Steganography integration into a low-bit rate speech codec

Low bit-rate speech codecs have been widely used in audio communications like VoIP and mobile communications, so that steganography in low bit-rate audio streams would have broad applications in practice. In this paper, the authors propose a new algorithm for steganography in low bit-rate VoIP audio streams by integrating information hiding into the process of speech encoding. The proposed algorithm performs data embedding while pitch period prediction is conducted during low bit-rate speech encoding, thus maintaining synchronization between information hiding and speech encoding. The steganography algorithm can achieve high quality of speech and prevent detection of steganalysis, but also has great compatibility with a standard low bit-rate speech codec without causing further delay by data embedding and extraction. Testing shows, with the proposed algorithm, the data embedding rate of the secret message can attain 4 bits / frame (133.3 bits / second)

Semifragile Speech Watermarking Based on Least Significant Bit Replacement of Line Spectral Frequencies

There are various techniques for speech watermarking based on modifying the linear prediction coefficients (LPCs); however, the estimated and modified LPCs vary from each other even without attacks. Because line spectral frequency (LSF) has less sensitivity to watermarking than LPC, watermark bits are embedded into the maximum number of LSFs by applying the least significant bit replacement (LSBR) method. To reduce the differences between estimated and modified LPCs, a checking loop is added to minimize the watermark extraction error. Experimental results show that the proposed semifragile speech watermarking method can provide high imperceptibility and that any manipulation of the watermark signal destroys the watermark bits since manipulation changes it to a random stream of bits

Media gateway utilizando um GPU

Mestrado em Engenharia de Computadores e Telemátic

A configurable vector processor for accelerating speech coding algorithms

The growing demand for voice-over-packer (VoIP) services and multimedia-rich applications has made increasingly important the efficient, real-time implementation of low-bit rates speech coders on embedded VLSI platforms. Such speech coders are designed to substantially reduce the bandwidth requirements thus enabling dense multichannel gateways in small form factor. This however comes at a high computational cost which mandates the use of very high performance embedded processors. This thesis investigates the potential acceleration of two major ITU-T speech coding algorithms, namely G.729A and G.723.1, through their efficient implementation on a configurable extensible vector embedded CPU architecture. New scalar and vector ISAs were introduced which resulted in up to 80% reduction in the dynamic instruction count of both workloads. These instructions were subsequently encapsulated into a parametric, hybrid SISD (scalar processor)–SIMD (vector) processor. This work presents the research and implementation of the vector datapath of this vector coprocessor which is tightly-coupled to a Sparc-V8 compliant CPU, the optimization and simulation methodologies employed and the use of Electronic System Level (ESL) techniques to rapidly design SIMD datapaths

New Cryptographic Algorithms for Enhancing Security of Voice Data

A real-time application Voice over Internet Protocol (VoIP) is the technology that enables voice packets transmission over internet protocol (IP). Security is of concern whenever open networks are to be used. In general, the real-time applications suffer from packet latency and loss due to the nature of IP network. Cryptographic systems may be used to achieve VoIP security, but their impact on the Quality of Services (QoS) should be minimized. Most of the known encryption algorithms are computationally expensive resulting in a significant amount of time added to packet delay. VoIP is usually used by public users resulting in a key exchange problem and a trusted intermediate authority normally takes this responsibility. In this research, VoIP security was enhanced via a proposed cryptographic system. The proposed solution consists of a simple, but strong encryption/decryption algorithm as well as an embedded method to exchange the keys between the users. In this research, a new keys is generated in a random fashion and then used to encrypt each new voice packet to strengthen the security level. Key exchange is carried out by inserting the key with the ciphered voice packet that depends on the table of the key positions at the sender and receiver sides, and the target receiver is the only one who is able to extract the key. The encryption process in this research is divided into three main stages: key generation, encryption process, and key insertion process. The decryption process on the other hand is divided into two main stages: key extraction process, and decryption process. The proposed solution was implemented and tested and the results showed that the required time for the security processes is minimized compared to some known algorithms such as AES_Rijndael algorithm. Furthermore, the analysis has proved that the security level has a direct relationship to the key length and the voice packet size in that large packet size requires more processing time. Finally, the implementation result in this research shows the average time needed to encrypt and decrypt a voice packet size using a proposed algorithm with the long key of 1024-bits is much smaller than AES_Rijndael algorithm with a short key length of 128-bits

Communication Platform for Evaluation of Transmitted Speech Quality, Journal of Telecommunications and Information Technology, 2011, nr 3

A voice communication system designed and implemented is described. The purpose of the presented platform was to enable a series of experiments related to the quality assessment of algorithms used in the coding and transmitting of speech. The system is equipped with tools for recordingsignals at each stage of processing, making it possible to subject them to subjective assessments by listening tests or, objective evaluation employing PESQ or PSQM algorithms. The functionality for the simulation of distortions typical for voice communication over the Internet was implemented, making itpossible to obtain reproducible, quantifiable results. An application of the presented platform for evaluation of acoustic echo canceler algorithm based on watermarking techniques, which was developed earlier is presented as an example of an effective deployment of the described technology

A New covert channel over RTP

In this thesis, we designed and implemented a new covert channel over the RTP protocol. The covert channel modifies the timestamp value in the RTP header to send its secret messages. The high frequency of RTP packets allows for a high bitrate covert channel, theoretically up to 350 bps. The broad use of RTP for multimedia applications, including VoIP, provides plentiful opportunities to use this channel. By using the RTP header, many of the challenges present for covert channels using the RTP payload are avoided. Using the reference implementation of this covert channel, bitrates of up to 325 bps were observed. Speed decreases on less reliable networks, though message delivery was flawless with up to 1% RTP packet loss. The channel is very difficult to detect due to expected variations in the timestamp field and the flexible nature of RTP