Robust Speech Processing in EW Environment (original) (raw)
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Continuous efforts are active to reduce the bit rates but maintaining channel noise tolerance, secure transmission and justified MOS(mean opinion score) among various communication networks. These networks at their endterminals may employ variety of vocoders operating at different individual bit rates. In order to maintain fidelity, transcoders are used to map the information when traffic flows from one channel operating at one bit rate to another channel operating on another bit rate as seen in the case of channels with different capacities. Some networks (like satellite communication and some private networks) employs codecs like AMBE (Advanced Multiband Excitation), CVSD for their very low bit rate, channel noise tolerant attributes and another features. In order to interface networks accompanying the said vocoders with that of public networks containing codec like PCM, ADPCM, we have done feasibility study for justified MOS using AMBE-3000 HDK. Also, we have compared Transcoders against MOS.
Speech coding and identification techniques for net centric communication
Speech has been considered to be a most important component for human to convey information to one another. Today, speech communications are available for a number of applications. In applications like military, net centric communications are accessed by large number of users so it has to be given more protection and security against attacks. This paper deals with the most promising research in the integrating of speech coding, speaker authentication and strong encryption. The special features such as encryption provide more confidentiality and authentication to guard against masquerade and error tolerance to operate under acoustic environment in battlefield. This paper gives an overview of the concepts and techniques available in speech coding, speaker Identification, Encryption and Decryption.
2002
(U) Currently, low echelon military commanders have no ability to directly access the remote Situation Awareness (SA) and Command and Control (C 2) information contained in Battlefield Tactical Operations Center (TOC) databases. Previous attempts, using Speech Recognition(SR) over a Narrowband Communications channel using a coded speech signal have been unsuccessful. However, our research and testing has shown that, due to recent advances in the areas of narrowband Digital Speech Processing and SR, it is now possible to perform robust recognition of a useful vocabulary over narrowband tactical communications channels. (U) This paper describes an implemented and tested system which was designed to provide secure, bandwidth efficient, hands-free, automatic remote database access to commanders in the field. This system uses a COTSbased Digital Speech Coder with an advanced Noise Preprocessor to communicate over a narrowband IPbased tactical network into a database server. The server employs COTS-based Speech Recognition(SR) system components and a SQL-generator program to provide direct access into a TOC-type database. Queries made by the user are transmitted to the database over a coded Narrowband Communications Channel and the answers are spoken back to the user.
Secure speech communication has been of great importance in civil, commercial and military communication systems. As speech communication becomes widely used and even more vulnerable, the importance of providing a high level of security becomes a major issue. The main objective of this paper is to increase the security, and to remove the redundancy for speech communication system under the global context of secure communication. So it deals with the integrating of speech coding, with speaker authentication and strong encryption. This paper also gives an overview and techniques available in speech coding, speaker Identification, Encryption and Decryption. The primary objective of this paper is to summarize some of the well known methods used in various stages for secure speech communication system.
Resilient Voice Communication for the MITRE Corporation (M.Eng. Report)
Voice communication is a vital tool for pilots flying combat missions. Tactical communication systems face extreme interference conditions that are uncommon in the commercial sector, including jamming from enemy-deployed electronic warfare systems. In our work, we investigate two strategies to increase the resilience of US Department of Defense (DoD) airborne voice communication systems, namely (1) improving the forward error correction (FEC) capabilities of existing systems through the application of Low Density Parity Check (LDPC) codes and, at the source side, (2) replacing DoD standardized voice coders with speech-to-text/text-to-speech (STT/TTS) technology to provide a reduction in source rate relative to the original coder. Given equal channel rates, a system using the text based approach as opposed to a traditional vocoder offers opportunities for improved forward error correction due to excess bitrate. We examine the operation of such a system under various FEC schemes including LDPC and repetition coding. We also consider a hybrid system that combines approaches (1) and (2).
This document describes the RTP payload format for the Tactical Secure Voice Cryptographic Interoperability Specification (TSVCIS) speech coder. TSVCIS is a scalable narrowband voice coder supporting varying encoder data rates and fallbacks. It is implemented as an augmentation to the Mixed Excitation Linear Prediction Enhanced (MELPe) speech coder by conveying additional speech coder parameters to enhance voice quality. TSVCIS augmented speech data is processed in conjunction with its temporally matched Mixed Excitation Linear Prediction (MELP) 2400 speech data. The RTP packetization of TSVCIS and MELPe speech coder data is described in detail.
A robust secure speech communication system using ITU-T G.723.1 and TMS320C6711 DSP
Microprocessors and Microsystems, 2006
Secure speech communication plays a dominant role in civil and military communication systems. Speech scrambling and descrambling, particularly in time-domain, has been a key component in these systems. Removal of redundant information is very significance in secure speech communication systems. In addition to security, fast implementation of security algorithms is also very important for real-time applications. Thus after the penetration of emerging technologies like DSP (Digital Signal Processors) this importance has increased even more. Traditional schemes and methods used for speech scrambling either do not remove the redundancy of the signal or pay less attention to this parameter, which thereby provides an opportunity to an interceptor to descramble the signal with convenience. So far most of the implementations of scrambling techniques based on DSP have been carried out either on 16-bit fixed-point DSPs or on multiprocessors, which offer low accuracy and comparatively slower speed and thus are not suitable for real-time applications. In this research we have focused on these three parameters simultaneously and proposed a novel redundancies-free, high speed, real-time secure speech communication system. This system is based on time-domain speech scrambling/descrambling, ITU-T dual rate speech codec G.723.1 and Texas Instrument's 32-bit floating point DSP TMS320 C6711. Real-time speech signal is captured through DSP port and redundancy is removed by compressing the original signal to 5.3 or 6.3 kb/s. Subsequently, compressed signal is scrambled using hopping window and sliding window techniques using linear congruent pseudo-random generator. The results show that our system is redundancies-free, fast, secure and more suitable for real-time applications.
Transform-domain and DSP based secure speech communication system
Microprocessors and Microsystems, 2007
Secure speech communication has been of great significance in civil, commercial and particularly in military communication systems. Speech scramblers play a major role in these systems and transform-domain, specially, DCT (Discrete Cosine Transform) based encryption of speech has often been a preferable choice for researchers and engineers. In such systems, besides security, removal of redundant information and the execution speed of the algorithms have also been the main focus of research and exploration. In this paper, we concentrate on the investigation of these three parameters in general and security and speed in particular. In this research these attributes are envisaged by exploiting ITU-T G.723.1 speech codec, DCT based speech encryption algorithm and Texas Instruments' 32-bit floatingpoint DSP (Digital Signal Processor) TMS320C6711, respectively. This work, in fact is a continuation and an advancement of the work carried out by Jameel et al. [A. Jameel, M.Y. Siyal, N. Ikram, A Robust Secure Speech Communication System using G.723.1 and TMS320C6711 DSP, Microprocessors and Microsystems 30 (1) (2006) 26-32.] in which they presented a redundancies-free, secure and fast speech communication system using G.723.1 codec, time-domain speech scrambling techniques and Texas Instruments TMS320C6711 DSP. On the contrary, in this exploration the same system is improved significantly in terms of security and speed. In order to reduce residual intelligibility thereby to enhance security a new DCT based algorithm using sub-framing and lookup table is proposed, implemented and analyzed. In terms of speed, the reference code of G.723.1 is further optimized on advanced level. Additionally, the schemed is designed in such a way that it also diminishes talk spurt and intonation present in the original speech signal which, as a result, increases the level of security. Hence a redundancies-free, secure and high speed communication system has been proposed, implemented and analyzed using various types of speech segments. The results show that the proposed system is suitable for real-time civil, commercial and military applications.
Adaptive Noise Cancellation, Compression And Encryption Of Speech Signals
An Adaptive noise cancellation system is preferred in applications where there is no prior knowledge of the signal to be filtered or the noise which corrupts it. Such a case occurs in many military applications. Also noise cancellation is very much required in day-to-day applications such as mobile and other wireless communication systems. Another very important issue in communication systems, especially military, is the security of the data. If the data that is being send is not encrypted in some way then, the enemy or anybody who intercepts the data can access it. Finally compressing the data will allow us to communicate much more information, with the transmission of much less data. In this project we aim to design and implement using suitable hardware, a communication system that incorporates all of the above requirements .That is, a noisy speech signal is filtered using an adaptive noise cancellation system, which is then compressed using predictive line coding techniques. The compressed data is the encrypted using a suitable private key algorithm and is transmitted in real time. Private key rather than public key is used to encrypt the speech because public key encryption is computationally intensive and transmitting information in real-time will be almost impossible. Currently the encryption of data is not used in mobile communication systems, but it will surely be implemented in the near future, as security issues are becoming a major concern. One of the main challenges is to optimise the processor usage to meet the real time requirements of the system.