本文關鍵詞： 語音加解密 FPGA Xilinx ISE 算法　出處：《電子科技大學》2014年碩士論文　論文類型：學位論文

【摘要】：GSM系統(tǒng)是一個成熟的網(wǎng)絡,要在該網(wǎng)絡上實現(xiàn)話音的安全傳輸,必須遵循該網(wǎng)絡的規(guī)范。既不能改變現(xiàn)有GSM系統(tǒng),又要實現(xiàn)較高的安全保密性。端到端加密就是在語音發(fā)送之前進行加密,在接收方收到語音信號之后進行解密。當前,GSM網(wǎng)絡中語音的端到端加密的實現(xiàn)途徑主要有兩種。第一種是通過網(wǎng)絡的數(shù)據(jù)通道,這種方法靈活、可采用的實現(xiàn)技術較多,但是,其也存在一定的不足和局限。第二種方法是通過傳統(tǒng)的語音通道,跟前一種方法相比,其具有延遲小、與現(xiàn)有的各種增值業(yè)務適應性強等優(yōu)點,更便于推廣使用。但由于跟CDMA系統(tǒng)采用CELP編碼方式不同,GSM系統(tǒng)的語音通道中采用的是RPE-LTP(Regular Pulse Excitation-Long Term Prediction,規(guī)則脈沖激勵長期線性預測編碼),目前,針對RPE-LTP的語音加解密算法尚不多見。為此,本文設計一種新的語音加密算法,在詳細分析RPE-LTP聲碼器工作原理的基礎上,提出了一種可通過GSM傳統(tǒng)語音通道上進行傳輸語音加密算法——抗RPE-LTP編碼算法。系統(tǒng)研究了語音平滑、濾波技術,采用FPGA芯片和VHDL硬件描述語言,實現(xiàn)一種新的抗RPE-LTP壓縮編碼語音加解密算法,完成幀置亂加密、加同步頭、去同步頭、幀置亂解密等模塊的設計,實現(xiàn)語音的加解密過程。同時,對結(jié)果進行仿真分析。除驗證各模塊功能外,還對整個系統(tǒng)的加、解密過程進行仿真驗證和性能分析。可以看出,本算法兼具語音信號處理、分組密碼加密運算的優(yōu)勢,平衡性較好。并且,該算法在語音信號進入聲碼器之前運行,不會對語音信號的語音特性產(chǎn)生破壞,可以在接收端很好的還原出原始信號,同時又不用對基站設備進行任何改造,成本低、效率高,是一種針對GSM網(wǎng)絡語音端到端加密的方法。該設計使用了Xilinx公司提供的ISE14.2集成開發(fā)環(huán)境,進一步優(yōu)化語音質(zhì)量,減少噪聲。
[Abstract]:GSM system is a mature network. In order to realize the secure transmission of voice on the network, we must follow the standard of the network. We can not change the existing GSM system. End-to-end encryption is to encrypt the voice before it is sent. At present, there are two ways to realize end-to-end encryption of voice in GSM network. The first is through the data channel of the network. This method is flexible and can be implemented with more techniques. The second method is through the traditional voice channel. Compared with the previous method, it has the advantages of small delay, strong adaptability to various existing value-added services, and so on. But because RPE-LTP(Regular Pulse Excitation-Long Term prediction is used in speech channel different from CELP coding in CDMA system, regular pulse excitation long term linear predictive coding is used. There are few speech encryption and decryption algorithms for RPE-LTP. In this paper, a new speech encryption algorithm is designed, based on the detailed analysis of the principle of RPE-LTP vocoder. This paper presents a speech encryption algorithm, anti-#en1# coding algorithm, which can be transmitted over the traditional speech channel of GSM. The speech smoothing and filtering techniques are studied systematically, and the FPGA chip and VHDL hardware description language are used. To implement a new anti-#en0# compression coding speech encryption and decryption algorithm, complete frame scrambling encryption, add synchronization head, desynchronize head, frame scrambling decryption and other modules, realize the encryption and decryption process of speech, at the same time, In addition to verifying the functions of each module, the whole system encryption and decryption process are also simulated and analyzed. It can be seen that this algorithm has the advantages of speech signal processing and block cipher encryption. Moreover, the algorithm can not destroy the speech characteristics of the speech signal before it enters the vocoder, and can restore the original signal very well in the receiver. At the same time, without any modification of base station equipment, it has low cost and high efficiency. It is a method of end-to-end encryption for voice in GSM network. This design uses ISE14.2 integrated development environment provided by Xilinx Company to further optimize voice quality. Reduce noise.
【學位授予單位】：電子科技大學
【學位級別】：碩士
【學位授予年份】：2014
【分類號】：TN929.532;TN918.4

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本文編號：1542174