The present invention solves the problems of the technologies described above the technical scheme that is adopted to be: a kind of method of rapid damage broadcast audio file virus; It is characterized in that; One by one the audio frame in the encoding stream is carried out the adjustment of quantization bit position; To write again in the encoding stream through the audio frame behind the weight coding, and audio frame carried out the bit adjustment may further comprise the steps:
A. resolve the frame head and the sub-band coding information of current audio frame, obtain the position distribution and sample code word of all passage subbands in the current audio frame;
B. in all passage subbands of participating in current audio frame coding, select 1 passage subband to carry out the position distribution earlier and subtract 1 adjustment, select one or more passage subbands to carry out the position again and divide with addition of 1 adjustment; The condition that selecteed passage subband satisfies is: carry out the position and distribute the passage subband that reduces adjustment distribution on the throne to reduce sample code word bits number that the back the discharged total bit number more than or equal to the sample code word of carrying out need expending after the position distributes all passage subbands that increase adjustment distribution on the throne to increase, and carry out the position and distribute the subband numbering that reduces adjustment to number different with the subband that increases adjustment;
C. to distributing the sample code word of all passage subbands of adjustment to decode and inverse quantization, generate the frequency samples of respective channel subband, according to coordination in adjusted minute these frequency samples are carried out re-quantization and coding again, generate new sample code word through the position;
All frame informations that d. will comprise new position distribution of passage subband and new samples code word generate the audio frame after changing by normal structure.
It is thus clear that the present invention need not become the PCM form with audio decoder, but only in compression domain the sample of a few passage subband is carried out simple process, its computation complexity is very low.
Because the sample code word of the different passage subbands in the audio frame is alternately arranged by joint; After the sample code word of some passage subbands in the audio frame changes; The code word that changes will be distributed in the whole passage sub-band samples code word zone of this audio frame; Again because this area size occupies the overwhelming majority of audio frame code stream, so the present invention can change audio file on a large scale, and then assurance destroys the virus of concealing in audio file.
In addition, the present invention does not change zoom factor in the audio frame, zoom factor selection etc. to the bigger information of quantization error influence, subtracts 1 adjustment and only the position of 1 passage subband is distributed, and makes that the tonequality loss is less.Simultaneously, the present invention distributes to other passage subbands as much as possible with the bit that discharges, and promptly the position of one or more passage subbands is distributed to add 1 adjustment, makes the tonequality loss obtain compensation.
In order further to reduce tonequality loss, when selecting to carry out the position and distribute the passage subband that reduces adjustment, preferentially select the position to distribute to reduce by 1 back institute to cause signal noise ratio (Signalto Noise Ratio, SNR) the minimum passage subband of slippage.This is the inspiration that from the position distribution principle of coding, obtains.In mp2 standard code algorithm, it is that iteration is accomplished that the position is distributed.Iteration each time, algorithm is distributed to the bit number of 1 unit has minimum masking noise than (Mask to Noise Ratio, passage subband MNR).Signal to noise ratio (S/N ratio) after masking noise is sheltered by other signals than the expression current demand signal, the noise after the more little expression of its value is sheltered is big more, and tonequality is poor more.Masking noise is than (Signal to Mask Ratio, difference SMR) decides, that is: by signal noise ratio and signal-to-mask ratio
MNR=SNR-SMR
Wherein SNR can directly obtain through tabling look-up, and SMR is calculated according to acoustic model by the decision of quantization bit position, after coding, can be considered to immobilize.Based on this result; This method is selected the minimum passage subband of SNR decline; And its position distribute reduced 1, and also just make the descend position of minimum passage subband of MNR distribute and reduce 1, promptly guaranteed again that from the acoustic model angle position distributes the subjective tonequality loss that causes for minimum.
Further, when selecting to carry out the passage subband of position distribution increase adjustment, the preferential maximum passage subband of increment of selecting a distribution to increase by 1 caused signal noise ratio makes the tonequality loss obtain the compensation of maximum.
Further, in order to remedy the tonequality loss to greatest extent, step a also calculates current audio frame unspent bit number when initial coding; In step b, the condition that selecteed passage subband satisfies becomes: carry out the position when distributing the passage subband that reduces adjustment distribution on the throne to reduce sample code word bits number that the back discharges with initial coding unspent bit number sum more than or equal to total bit number of the sample code word of carrying out need expending after the position distributes all passage subbands that increase adjustment distribution on the throne to increase.
The invention has the beneficial effects as follows that computation complexity is very low, and can change audio file on a large scale, the virus of concealing in audio file is destroyed with assurance; Compare with original, the subjective tonequality of the file after the processing is lost hardly; The position can't be recovered it after distributing adjustment, can avoid this method itself by assault; File after the processing kept original like original attributes such as size, port number, sampling rate, bit rates, can be by the decoding of any standard decoder.
Description of drawings
Fig. 1 is the bitstream data frame format of the MPEG-I second layer;
Fig. 2 is the inventive method process flow diagram;
Fig. 3 is the contrast of data before and after the inventive method is handled of the 2000th frame.
Embodiment
The present invention obtains the allocation information of former each passage subband from encoding stream, according to psychoacoustic model, the SELCH subband carries out the small adjustment that the position is distributed; And with weight coding behind the sample codeword decoding inverse quantization of respective sub-bands; Reorganize composite coding stream at last and write file, thereby change the original data to reach the purpose of break virus, as shown in Figure 2; By frame audio file is handled, wherein the treatment step to each audio frame comprises:
1. parsing frame head: from audio frame, resolve the audio format information that obtains being correlated with according to the ISO11172-3 standard;
2. resolve sub-band coding information: resolves from audio frame according to the ISO11172-3 standard and obtains that each passage subband position is distributed and information such as sample code word, wherein an allocation table of j subband of i passage be shown A (i, j); Calculate unspent bit number adb when initial coding;
3. SELCH subband and adjustment position are distributed: according to the numbering sublimit of the highest subband of current participation coding; Belong to [1 in the subband numbering; Sublimit] all passage subbands in select 2 or more passage subband; It is recorded among the passage sets of subbands M that chooses, and the passage subband of choosing is carried out the position distribute adjustment, concrete step is following:
3.1., calculate if the slippage that reduces by 1 caused signal noise ratio is distributed in current position, and it recorded among the key assignments member of a structural array snr_dec to each passage subband.This structural array has also write down corresponding channel number ch and the subband numbering sub information of key assignments key; This snr_dec array is pressed key value size, sort from small to large; The item identical to the key value determines its sequencing at random;
3.2., calculate if the increment of increase by 1 caused signal noise ratio is distributed in position that will be current, and it recorded among the key assignments member of another structural array snr_enc to be assigned as 0 and reached each passage subband the maximum allocated except that the position.This structural array has also write down corresponding channel number ch and the subband numbering sub information of key assignments key; This snr_enc array is pressed key value size, sort from big to small; The item identical to the key value determines its sequencing at random;
3.3. establish the subscript of browsing that m and n are respectively array snr_dec and snr_enc.Put initial value m=1, n=1, M are empty set;
3.4. get the m item snr_dec [m] of snr_dec; This is corresponding to the minimum passage subband of SNR slippage; If its channel number and subband numbering are expressed as snr_dec [m] .ch and snr_dec [m] .sub respectively, calculate if its present bit is distributed the releasable bit number DEC in minimizing 1 back; Calculating current available bit number avail_bits is that the current bit number that discharges adds that initial coding does not use up the summation of bit number, i.e. avail_bits=DEC+adb;
3.5. each n to snr_enc does: the channel number and the subband numbering of getting n item snr_enc [n]; Be expressed as snr_enc [n] .ch and snr_enc [n] .sub respectively; Calculate if its present bit distribution is increased the bit number ENC that need consume after 1; And judge if the bit number that current available bit number consumes more than or equal to needs and (to be avail_bits >=ENC) and to discharge and to consume the subband numbering of bit number inequality; Then SELCH snr_enc [n] .ch, subband snr_enc [n] .sub are as carrying out the passage subband that the position is distributed increases adjustment; And its band joined among the M, upgrade avail_bits simultaneously, make avail_bit=avail_bits-ENC;
3.6 if the M non-NULL, then SELCH snr_dec [m] .ch, subband snr_dec [m] .sub distribute the passage subband that reduces adjustment as carrying out the position, and its band is joined among the M, otherwise put m=m+1, and forward the selection that step 3.4 is carried out next round to;
3.7 the position distribution to carrying out 1 a passage subband that distributes the minimizing adjustment among the M subtracts 1; Distribute the position of one or more passage subbands that increase adjustment to divide to carrying out the position among the M with addition of 1;
4. decoding inverse quantization sample code word: according to the ISO11172-3 standard sample code word of all passage subbands among the M is decoded and inverse quantization, obtain the correspondent frequency sample;
5. weight coding frequency samples: according to coordination in adjusted minute, the frequency samples of all passage subbands among the M is carried out re-quantization and coding, generate new sample code word according to the ISO11172-3 standard;
6. frame encoding stream format: all frame informations that will comprise new position distribution of passage subband and new samples code word according to the ISO11172-3 standard write in the encoding stream.
Embodiment
Mp2 audio frequency " Fei Xiang-Cloud Of Hometown .mp2 " file with standard 256kbit/s is an example as the original audio of importing.This audio frequency duration is 4 minutes and 24 seconds, and totalframes is 11020 frames.This embodiment is that unit handles to the audio file of input by coded frame, and single treatment one frame is up to handling 11020 whole frames.Be example with 2000 frames below, the concrete implementation procedure of processing is following:
1. parsing frame head: resolve from audio frame according to the ISO11172-3 standard that to obtain port number be 2, sampling rate is 48000kHz, and bit rate is 256kbps, and the numbering sublimit that participates in the highest subband of coding is 27, and decoding table is the 0th information such as table;
2. resolve sub-band coding information: resolves from audio frame that the position that obtains each passage subband is distributed and information such as sample code word according to the ISO11172-3 standard, wherein the value of each passage subband position distribution array A is { { 6,7,5,7,8,7,6,5,5,4,3; 2,1,1,1,1,1,1,1,0,0,0,0; 0,0,0,0}, { 6,7,5,8,8,6,5,6; 4,5,4,3,2,3,3,2,2,1,1,1; 0,0,0,0,0,0,0}}, wherein two groups of data position distribution of two each 27 subbands of passage about correspondence respectively, for example the value of A (2,13) is that 2 expressions are assigned as 2 when the position of prepass subband (2,13); Calculate that the unspent bit number adb of this frame is 4 bits when initial coding;
3. SELCH subband and adjustment position are distributed: according to psychoacoustic model, SELCH subband from 54 candidates of subband 1 to the subband 27 that comprises left and right sides passage is adjusted its distribution, and concrete step is following:
3.1. to each passage subband; Calculate if the slippage that reduces by 1 caused signal noise ratio is distributed in current position; Fill in the key assignments of structural array snr_dec; Channel number and 3 members of subband numbering sort to the snr_dec key value with the quicksort method from small to large, obtain orderly snr_dec; Wherein the value of first snr_dec [1] be 4.00,2,13}, the expression key assignments (or SNR slippage) be 4.00, channel number be 2 with subband be numbered 13, this is corresponding to the SNR slippage of current minimum.
3.2. to be assigned as 0 and be assigned to 11 passage subbands maximum except that the position; Calculate if the increment that increases by 1 caused signal noise ratio is distributed in current position; Fill in the key assignments of structural array snr_enc; Channel number and 3 members of subband numbering sort to the snr_enc key value with the quicksort method from big to small, obtain orderly snr_enc; The value of its first snr_enc [1] be 4.84,1,11}, the expression key assignments (or SNR increment) be 4.84, channel number be 1 with subband be numbered 11, this is corresponding to the SNR increment of current maximum;
3.3. that puts array snr_dec browses subscript m=1, snr_enc browses subscript n=1, and M is an empty set;
3.4. investigate the m item of snr_dec array, for the first time, promptly, calculate if it is 24 that its distribution is reduced to 1 releasable bit number DEC from 2 to the 13rd subband of the 2nd passage of snr_dec [1] representative, calculating avail_bits is 28;
3.5. investigate each n of snr_enc one by one, it is calculated if its present bit is distributed the bit number ENC that need consume after the increase by 1, and Rule of judgment avail_bits >=ENC and snr_enc [n] .sub ≠ snr_dec [m] .sub.After finishing, investigation finds to have only the 1st to satisfy condition; Its corresponding ENC is 12, and channel number is 1, and subband is numbered 11; Then choose the 11st subband of the 1st passage to increase the passage subband of adjusting as carrying out bit; And passage subband (1,11) joined among the M, avail_bits is updated to 16;
3.6 because the M non-NULL is selected the 13rd subband of the 2nd passage to reduce the passage subband of adjusting as carrying out bit, and passage subband (2,13) band is joined among the M;
3.7 being distributed, the position of passage subband (2,13) subtracts 1, i.e. A (2,13)=A (2,13)-1; The position of passage subband (1,11) is divided with addition of 1, i.e. A (1,11)=A (1,11)+1;
4. decoding inverse quantization sample code word: according to the ISO11172-3 standard sample code word of passage subband (2,13) and passage subband (1,11) is decoded and inverse quantization, obtain the correspondent frequency sample;
5. weight coding frequency samples: according to the updated space assignment information, carry out re-quantization and coding, generate new sample code word according to the frequency samples of ISO11172-3 standard channel subband (2,13) and passage subband (1,11);
6. frame encoding stream format: all frame informations that will comprise new position distribution of passage subband and new samples code word according to the ISO11172-3 standard write in the encoding stream, and unspent 16 bits are filled up with 0.
From maintenance and three aspects of processing speed of change, tonequality and the form of file validity of the present invention is verified below.Keep and the processing speed link in test tonequality qualifying formula; Except method of the present invention; Also will introduce a kind of scheme that simplifies; Promptly remove the 4th and 5 two step (distributing the inverse quantization decoding and the weight coding step of the sample code word of the passage subband of adjusting) of embodiment to carrying out the position; Ascend the throne and the passage subband that changes is not carried out the re-quantization coding after distributing adjustment, and directly be made into frame, like this allocation information that the passage subband is only arranged of change and be not enough to break virus with former subband sample Codeword Sets.Adopt subjective audio frequency quality assessment (the Perceptual Evaluation of Audio Quality that describes among the standard I TU-R BS.1387 in the assessment of acoustics link; PEAQ) method, it be in the current audio Objective Quality Assessment method with the subjective assessment the highest method of the degree of correlation as a result.This method utilizes people's ear subjective perception property calculation to go out the masking threshold and the distortion threshold value of signal; Adopt artificial neural network to merge then and an evaluating ODG (Object Difference Grade), this enforcement just adopts this parameter to come audio quality is carried out objective evaluation.
1. the change of file
File and original after at first will handling are carried out the data comparative analysis.Know that by preceding surface analysis data movement can occur in the position of frame distributes and passage sub-band samples code word two parts, and the size of data movement reduces institute to discharge bit number relevant with a difference diff_bit of the spent bit number of distribution increase main the distribution with the position.Because the code word of each passage sub-band samples is alternately to arrange in encoding stream, in a single day parameter d iff_bit is not 0, will cause the serious dislocation of handling the back bit arrangement, and the large-scale change that brings frame data.
Fig. 3 has shown the data contrast situation of the 2000th frame before and after handling of listed, and the background that wherein changes byte shows with shade.The bit number that this framing bit distributes minimizing to be discharged is 24, and the position spent bit number of distribution increase is 12, i.e. diff_bit=12.Can see in the change of frame previous section 2 bytes being arranged, corresponding to the allocation information of reformed two passage subbands.In the passage sub-band samples code word zone that accounts for the audio frame overwhelming majority, data nearly all have been changed, and this is the effect of the serious dislocation of bit arrangement.To entire frame, the data variation rate reaches 84.77%, and constant maximum region appears at the previous section of frame continuously, and its length is 95 bytes.
2. the maintenance of tonequality and form
With same standard decoder to former mp2 file and handle the file obtain through the inventive method and its method that simplifies and decode.Table 1 has been listed 3 decoded tonequality contrasts of audio file, and evaluating is ODG.This parameter value is more near 0, and expression tonequality is good more.
The tonequality of table 1 original, the file after handling through the present invention and the method that simplifies thereof relatively
| Original | After the present invention handles | After the method that simplifies is handled |
| -3.326 | -3.330 | -3.354 |
Can see, very little through the tonequality difference of audio frequency after the inventive method processing and former audio frequency, and the method that simplifies is inconsistent with the distribution of new position owing to former sample code word corresponding quantitative number of bits, so the tonequality after its processing has decline slightly.In addition, the inventive method does not change the frame head of original audio frame, and has kept frame sign constant, so the file after the wherethrough reason has kept original attributes such as the size, port number, sampling rate, bit rate of original.
3. processing speed
With the inventive method and its method that simplifies the testing audio file is become sign indicating number and handle, table 2 has been listed the result of handled time (unit is second), and this result has comprised the time that reading and writing of files is used.Test machine is Intel (R) i3 processor, and dominant frequency is 2.53GHz.For ease of relatively, also listed in the table 2 the pairing former not compacted voice file of test file is carried out the coding of standard and former mp2 file is carried out the processing time that standard is decoded.
Table 2 the present invention compared with the processing time that it simplifies method and four kinds of processing of coding and decoding
| Coding | Decoding | The present invention handles | The method that simplifies is handled |
| 6.328 | 4.282 | 0.406 | 0.360 |
Can see that the processing speed that simplifies method is the fastest during each is handled.With respect to Code And Decode; The processing speed of the inventive method also was very fast, is respectively 15.59 and 10.55 times of Code And Decode, to 4 minutes 24 seconds audio frequency; Only just accomplished with 0.406 second, the real-time speed of its processing than (audio frequency T.T./processing time) up to 650.25.
Claims (4)
1. the method for a rapid damage broadcast audio file virus; It is characterized in that; One by one the audio frame in the encoding stream is carried out the adjustment of quantization bit position, will write again in the encoding stream, audio frame is carried out the bit adjustment may further comprise the steps through the audio frame behind the weight coding:
A. resolve the frame head and the sub-band coding information of current audio frame, obtain the position distribution and sample code word of all passage subbands in the current audio frame;
B. in all passage subbands of participating in current audio frame coding, select 1 passage subband to carry out the position distribution earlier and subtract 1 adjustment, select one or more passage subbands to carry out the position again and divide with addition of 1 adjustment; The condition that selecteed passage subband satisfies is: carry out the position and distribute the passage subband that reduces adjustment distribution on the throne to reduce sample code word bits number that the back the discharged total bit number more than or equal to the sample code word of carrying out need expending after the position distributes all passage subbands that increase adjustment distribution on the throne to increase, and carry out the position and distribute the subband numbering that reduces adjustment to number different with the subband that increases adjustment;
C. to distributing the sample code word of all passage subbands of adjustment to decode and inverse quantization, generate the frequency samples of respective channel subband, according to coordination in adjusted minute these frequency samples are carried out re-quantization and coding again, generate new sample code word through the position;
All frame informations that d. will comprise new position distribution of passage subband and new samples code word generate the audio frame after changing by normal structure.
2. a kind of according to claim 1 method of rapid damage broadcast audio file virus is characterized in that, when selecting to carry out the passage subband of a distribution minimizing adjustment, a preferential selection distribution reduces the passage subband of 1 signal noise ratio slippage that the back causes minimum.
3. like the method for the said a kind of rapid damage broadcast audio file virus of claim 2; It is characterized in that; When selecting to carry out the passage subband of position distribution increase adjustment, the preferential maximum passage subband of increment of selecting a distribution to increase by 1 caused signal noise ratio.
4. a kind of according to claim 1 method of rapid damage broadcast audio file virus is characterized in that, step a also calculates current audio frame unspent bit number when initial coding; In step b, the condition that selecteed passage subband satisfies becomes: carry out the position when distributing the passage subband that reduces adjustment distribution on the throne to reduce sample code word bits number that the back discharges with initial coding unspent bit number sum more than or equal to total bit number of the sample code word of carrying out need expending after the position distributes all passage subbands that increase adjustment distribution on the throne to increase.
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| CN107609398A (en) * | 2017-08-15 | 2018-01-19 | 北京英夫美迪科技股份有限公司 | To breaking method and equipment viral in the audio file of linear PCM coding |
| CN107609398B (en) * | 2017-08-15 | 2019-10-29 | 北京英夫美迪科技股份有限公司 | To breaking method and equipment viral in the audio file of linear PCM coding |
| WO2020188421A1 (en) * | 2019-03-17 | 2020-09-24 | International Business Machines Corporation | Low delay content disarm and reconstruction (cdr) of live streaming video |
| GB2595426A (en) * | 2019-03-17 | 2021-11-24 | Ibm | Low delay content disarm and reconstruction (CDR) of live streaming video |
| GB2595426B (en) * | 2019-03-17 | 2022-04-13 | Ibm | Low delay content disarm and reconstruction (CDR) of live streaming video |
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