CN102831893B - Method for rapidly destroying broadcast audio file viruses - Google Patents
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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 Digital Audio-Frequency Processing Techniques, particularly broadcast audio file processing technology.
Background technology
China's radio station network, through the development of decades, has progressively formed to broadcast net (Intranet) and Office Network (outer net) network system as core.For the safety that ensures that radio station business and program broadcast, SARFT(The State Administration of Radio and Television) specifies that the Intranet in broadcasting station must isolate with outer net, to prevent the attack of malice and viral infection.But on the other hand, a large amount of material information need to exchange as music program, interview recording, Press release etc. between intranet and extranet, the Intranet of sealing will have a strong impact on work efficiency and the program quality in radio station.For addressing this problem, safety isolation network gate product arises at the historic moment and fast development, is widely used at home in radio station.
The security threat that comes from internet mainly contains two kinds of assault and virus infectionses.Gateway provides the preventive effect to front a kind of assault, but helpless to rear a kind of virus infections.Virus is propagated taking file as carrier conventionally, a lot of gateway equipment checks that by employing file extension, the first-class form examination of file means take precautions against file virus, but these methods all cannot fundamentally prevent virus infections, because virus may be concealed in file data (as voice data).
The industry standard of the existing audio file formats in China broadcasting station is the MPEG-I second layer form (being called for short mp2) in ISO11172-3 standard.This formatted file is that a kind of voice data compressed encoding second layer standard method by adopting Motion Picture Experts Group (Moving Pictures Expert Group, MPEG) to formulate is compressed and produced original voice data.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, as shown in Figure 1.Wherein, the coded message of 32 frequency subband samples that passage sub-band samples code word part has comprised left and right passage, is determined by position distribution portion above and quantize each passage sub-band samples bit number used.The value that position is distributed is larger, and the quantization bit position of distributing to respective channel subband is just more, and the expression of this passage sub-band samples value is just more accurate.Passage subband described herein refers to by channel number and the together subband of definite respective channel of subband numbering, is i as passage subband (i, j) is expressed as channel number, and subband is numbered the subband that j is corresponding.
In standard, 36 samples of a passage subband, are divided for 12 joints (granule), and every joint comprises 3 continuous samples.Known by Fig. 1, the sample code word of each passage subband in audio frame by joint alternative arrangement, the Section 1 of first arranging each passage subband, then exclusive Section 2, until last Section 12.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 not realized.Therefore, audio file can become viral excellent carrier.The virus of concealing in audio file can utilize the leak of playout software to realize the attack to Intranet machine, cannot take precautions against and this virus attack is using form checking method.The method that prevents at present file virus invasion is divided two classes substantially: detect virus and break virus.
Detecting viral method is to realize defence by detecting virus.Typical method has relative method and anti-tamper method.Relative method is by comparing to find virus by file data and virus characteristic.Be similar to virus investigation software, this method need to be constantly updated virus characteristic; Anti-tamper method is to add hereof in advance certification or check code, whether is tampered to judge whether to infect virus by detecting file, and the source that this method need to be made at audio frequency deals with, and uses also very inconvenient.
The method of break virus is a kind of active defense method preferably, and its target is direct break virus.These class methods can be divided into uncompressed domain processing and compression domain is processed two classes.Uncompressed domain is processed and first the audio decoder of compression is become to original pulse coded modulation (Pulse Code Modulation, PCM) data, then changes PCM data by certain mode, is finally reduced into compressed file by coding.This method has experienced decoding and cataloged procedure again, and computation complexity is higher; Compression domain is processed and directly the bit stream of compressed encoding is changed, as adds watermark, change zoom factor or coding codeword etc.Although compression domain disposal route is simple, often tonequality is had to larger damage.In a word, break virus method is mainly faced with two technological difficulties at present, how to reduce computation complexity and how in break virus, to ensure as much as possible tonequality.These two problems are not well solved in existing method.
Summary of the invention
Technical matters to be solved by this invention is to provide a kind of viral destruction methods newly, that also can ensure fast broadcast audio file tonequality.
The present invention for solving the problems of the technologies described above adopted technical scheme is: a kind of method of rapid damage broadcast audio file virus, it is characterized in that, one by one the audio frame in encoding stream is carried out the adjustment of quantization bit position, audio frame after heavy quantization encoding is re-write in encoding stream, audio frame is carried out to bit adjustment and comprise the following steps:
A. resolve frame head and the sub-band coding information of current audio frame, obtain the position of all passage subbands in current audio frame and distribute and sample code word;
B. in all passage subbands that participate in current audio frame coding, first select 1 passage subband to carry out position distribution and subtract 1 adjustment, then select one or more passage subbands to carry out position and divide with addition of 1 adjustment; The satisfied condition of selecteed passage subband is: carry out position and distribute and reduce the sample code word bits number discharging after a passage subband distribution in place of adjusting reduces and be more than or equal to and carry out position and distribute to increase after the distribution in place of all passage subbands of adjusting increases and need total bit number of the sample code word expending, and carry out position and distribute and reduce the subband numbering of adjusting and number different with the subband that increases adjustment;
C. to distributing the sample code word of all passage subbands of adjusting to decode and inverse quantization through position, generate the frequency samples of respective channel subband, then according to point coordination after adjusting, these frequency samples are carried out to re-quantization and coding, generate new sample code word;
D. by comprising, the new position of passage subband is distributed and all frame informations of new samples code word generate the audio frame after changing by normal structure.
Visible, the present invention does not need audio decoder to become PCM form, but only in compression domain, the sample of a few passage subband is carried out to simple process, and its computation complexity is very low.
Because the sample code word of the different passage subbands in audio frame is by joint alternative arrangement, after the sample code word of some passage subbands in audio frame changes, the code word changing will be distributed in the whole passage sub-band samples code word region of this audio frame, occupy again the overwhelming majority of audio frame code stream due to this area size, therefore the present invention can change audio file on a large scale, and then ensure the virus of concealing in audio file to destroy.
In addition, zoom factor, zoom factor selection etc. that the present invention does not change in audio frame affect larger information to quantization error, subtract 1 adjustment and only the position of 1 passage subband is distributed, and make tonequality loss less.Meanwhile, the bit of release is distributed to as much as possible other passage subbands by the present invention, the position of one or more passage subbands distributed and added 1 adjustment, makes tonequality loss obtain compensation.
In order further to reduce tonequality loss, in the time selecting to carry out the passage subband of position distribution minimizing adjustment, a preferential passage subband for distribution minimizing 1 rear caused signal noise ratio (Signalto Noise Ratio, SNR) slippage minimum of selecting.This is the inspiration obtaining from the position distribution principle of coding.In mp2 standard code algorithm, it is that iteration completes that position is distributed.Iteration each time, algorithm is distributed to the bit number of 1 unit to have minimum masking noise than the passage subband of (Mask to Noise Ratio, MNR).Masking noise is sheltered signal to noise ratio (S/N ratio) afterwards than expression current demand signal by other signals, and it is larger that noise is afterwards sheltered in the less expression of its value, and tonequality is poorer.Masking noise decides than by the difference of signal noise ratio and signal-to-mask ratio (Signal to Mask Ratio, SMR), that is:
MNR=SNR-SMR
Wherein SNR is determined by quantization bit position, can directly obtain by tabling look-up, and SMR calculates according to acoustic model, after coding, can be considered and immobilizes.Based on this result, the passage subband that this method selects SNR to decline minimum, and its position is distributed and reduced 1, also just make the position of the passage subband that MNR declines minimum distribute and reduce 1, ensure again that from acoustic model angle position distributes the subjective tonequality loss causing for minimum.
Further, in the time selecting to carry out the passage subband of position distribution increase adjustment, a preferential passage subband for the increment maximum of distribution increase by 1 caused signal noise ratio of selecting, makes tonequality loss obtain maximum compensation.
Further, in order to make up to greatest extent tonequality loss, step a also calculates current audio frame unspent bit number in the time of initial coding; In step b, the satisfied condition of selecteed passage subband becomes: carry out position and distribute while reducing the sample code word bits number discharging after the passage subband distribution in place of adjusting reduces with initial coding unspent bit number sum to be more than or equal to carry out position to distribute to increase the total bit number that needs the sample code word expending after the distribution in place of all passage subbands of adjusting increases.
The invention has the beneficial effects as follows, computation complexity is very low, and can change audio file on a large scale, to ensure that the virus to concealing in audio file destroys; Compared with original, the subjective tonequality of file after treatment is lost hardly; Position is distributed after adjustment, cannot recover it, can avoid the method itself by assault; File after treatment kept original as original attributes such as size, port number, sampling rate, bit rates, can be decoded by any standard decoder.
Brief description of the 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 that the data of the 2000th frame are processed the contrast of front and back in the inventive method.
Embodiment
The present invention obtains the allocation information of former each passage subband from encoding stream, according to psychoacoustic model, selector channel subband carries out the small adjustment that position is distributed, and by heavy quantization encoding after the sample codeword decoding inverse quantization of respective sub-bands, finally reorganize composite coding stream writing in files, thereby change original data to reach the object of break virus, as shown in Figure 2, frame by frame audio file is processed, wherein the treatment step of each audio frame is comprised:
1. resolve frame head: from audio frame, resolve the relevant audio format information that obtains according to ISO11172-3 standard;
2. resolve sub-band coding information: from audio frame, resolve information such as obtaining the distribution of each passage subband position and sample code word according to ISO11172-3 standard, wherein a position allocation table for j subband of i passage is shown A (i, j); Calculate unspent bit number adb in the time of initial coding;
3. selector channel subband adjusted position are distributed: according to the numbering sublimit of the highest subband of current participation coding, belong to [1 in subband numbering, sublimit] all passage subbands in select 2 or more passage subband, be recorded in the passage sets of subbands M choosing, and the passage subband of choosing is carried out to position and distribute adjustment, concrete step is as follows:
3.1. to each passage subband, if calculate, the slippage that reduces by 1 caused signal noise ratio is distributed in current position, and be recorded in the key assignments member of a structural array snr_dec.This structural array has also recorded corresponding channel number ch and the subband numbering sub information of key assignments key; This snr_dec array is pressed to key value size, sort from small to large; The item identical to key value, determines its sequencing at random;
3.2. to be assigned as 0 and reached each passage subband maximum allocated except position, if calculate, the increment that increases by 1 caused signal noise ratio is distributed in current position, and be recorded in the key assignments member of another structural array snr_enc.This structural array has also recorded corresponding channel number ch and the subband numbering sub information of key assignments key; This snr_enc array is pressed to key value size, sort from big to small; The item identical to key value, determines its sequencing at random;
3.3. establish m and n and be respectively the subscript of browsing of array snr_dec and snr_enc.Put initial value m=1, n=1, M is empty set;
3.4. get the m item snr_dec[m of snr_dec], this is corresponding to the passage subband of SNR slippage minimum, if its channel number and subband numbering are expressed as snr_dec[m] .ch and snr_dec[m] .sub, distributes its present bit to reduce by 1 rear releasable bit number DEC if calculate; Calculating current available bit number avail_bits is that the current bit number that discharges adds that initial coding is not finished the summation of bit number, i.e. avail_bits=DEC+adb;
3.5. each n of snr_enc is done: get n item snr_enc[n] channel number and subband numbering, be expressed as snr_enc[n] .ch and snr_enc[n] .sub, if calculate, its present bit is distributed and increased the bit number ENC that needs consumption after 1, if and the current available bit number of judgement is more than or equal to the bit number (being avail_bits >=ENC) of needs consumption and discharges not identical with the subband numbering of consumption bit number, selector channel snr_enc[n] .ch, subband snr_enc[n] .sub as will carry out position distribute and increase a passage subband of adjusting, and its band is joined in M, upgrade avail_bits simultaneously, make avail_bit=avail_bits-ENC,
If 3.6 M non-NULLs, selector channel snr_dec[m] .ch, subband snr_dec[m] .sub as will carry out position distribute and reduce a passage subband of adjusting, and its band is joined in M, otherwise puts m=m+1, and forward step 3.4 to and carry out the selection of next round;
In 3.7 couples of M, the position distribution of carrying out a 1 passage subband that distributes minimizing adjustment is subtracted to 1; To dividing with addition of 1 carrying out a position for one or more passage subbands that distribution increase is adjusted in M;
4. decoding inverse quantization sample code word: according to ISO11172-3 standard, the sample code word of all passage subbands in M is decoded and inverse quantization, obtain corresponding frequency samples;
5. weightization coding frequency samples: according to point coordination after adjusting, according to ISO11172-3 standard, the frequency samples of all passage subbands in M is carried out to re-quantization and coding, generate new sample code word;
6. frame encoding stream format: the new position of passage subband is distributed and all frame informations of new samples code word write in encoding stream by comprising according to ISO11172-3 standard.
Embodiment
Original audio taking mp2 audio frequency " Fei Xiang-Cloud Of Hometown .mp2 " file of standard 256kbit/s as input is as example.This audio frequency duration be 4 points 24 seconds, totalframes is 11020 frames.Present embodiment is that unit processes to the audio file of input by coded frame, and single treatment one frame, until handle 11020 whole frames.Taking 2000 frames as example, the specific implementation process of processing is as follows below:
1. resolve frame head: resolving from audio frame according to ISO11172-3 standard and obtaining port number is 2, and sampling rate is 48000kHz, and bit rate is 256kbps, the numbering sublimit of the highest subband that participates in encoding is 27, and decoding table is the information such as the 0th table;
2. resolve sub-band coding information: from audio frame, resolve the information such as position distribution and sample code word that obtains each passage subband according to 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 are distributed the position of two each 27 subbands of passage in corresponding left and right respectively, for example A (2, 13) value is that 2 expressions are when prepass subband (2, 13) position is assigned as 2, calculating the unspent bit number adb of this frame in the time of initial coding is 4 bits,
3. selector channel subband adjusted position are distributed: according to psychoacoustic model, from the subband 1 that comprises left and right passage to 54 candidates of subband 27 selector channel subband, its distribution is adjusted, concrete step is as follows:
3.1. to each passage subband, if calculate, 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, snr_dec key value is sorted from small to large by quicksort method, obtain orderly snr_dec; Wherein Section 1 snr_dec[1] value be 4.00,2,13}, and represent key assignments (or SNR slippage) be 4.00, channel number be 2 and 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 position, if calculate, 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, snr_enc key value is sorted from big to small by quicksort method, obtain orderly snr_enc; Its Section 1 snr_enc[1] value be 4.84,1,11}, and represent key assignments (or SNR increment) be 4.84, channel number be 1 and subband be numbered 11, this is corresponding to the SNR increment of current maximum;
That 3.3. puts array snr_dec browses subscript m=1, and snr_enc browses subscript n=1, and M is empty set;
3.4. investigate the m item of snr_dec array, for the first time, to snr_dec[1] the 13rd subband of the 2nd passage of representative, if calculate, its distribution being reduced to 1 releasable bit number DEC from 2 is 24, calculating avail_bits is 28;
3.5. investigate one by one each n of snr_enc, if its calculating is distributed its present bit to increase the bit number ENC that needs consumption after 1, and Rule of judgment avail_bits >=ENC and snr_enc[n] .sub ≠ snr_dec[m] .sub.After investigation, find to only have the 1st to satisfy condition, its corresponding ENC is 12, channel number is 1, subband is numbered 11, choose the 11st subband of the 1st passage as increasing by carrying out bit the passage subband of adjusting, and passage subband (1,11) is joined in M, avail_bits is updated to 16;
3.6 due to M non-NULL, selects the 13rd subband of the 2nd passage as reducing by carrying out bit the passage subband of adjusting, and passage subband (2,13) band is joined in M;
3.7 make the position distribution of passage subband (2,13) subtract 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 ISO11172-3 standard, the sample code word of passage subband (2,13) and passage subband (1,11) is decoded and inverse quantization, obtain corresponding frequency samples;
5. weightization coding frequency samples: according to updated space assignment information, carry out re-quantization and coding according to the frequency samples of ISO11172-3 standard channel subband (2,13) and passage subband (1,11), generate new sample code word;
6. frame encoding stream format: the new position of passage subband is distributed and all frame informations of new samples code word write in encoding stream by comprising according to ISO11172-3 standard, 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 processing speed link in test tonequality qualifying formula, except method of the present invention, also introducing one is simplified to scheme, remove the 4th and 5 two steps (distributing inverse quantization decoding and the weight coding step of the sample code word of the passage subband of adjusting to carrying out position) of embodiment, ascend the throne after distribution is adjusted and the passage subband changing is not carried out to re-quantization coding, and be directly made into frame by former sub-band samples Codeword Sets, like this change the allocation information that only has passage subband and be not enough to break virus.Adopt in assessment of acoustics link subjective audio frequency quality assessment (the Perceptual Evaluation of Audio Quality describing in standard I TU-R BS.1387, PEAQ) method, it is method the highest with the subjective assessment result degree of correlation in current audio assessment method for encoding quality.The method utilizes people's ear subjective perception property calculation to go out masking threshold and the distortion threshold of signal, then adopt artificial neural network to merge and an evaluating ODG(Object Difference Grade), this enforcement just adopts this parameter to carry out objective evaluation to audio quality.
1. the change of file
First file after treatment and original are carried out to Data Comparison analysis.Known by analysis above, 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 mainly reduces discharge bit number with a distribution and position distributes the difference diff_bit of the spent bit number of increase relevant.Because the code word of each passage sub-band samples is alternative arrangement in encoding stream, once parameter d iff_bit is not 0, will cause the serious dislocation of bit arrangement after processing, and the large-scale variation that brings frame data.
Fig. 3 has shown the Data Comparison situation of the 2000th frame of enumerating before and after processing above, and the background shade that wherein changes byte shows.The bit number that this framing bit distributes minimizing to discharge is 24, and the position spent bit number of distribution increase is 12, i.e. diff_bit=12.Can see there are 2 bytes in the variation of frame previous section, corresponding to the allocation information of reformed two passage subbands.In the passage sub-band samples code word region that accounts for the audio frame overwhelming majority, data have nearly all been changed, and this is the effect of the serious dislocation of bit arrangement.To whole frame, 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 simplify by the inventive method and its file that method processing obtains and decode.Table 1 has been listed 3 decoded tonequality contrasts of audio file, and evaluating is ODG.This parameter value more approaches 0, represents that tonequality is better.
Table 1 original, by the present invention and simplify the tonequality comparison of method file after treatment
| Original | After the present invention processes | After the method that simplifies is processed |
| -3.326 | -3.330 | -3.354 |
Can see, very little by the tonequality difference of the inventive method audio frequency after treatment and former audio frequency, and the method that simplifies is inconsistent with the distribution of new position due to quantization bit figure place corresponding to former sample code word, therefore its tonequality after treatment has decline slightly.In addition, the inventive method does not change the frame head of original audio frame, and has kept frame sign constant, therefore the file after wherethrough reason has kept original attributes such as the size, port number, sampling rate, bit rate of original.
3. processing speed
Simplify method with the inventive method and its testing audio file is become to code processing, table 2 has been listed the result of respective handling 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, dominant frequency is 2.53GHz.For ease of relatively, in table 2, also list the processing time that the corresponding former not compacted voice file of test file is carried out the coding of standard and former mp2 file is carried out to standard decoding.
Table 2 the present invention and its simplify the processing time comparison of method and four kinds of processing of coding and decoding
| Coding | Decoding | Processing of the present invention | Simplify method processing |
| 6.328 | 4.282 | 0.406 | 0.360 |
Can see, the processing speed that simplifies method is the fastest in each processing.With respect to Code And Decode, the processing speed of the inventive method is also very fast, is respectively 15.59 and 10.55 times of Code And Decode, to 4 points of audio frequency of 24 seconds, only just completed 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 encoding stream is carried out the adjustment of quantization bit position, the audio frame after heavy quantization encoding is re-write in encoding stream, audio frame is carried out to bit adjustment and comprise the following steps:
A. resolve frame head and the sub-band coding information of current audio frame, obtain the position of all passage subbands in current audio frame and distribute and sample code word;
B. in all passage subbands that participate in current audio frame coding, first select 1 passage subband to carry out position distribution and subtract 1 adjustment, then select one or more passage subbands to carry out position and divide with addition of 1 adjustment; The satisfied condition of selecteed passage subband is: carry out position and distribute and reduce the sample code word bits number discharging after a passage subband distribution in place of adjusting reduces and be more than or equal to and carry out position and distribute to increase after the distribution in place of all passage subbands of adjusting increases and need total bit number of the sample code word expending, and carry out position and distribute and reduce the subband numbering of adjusting and number different with the subband that increases adjustment;
C. to distributing the sample code word of all passage subbands of adjusting to decode and inverse quantization through position, generate the frequency samples of respective channel subband, then according to point coordination after adjusting, these frequency samples are carried out to re-quantization and coding, generate new sample code word;
D. by comprising, the new position of passage subband is distributed and all frame informations of new samples code word generate the audio frame after changing by normal structure.
2. a kind of method of rapid damage broadcast audio file virus as claimed in claim 1, is characterized in that, distributes while reducing the passage subband of adjusting selecting to carry out position, and a passage subband that reduces by 1 rear caused signal noise ratio slippage minimum is distributed in the preferential position of selecting.
3. a kind of method of rapid damage broadcast audio file virus as claimed in claim 2, it is characterized in that, in the time selecting to carry out the passage subband of position distribution increase adjustment, a preferential passage subband for the increment maximum of distribution increase by 1 caused signal noise ratio of selecting.
4. a kind of method of rapid damage broadcast audio file virus as claimed in claim 1, is characterized in that, step a also calculates current audio frame unspent bit number in the time of initial coding; In step b, the satisfied condition of selecteed passage subband becomes: carry out position and distribute while reducing the sample code word bits number discharging after the passage subband distribution in place of adjusting reduces with initial coding unspent bit number sum to be more than or equal to carry out position to distribute to increase the total bit number that needs the sample code word expending after the distribution in place of all passage subbands of adjusting increases.
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| CN1971734A (en) * | 2006-10-20 | 2007-05-30 | 宁波大学 | Method for embedding and extracting authentication information of numeric music works |
| CN101394402A (en) * | 2008-10-13 | 2009-03-25 | 邓学锋 | Method for fast code changing in large range to audio information to break virus |
| CN101840701A (en) * | 2009-03-18 | 2010-09-22 | 数维科技（北京）有限公司 | Hierarchical audio coding frame structure |
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| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1971734A (en) * | 2006-10-20 | 2007-05-30 | 宁波大学 | Method for embedding and extracting authentication information of numeric music works |
| CN101394402A (en) * | 2008-10-13 | 2009-03-25 | 邓学锋 | Method for fast code changing in large range to audio information to break virus |
| CN101840701A (en) * | 2009-03-18 | 2010-09-22 | 数维科技（北京）有限公司 | Hierarchical audio coding frame structure |
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