CN101394402A - Method for fast code changing in large range to audio information to break virus - Google Patents
Method for fast code changing in large range to audio information to break virus Download PDFInfo
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- CN101394402A CN101394402A CNA2008101430995A CN200810143099A CN101394402A CN 101394402 A CN101394402 A CN 101394402A CN A2008101430995 A CNA2008101430995 A CN A2008101430995A CN 200810143099 A CN200810143099 A CN 200810143099A CN 101394402 A CN101394402 A CN 101394402A
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Abstract
The invention discloses a method for rapidly code-variable destroying virus at large range for audio information. The method comprises the following steps: respectively decoding a mp2 audio frame to bit allocation information, scaling factor information and sample code information according to the standard of iso11171-3, subjecting the bit allocation, the scaling factor and the sample code to unified code-variable re-calculation to change all bits of each coded data frame except the initial 20 bits of the mp2 audio information, and coding storage bit allocation information, quantification factor selection information, quantification factor information and sub-band quantification code information after the re-calculation together with an original header and additional information to generate a frame format. The method can achieve large comprehensive variable coding rate (up to 98.57%) of each frame, and the variable coding rate of the audio coding section is up to 100%.
Description
Technical field
The invention provides a kind of method to audio-frequency information fast code changing in large range break virus.
Background technology
MpegI is a kind of video/audio data Compression Standard that dynamic image expert group formulates, MpegILayer2 abbreviates mp2 as, the mp2 compressed encoding is one of voice data compression standard wherein (second layer), its coding after data file generally with mp2 as extension name.
The principle of Mp2 compressed encoding is:
The first step is sampled audio signal earlier according to the even time interval, the data file that obtains after the sampling generally is stored as pcm (Pulse Code Modulation) coding, generally can be play-overed by hardware.
In second step, adopt the Fourier Time frequency Filter that acoustic coding (pcm) is converted to frequency-region signal by time-domain signal then.Wherein the Fourier Time frequency Filter is a kind of typical high-pass and low-pass filter, filtered frequency-region signal is divided into 32 subbands (sub band), low frequency sub-band has comprised people wishes most information of keeping on the contrary, to have comprised less useful information and noise in the high-frequency sub-band.
The 3rd step, frequency domain data is carried out quantizing factor select, use the frequency domain data behind the quantizing factor " convergent-divergent " can be with less unsigned int data representation.
The 4th step, frequency domain data is carried out bit distribute, the subband that promptly contains much information obtains more bit, and the subband that amount of information is little obtains less bit or directly abandons.
In the 5th step, mp2 file frame coding generates frame head, the stored bits allocation information, and zoom factor is selected information, scale factor information and subband scalable coded information.
Frame data format behind the coding such as Fig. 1: every frame distributes (ALLOC), zoom factor to select (SCFSI), zoom factor (SCALEFAC), sample coding (SAMPLES), additional information (ANC) by frame head (HEADER), the position of fixed size.The Mp2 file is made of described frame (as Fig. 1) order of some.
But because the characteristics of MPEG I Layer 2 audio files self, the audio samples data of every frame 650 bytes possess replaceability, and the audio file of replacing behind code according to this rule still is considered to legal audio file, because can't detect with current antivirus software and other tool software.Therefore MPEG ILayer 2 audio files might become the excellent carrier of malicious attacker, utilize it to embed some malicious codes or as the carrier of fallacious message and instrument, enter Intranet legally by outer net, in case utilize the leak of player or other software to obtain the control of machine, and then Intranet attacked.
In order to ensure the transmission data security of the audio file between outer net (office net) and the Intranet, be badly in need of wanting a kind of MPEG of conversion by force I Layer 2 audio samples Methods for Coding now, initiatively destroy the malicious code that might conceal in audio file.
Summary of the invention
Defective at the prior art existence, in conjunction with the MP2 compression coding technology, the present invention aims to provide a kind of method to audio-frequency information fast code changing in large range break virus, can be under the unconverted prerequisite of file attribute, effectively improve escape rate and escape speed, thereby stopped the possibility of virus and malicious code existence.
To achieve the above object of the invention, the technical solution used in the present invention is: a kind of method to audio-frequency information fast code changing in large range break virus, the mp2 audio frame is distinguished decoded bit assignment information, scale factor information, sample coded message according to the iso11172-3 standard, the change yardage that last rheme distribution, zoom factor and sample coding are carried out unification is calculated again, make behind the mp2 coding every frame data except that beginning 20 surplus all changes of byte; Bank bit assignment information after will becoming yardage at last and calculating, quantizing factor is selected information, and quantizing factor information and quantized subband coded message are together with former frame head and the additional information delta frame form of encoding.
The step that wherein said change yardage is calculated into:
1) each audio frame is transformed to 32 subbands by fft filters, the different frequency feature that has according to each subband again selects wherein the data of high-frequency sub-band and carries out conversion;
2) perceptual audio coder in the mp2 compression standard is at first analyzed the frequency and the amplitude of above-mentioned high-frequency sub-band, and the auditory perception model with itself and people compares then, removes the irrelevant part and the statistical redundancy part of audio signal by auditory perception model; By analysis, find again apart from the near subband of masking value in the iso11172-3 psychoacoustic model to high-frequency sub-band;
3) utilize the affluence in frame space, the subband that above-mentioned steps is found carries out the position and distributes and expand;
4) zoom factor and sample coding are repeated above-mentioned shift step, realize that the change yardage of contraposition distribution, zoom factor and sample coding unification is calculated.
Operation principle of the present invention is as described below:
The principle that mp2 becomes sign indicating number into, each Frame is taked the following step:
The first step is according to iso11172-3 standard decoding allocation information;
Second step is according to iso11172-3 standard decoding scale factor information;
The 3rd step is according to the sample coding after the iso11172-3 standard decoding quantification;
The 4th step, last rheme distribution, zoom factor and sample coding are carried out the change code weight of unification and calculate, make behind the mp2 coding every frame data except that beginning 20 surplus all changes of byte;
In the 5th step, with the bank bit assignment information after the re-computation, quantizing factor is selected information, and quantizing factor information and quantized subband coded message are together with former frame head and the additional information delta frame form of encoding.
For the consideration of conversion rate, conversion can not relate to the FFT conversion and bit distributes (the most consuming time).Therefore the step that described change code weight is calculated into:
At first, each audio frame is transformed to 32 subbands by fft filters, and each subband has different frequecy characteristics, and the data energy that low frequency sub-band comprises is higher, if add noise therein to the tonequality influence greatly; High-frequency sub-band antithesis, it is little to tonequality influence to introduce noise therein.Therefore, the present invention chooses the data of high-frequency sub-band and carries out conversion, to avoid the change of tonequality;
The second, the perceptual audio coder in the mp2 compression standard is at first analyzed the frequency and the amplitude of input signal, and the auditory perception model with itself and people compares then.Encoder is removed the irrelevant part and the statistical redundancy part of audio signal with this model.Although this method diminishes, the decline of the imperceptible encoded signal quality of people's ear.The present invention finds apart from the near subband of masking value in the iso11172-3 psychoacoustic model by the analysis to each subband.
The 3rd, the present invention utilizes the affluence in frame space, the subband that finds in the second is carried out the position distribute expansion, neither changes tonequality, also can make the sample coding produce " phase drift ".
The 4th, quantizing factor and sample become the frequency domain character that sign indicating number has reflected audio signal together, and the present invention becomes sign indicating number by the consistent change quantization factor and sample, can change frame and become sign indicating number under the condition that does not change tonequality.
In sum, the method to audio-frequency information fast code changing in large range break virus provided by the invention can reach following basic demand:
1. the comprehensive variable code rate of every frame (comprising the frame head part) reaches 98.57%; The variable code rate of audio coding part reaches 100%.
2, file and the original tonequality sense organ indistinction behind the change sign indicating number;
3. the file behind the change sign indicating number is compared original attributes such as not changing file size, compression ratio, sample rate, frame energy, waveform with original;
4. becoming file behind the sign indicating number uses original decoding software can normal decoder and editor.
Description of drawings
Fig. 1 is the frame data format behind the Mp2 compressed encoding in the prior art;
Fig. 2 is the schematic flow sheet of the method for the invention.
Embodiment
Present embodiment is that the MPEG1Layer2 audio file of 768 bytes is a test file with every frame length, adopts the code changing method of scrambling code changing method of the present invention and other four major types prior aries to become sign indicating number respectively its all bytes, and is specific as follows:
1, the method to audio-frequency information fast code changing in large range break virus of the present invention, step comprises:
1) reads in a plurality of mp2 Frames to handling buffering area;
2) successively each frame is carried out following operation:
A) frame head is resolved, and the content that parses comprises: synchronization character, audio types, redundancy protecting position, code check, sample frequency, extra groove, privately owned mark, pattern, copyright protection position, copy flag position, separate and increase the weight of type.If wherein synchronization character, audio types mistake occurs then jump out change coded program, reporting errors.
B) resolve allocation information according to the value of redundancy protecting position, pattern, code check and sample frequency;
C) resolve zoom factor according to the value of redundancy protecting position, pattern, code check and sample frequency and select information;
D) resolve scale factor information according to the value of redundancy protecting position, pattern, code check and sample frequency;
E) according to the value of redundancy protecting position, pattern, code check, sample frequency, and according to the position of corresponding sample point distribute, zoom factor is selected and scale factor information is resolved sample and become a sign indicating number information;
F) parse for frame satellite information;
G) for the allocation information of this frame, find the maximum subband of redundancy, this subband punctured bit is distributed;
H) select information for the zoom factor of this frame, select high-frequency sub-band and carry out the displacement of zoom factor selection information;
I) for the scale factor information of this frame, selectively carry out the zoom factor increasing or decreasing;
J) sample for this frame becomes sign indicating number information, at first reverts to the value (being Frequency Domain Coding) of PCM sign indicating number after filter transform, carries out sample coding re-computation according to new zoom factor and allocation information again, and concrete steps are:
I. obtain (complement code) true form of sample point;
Ii. obtain the sequence number in the QUANCLASS table in the iso11172-3 received text;
Iii. calculate the signal of Time frequency Filter output;
Iv. re-quantization;
V. calculate complement code and obtain the sample encoded radio;
K) new frame head, allocation information, zoom factor are selected mp2 frame after information, scale factor information, sample coding and frame satellite information are combined as conversion;
3) return the 1st) go on foot up to handling all mp2 frames.
2. the special-purpose processing mode that becomes the sign indicating number technology
1. adopt a PC that performance is medium, handling implement is the change sign indicating number software DEMO of our special exploitation;
2. utilize software DEMO that test file is carried out the high density scrambling and become the sign indicating number processing, processing speed is carried out measurements and calculations; File tonequality, file attribute and original after handling are compared evaluation and test, and measure the variable code rate that becomes the sign indicating number technology with third party software ultra Edit;
3. the processing mode of audio frequency recompile
1. adopt a PC that performance is medium, handling implement is a SoundPaint audio work station software;
2. utilize audio edited software that test file is transferred the level recompile respectively and handle, the processing speed that becomes sign indicating number is carried out measurements and calculations with transferring high bass recompile; File tonequality, file attribute and original after handling are compared evaluation and test, and measure variable code rate with third party software ultra Edit;
4. the simulation balance is to the mode of record
1. a PC main frame loads the pcx22 sound card, is used for audio signal output.Another PC main frame loads the pcx924 sound card and is used for the audio signal input.2. the PCX22 sound card adopts the simulation balance to be connected with the PCX924 sound card.Middle without sound console, directly to record.
3. Analog records, tone frequency channel wire butt joint schematic diagram.
4. PCX22 sound card displaying audio file adopts diagram to carry software NP Play, and PCX924 sound card recording audio file also adopts diagram to carry software NP Play.
5. to the speed of record is carried out measurements and calculations; File tonequality, file attribute and original after handling are compared evaluation and test, and measure variable code rate with third party software ultra Edit;
5. digital AES is to the mode of record
1. a PC main frame loads the pcx22 sound card, is used for audio signal output.Another PC main frame loads the pcx924 sound card and is used for the audio signal input.2. the PCX22 sound card adopts digital AES to be connected with the PCX924 sound card.Middle without sound console, directly to record.
3. Digital records, tone frequency channel wire butt joint schematic diagram.
4. PCX22 sound card displaying audio file adopts diagram to carry software NP Play, and PCX924 sound card recording audio file also adopts diagram to carry software NP Play.
5. to the speed of record is carried out measurements and calculations; To file tonequality, file attribute and the original after handling
Compare evaluation and test, and measure variable code rate with third party software ultra Edit.
Above-mentioned comparison and the content of evaluation and test comprise the contrast of three class data: escape speed, escape rate, the file after handling and base attribute, the also comparison of file attributes such as proper mass, compression ratio, sample rate, frame energy of audio frequency of original document.Specifically relatively see the following form:
The concrete data of the escape speed evaluating standard that wherein adopts in the said method are as follows:
| Grade | Excellent | Very | Pass | Difference |
| Escape speed | ≥4MB/s | ≥2MB/s | 1MB/s | ≤0.5MB/s |
Claims (2)
1, a kind of method to audio-frequency information fast code changing in large range break virus, it is characterized in that, the mp2 audio frame is distinguished decoded bit assignment information, scale factor information, sample coded message according to the iso11172-3 standard, the change yardage that last rheme distribution, zoom factor and sample coding are carried out unification is calculated again, make behind the mp2 coding every frame data except that beginning 20 surplus all changes of byte; Bank bit assignment information after will becoming yardage at last and calculating, quantizing factor is selected information, and quantizing factor information and quantized subband coded message are together with former frame head and the additional information delta frame form of encoding.
2, according to the described method of claim 1 to audio-frequency information fast code changing in large range break virus, it is characterized in that step that described change yardage calculates into:
1) each audio frame is transformed to 32 subbands by fft filters, the different frequency feature that has according to each subband again selects wherein the data of high-frequency sub-band and carries out conversion;
2) perceptual audio coder in the mp2 compression standard is at first analyzed the frequency and the amplitude of above-mentioned high-frequency sub-band, and the auditory perception model with itself and people compares then, removes the irrelevant part and the statistical redundancy part of audio signal by auditory perception model; By analysis, find again apart from the near subband of masking value in the iso11172-3 psychoacoustic model to high-frequency sub-band;
3) utilize the affluence in frame space, the subband that above-mentioned steps is found carries out the position and distributes and expand;
4) zoom factor and sample coding are repeated above-mentioned shift step, realize that the change yardage of contraposition distribution, zoom factor and sample coding unification is calculated.
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| CNA2008101430995A CN101394402A (en) | 2008-10-13 | 2008-10-13 | Method for fast code changing in large range to audio information to break virus |
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| CNA2008101430995A CN101394402A (en) | 2008-10-13 | 2008-10-13 | Method for fast code changing in large range to audio information to break virus |
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Cited By (6)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN102831893A (en) * | 2012-05-07 | 2012-12-19 | 电子科技大学 | Method for rapidly destroying broadcast audio file viruses |
| CN105205102A (en) * | 2015-08-25 | 2015-12-30 | 中山大学 | Method for identifying plurality of times of compression of ACC format of digital audio |
| CN107430864A (en) * | 2015-03-31 | 2017-12-01 | 高通技术国际有限公司 | The embedded code in audio signal |
| 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 |
| CN110210230A (en) * | 2019-05-14 | 2019-09-06 | 深圳市腾讯网域计算机网络有限公司 | Improve method, apparatus, electronic equipment and the storage medium of security of system |
| CN111178540A (en) * | 2019-12-29 | 2020-05-19 | 浪潮(北京)电子信息产业有限公司 | Training data transmission method, device, equipment and medium |
-
2008
- 2008-10-13 CN CNA2008101430995A patent/CN101394402A/en active Pending
Cited By (10)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN102831893A (en) * | 2012-05-07 | 2012-12-19 | 电子科技大学 | Method for rapidly destroying broadcast audio file viruses |
| CN102831893B (en) * | 2012-05-07 | 2014-07-16 | 电子科技大学 | Method for rapidly destroying broadcast audio file viruses |
| CN107430864A (en) * | 2015-03-31 | 2017-12-01 | 高通技术国际有限公司 | The embedded code in audio signal |
| CN105205102A (en) * | 2015-08-25 | 2015-12-30 | 中山大学 | Method for identifying plurality of times of compression of ACC format of digital audio |
| CN105205102B (en) * | 2015-08-25 | 2018-08-14 | 中山大学 | A method of differentiating that digital audio AAC formats repeatedly compress |
| 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 |
| CN110210230A (en) * | 2019-05-14 | 2019-09-06 | 深圳市腾讯网域计算机网络有限公司 | Improve method, apparatus, electronic equipment and the storage medium of security of system |
| CN110210230B (en) * | 2019-05-14 | 2021-10-22 | 深圳市腾讯网域计算机网络有限公司 | Method and device for improving system security, electronic equipment and storage medium |
| CN111178540A (en) * | 2019-12-29 | 2020-05-19 | 浪潮(北京)电子信息产业有限公司 | Training data transmission method, device, equipment and medium |
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Open date: 20090325 |