Spread Spectrum Based High Embedding Capacity Watermarking Method for Audio Signals

Audio watermarking is a promising technology for copyright protection of audio data. Built upon the concept of spread spectrum (SS), many SS-based audio watermarking methods have been developed, where a pseudonoise (PN) sequence is usually used to introduce security. A major drawback of the existing SS-based audio watermarking methods is their low embedding capacity. In this paper, we propose a new SS-based audio watermarking method which possesses much higher embedding capacity while ensuring satisfactory imperceptibility and robustness. The high embedding capacity is achieved through a set of mechanisms: embedding multiple watermark bits in one audio segment, reducing host signal interference on watermark extraction, and adaptively adjusting PN sequence amplitude in watermark embedding based on the property of audio segments. The effectiveness of the proposed audio watermarking method is demonstrated by simulation examples

Parameterization of LSB in Self-Recovery Speech Watermarking Framework in Big Data Mining

The privacy is a major concern in big data mining approach. In this paper, we propose a novel self-recovery speech watermarking framework with consideration of trustable communication in big data mining. In the framework, the watermark is the compressed version of the original speech. The watermark is embedded into the least significant bit (LSB) layers. At the receiver end, the watermark is used to detect the tampered area and recover the tampered speech. To fit the complexity of the scenes in big data infrastructures, the LSB is treated as a parameter. This work discusses the relationship between LSB and other parameters in terms of explicit mathematical formulations. Once the LSB layer has been chosen, the best choices of other parameters are then deduced using the exclusive method. Additionally, we observed that six LSB layers are the limit for watermark embedding when the total bit layers equaled sixteen. Experimental results indicated that when the LSB layers changed from six to three, the imperceptibility of watermark increased, while the quality of the recovered signal decreased accordingly. This result was a trade-off and different LSB layers should be chosen according to different application conditions in big data infrastructures

반향 환경에 강인한 음향 데이터 전송을 위한 오디오 정보 은닉 기법 연구

학위논문 (박사)-- 서울대학교 대학원 : 전기·컴퓨터공학부, 2014. 2. 김남수.In this dissertation, audio data hiding methods suitable for acoustic data transmission are studied. Acoustic data transmission implies a technique which communicates data in short-range aerial space between a loudspeaker and a microphone. Audio data hiding method implies a technique that embeds message signals into audio such as music or speech. The audio signal with embedded message is played back by the loudspeaker at a transmitter and the signal is recorded by the microphone at a receiver without any additional communication devices. The data hiding methods for acoustic data transmission require a high level of robustness and data rate than those for other applications. For one of the conventional methods, the acoustic orthogonal frequency division multiplexing (AOFDM) technique was developed as a reliable communication with reasonable bit rate. The conventional methods including AOFDM, however, are considered deficient in transmission performance or audio quality. To overcome this limitation, the modulated complex lapped transform (MCLT) is introduced in the second chapter of the dissertation. The system using MCLT does not produce blocking artifacts which may degrade the quality of the resulting data-embedded audio signal. Moreover, the interference among adjacent coefficients due to the overlap property is analyzed to take advantage of it for data embedding and extraction. In the third chapter of the dissertation, a novel audio data hiding method for the acoustic data transmission using MCLT is proposed. In the proposed system, audio signal is transformed by the MCLT and the phases of the coefficients are modified to embed message based on the fact that human auditory perception is more sensitive to the variation in magnitude spectra. In the proposed method, the perceived quality of the data-embedded audio signal can be kept almost similar to that of the original audio while transmitting data at several hundreds of bits per second (bps). The experimental results have shown that the audio quality and transmission performance of proposed system are better than those of the AOFDM based system. Moreover, several techniques have been found to further improve the performance of the proposed acoustic data transmission system which are listed as follows: incorporating a masking threshold (MM), clustering based decoding (CLS), and a spectral magnitude adjustment (SMA). In the fourth chapter of the dissertation, an audio data hiding technique more suitable for acoustic data transmission in reverberant environments is proposed. In this approach, sophisticated techniques widely deployed in wireless communication is incorporated which can be summarized as follows: First, a proper range of MCLT length to cope with reverberant environments is analyzed based on the wireless communication theory. Second, a channel estimation technique based on the Wiener estimator to compensate the effect of channel is applied in conjunction with a suitable data packet structure. From the experimental result, the MCLT length longer than the reverberation time is found to be robust against the reverberant environments at the cost of the quality of the data-embedded audio. The experimental results have also shown that the proposed method is robust against various forms of attacks such as signal processing, overwriting, and malicious removal methods. However, it would be the most severe problem to find a proper window length which satisfies both the inaudible distortion and robust data transmission in the reverberant environments. For the phase modification of the audio signal, it would be highly likely to incur a significant quality degradation if the length of time-frequency transform is very long due to the pre-echo phenomena. In the fifth chapter, therefore, segmental SNR adjustment (SSA) technique is proposed to further modify the spectral components for attenuating the pre-echo. In the proposed SSA technique, segmenatal SNR is calculated from short-length MCLT analysis and its minimum value is limited to a desired value. The experimental results have shown that the SSA algorithm with a long MCLT length can attenuate the pre-echo effectively such that it can transmit data more reliably while preserving good audio quality. In addition, a good trade-off between the audio quality and transmission performance can be achieved by adjusting only a single parameter in the SSA algorithm. If the number of microphones is more than one, the diversity technique which takes advantage of transmitting duplicates through statistically independent channel could be useful to enhance the transmission reliability. In the sixth chapter, the acoustic data transmission technique is extended to take advantage of the multi-microphone scheme based on combining. In the combining-based multichannel method, the synchronization and channel estimation are respectively performed at each received signal and then the received signals are linearly combined so that the SNR is increased. The most noticeable property for combining-based technique is to provide compatibility with the acoustic data transmission system using a single microphone. From the series of the experiments, the proposed multichannel method have been found to be useful to enhance the transmission performance despite of the statistical dependency between the channels.Abstract i List of Figures ix List of Tables xv Chapter 1 Introduction 1 1.1 Audio Data Hiding and Acoustic Data Transmission 1 1.2 Previous Methods 4 1.2.1 Audio Watermarking Based Methods 4 1.2.2 Wireless Communication Based Methods 6 1.3 Performance Evaluation 9 1.3.1 Audio Quality 9 1.3.2 Data Transmission Performance 10 1.4 Outline of the Dissertation 10 Chapter 2 Modulated Complex Lapped Transform 13 2.1 Introduction 13 2.2 MCLT 14 2.3 Fast Computation Algorithm 18 2.4 Derivation of Interference Terms in MCLT 19 2.5 Summary 24 Chapter 3 Acoustic Data Transmission Based on MCLT 25 3.1 Introduction 25 3.2 Data Embedding 27 3.2.1 Message Frame 27 3.2.2 Synchronization Frame 29 3.2.3 Data Packet Structure 32 3.3 Data Extraction 32 3.4 Techniques for Performance Enhancement 33 3.4.1 Magnitude Modification Based on Frequency Masking 33 3.4.2 Clustering-based Decoding 35 3.4.3 Spectral Magnitude Adjustment Algorithm 37 3.5 Experimental Results 39 3.5.1 Comparison with Acoustic OFDM 39 3.5.2 Performance Improvements by Magnitude Modification and Clustering based Decoding 47 3.5.3 Performance Improvements by Spectral Magnitude Adjustment 50 3.6 Summary 52 Chapter 4 Robust Acoustic Data Transmission against Reverberant Environments 55 4.1 Introduction 55 4.2 Data Embedding 56 4.2.1 Data Embedding 57 4.2.2 MCLT Length 58 4.2.3 Data Packet Structure 60 4.3 Data Extraction 61 4.3.1 Synchronization 61 4.3.2 Channel Estimation and Compensation 62 4.3.3 Data Decoding 65 4.4 Experimental Results 66 4.4.1 Robustness to Reverberation 69 4.4.2 Audio Quality 71 4.4.3 Robustness to Doppler Effect 71 4.4.4 Robustness to Attacks 71 4.5 Summary 75 Chapter 5 Segmental SNR Adjustment for Audio Quality Enhancement 77 5.1 Introduction 77 5.2 Segmental SNR Adjustment Algorithm 79 5.3 Experimental Results 83 5.3.1 System Configurations 83 5.3.2 Audio Quality Test 84 5.3.3 Robustness to Attacks 86 5.3.4 Transmission Performance of Recorded Signals in Indoor Environment 87 5.3.5 Error correction using convolutional coding 89 5.4 Summary 91 Chapter 6 Multichannel Acoustic Data Transmission 93 6.1 Introduction 93 6.2 Multichannel Techniques for Robust Data Transmission 94 6.2.1 Diversity Techniques for Multichannel System 94 6.2.2 Combining-based Multichannel Acoustic Data Transmission 98 6.3 Experimental Results 100 6.3.1 Room Environments 101 6.3.2 Transmission Performance of Simulated Environments 102 6.3.3 Transmission Performance of Recorded Signals in Reverberant Environment 105 6.4 Summary 106 Chapter 7 Conclusions 109 Bibliography 113 국문초록 121Docto

Robust digital image watermarking

This research presents a novel rank based image watermarking method and improved moment based and histogram based image watermarking methods. A high-frequency component modification step is also proposed to compensate the side effect of commonly used Gaussian pre-filtering. The proposed methods outperform the latest image watermarking methods

Digital watermarking methods for data security and authentication

Philosophiae Doctor - PhDCryptology is the study of systems that typically originate from a consideration of the ideal circumstances under which secure information exchange is to take place. It involves the study of cryptographic and other processes that might be introduced for breaking the output of such systems - cryptanalysis. This includes the introduction of formal mathematical methods for the design of a cryptosystem and for estimating its theoretical level of securit

Proposta e avaliação de sistema de comunicação de informação escondida em propagandas sonoras

Dissertação (mestrado) - Universidade Federal de Santa Catarina, Centro Tecnológico, Programa de Pós-Graduação em Engenharia Elétrica, Florianópolis, 2014.Esta dissertação propõe um novo sistema de comunicação informado projetado para incorporar a informação escondida em anúncios de áudio. A informação é modulada e transmitida através de uma marca d'água em áudio digital incorporada ao sinal de áudio hospedeiro. A inserção de marca d'água é projetada para alcançar um impacto perceptual mínimo e alta robustez às distorções do ambiente, a fim de cumprir com a proposta de aplicação em marketing. A energia da incorporação está devidamente dimensionada para atingir a alta transparência. Novas técnicas de sincronização e de equalização adaptativa são investigadas e propostas para extrair as informações de forma confiável após as interferências do canal ambiente. Os resultados experimentais, utilizando canais ambientes simulados e reais, ilustram o desempenho do sistema proposto e os resultados estão em contraste a outras técnicas existentes. O objetivo deste trabalho é expor um sistema de marca d'água digital de áudio que integre em um sistema de comunicação informada escondida em propagandas sonoras, permitindo a utilização deste sistema em aplicações de marketing.This dissertation proposes a new informed communication system designed to embed information hidden into audio advertisements. The information is modulated and transmitted through a audio digital watermark embedded into the audio signal host. The watermark embedding is designed to achieve minimal perceptual impact and high robustness to environment distortions in order to comply with the proposed application in marketing. The embedding energy is properly scaled in order to achieve high transparency. Novel synchronization and adaptive equalization techniques are investigated and proposed to reliably extract the information after the channel interferences. Experimental results using simulated and real environment channels illustrate the performance of the proposed system and results are contrasted to other existing techniques

Acoustic-channel attack and defence methods for personal voice assistants

Personal Voice Assistants (PVAs) are increasingly used as interface to digital environments. Voice commands are used to interact with phones, smart homes or cars. In the US alone the number of smart speakers such as Amazon’s Echo and Google Home has grown by 78% to 118.5 million and 21% of the US population own at least one device. Given the increasing dependency of society on PVAs, security and privacy of these has become a major concern of users, manufacturers and policy makers. Consequently, a steep increase in research efforts addressing security and privacy of PVAs can be observed in recent years. While some security and privacy research applicable to the PVA domain predates their recent increase in popularity and many new research strands have emerged, there lacks research dedicated to PVA security and privacy. The most important interaction interface between users and a PVA is the acoustic channel and acoustic channel related security and privacy studies are desirable and required. The aim of the work presented in this thesis is to enhance the cognition of security and privacy issues of PVA usage related to the acoustic channel, to propose principles and solutions to key usage scenarios to mitigate potential security threats, and to present a novel type of dangerous attack which can be launched only by using a PVA alone. The five core contributions of this thesis are: (i) a taxonomy is built for the research domain of PVA security and privacy issues related to acoustic channel. An extensive research overview on the state of the art is provided, describing a comprehensive research map for PVA security and privacy. It is also shown in this taxonomy where the contributions of this thesis lie; (ii) Work has emerged aiming to generate adversarial audio inputs which sound harmless to humans but can trick a PVA to recognise harmful commands. The majority of work has been focused on the attack side, but there rarely exists work on how to defend against this type of attack. A defence method against white-box adversarial commands is proposed and implemented as a prototype. It is shown that a defence Automatic Speech Recognition (ASR) can work in parallel with the PVA’s main one, and adversarial audio input is detected if the difference in the speech decoding results between both ASR surpasses a threshold. It is demonstrated that an ASR that differs in architecture and/or training data from the the PVA’s main ASR is usable as protection ASR; (iii) PVAs continuously monitor conversations which may be transported to a cloud back end where they are stored, processed and maybe even passed on to other service providers. A user has limited control over this process when a PVA is triggered without user’s intent or a PVA belongs to others. A user is unable to control the recording behaviour of surrounding PVAs, unable to signal privacy requirements and unable to track conversation recordings. An acoustic tagging solution is proposed aiming to embed additional information into acoustic signals processed by PVAs. A user employs a tagging device which emits an acoustic signal when PVA activity is assumed. Any active PVA will embed this tag into their recorded audio stream. The tag may signal a cooperating PVA or back-end system that a user has not given a recording consent. The tag may also be used to trace when and where a recording was taken if necessary. A prototype tagging device based on PocketSphinx is implemented. Using Google Home Mini as the PVA, it is demonstrated that the device can tag conversations and the tagging signal can be retrieved from conversations stored in the Google back-end system; (iv) Acoustic tagging provides users the capability to signal their permission to the back-end PVA service, and another solution inspired by Denial of Service (DoS) is proposed as well for protecting user privacy. Although PVAs are very helpful, they are also continuously monitoring conversations. When a PVA detects a wake word, the immediately following conversation is recorded and transported to a cloud system for further analysis. An active protection mechanism is proposed: reactive jamming. A Protection Jamming Device (PJD) is employed to observe conversations. Upon detection of a PVA wake word the PJD emits an acoustic jamming signal. The PJD must detect the wake word faster than the PVA such that the jamming signal still prevents wake word detection by the PVA. An evaluation of the effectiveness of different jamming signals and overlap between wake words and the jamming signals is carried out. 100% jamming success can be achieved with an overlap of at least 60% with a negligible false positive rate; (v) Acoustic components (speakers and microphones) on a PVA can potentially be re-purposed to achieve acoustic sensing. This has great security and privacy implication due to the key role of PVAs in digital environments. The first active acoustic side-channel attack is proposed. Speakers are used to emit human inaudible acoustic signals and the echo is recorded via microphones, turning the acoustic system of a smartphone into a sonar system. The echo signal can be used to profile user interaction with the device. For example, a victim’s finger movement can be monitored to steal Android unlock patterns. The number of candidate unlock patterns that an attacker must try to authenticate herself to a Samsung S4 phone can be reduced by up to 70% using this novel unnoticeable acoustic side-channel

Tatouage pour le renforcement de la qualité audio des systèmes de communication bas débit

L'objectif de cette thèse est d'étudier l'idée du tatouage dans le traitement du son.Les recherches en tatouage audio se sont principalement tournées vers des applications sécuritaires ou de transmission de données auxiliaires. Une des applications visées par ce concept consiste à améliorer la qualité du signal hôte ayant subi des transformations et ceci en exploitant l'information qu'il véhicule. Le tatouage audio est donc considéré comme mémoire porteuse d'informations sur le signal originel. La compression à bas débit des signaux audio est une des applications visée par ce concept. Dans ce cadre, deux objectifs sont proposés : la réduction du pré-écho et de l'amollissement d'attaque, deux phénomènes introduits par les codeurs audio perceptifs, en particulier les codeurs AAC et MP3; la préservation de l'harmonicité des signaux audio dégradée par les codeurs perceptifs à extension de bande, en particulier le codeur HE-AAC.La première partie de ce manuscrit présente les principes de base des systèmes de codage bas débit et étudie les différentes distorsions introduites par ces derniers. Fondées sur cette étude, deux solutions sont proposées. La première, visant principalement la réduction du pré-écho, consiste à corriger l'enveloppe temporelle du signal après réception en exploitant la connaissance a priori de l'enveloppe temporelle du signal original, supposée transmise par un canal auxiliaire à faible débit (< 500 bits/s). La seconde solution vise à corriger les ruptures d'harmonicité générées par les codeurs à extension de bande. Ce phénomène touche essentiellement les signaux fortement harmoniques (exemple : violon) et est perçu comme une dissonance. Une préservation de l'harmonicité des signaux audio par des opérations de translation spectrale est alors proposée, les paramètres étant là encore transmis par un canal auxiliaire à faible débit.La seconde partie de ce document est consacrée à l'intégration du tatouage audio dans les techniques de renforcement de la qualité des signaux audio précitées. Dans ce contexte, le tatouage audio remplace le canal auxiliaire précédent et œuvre comme une mémoire du signal originel, porteuse d'informations nécessaires pour la correction d'harmonicité et la réduction de pré-écho. Cette seconde partie a été précédée par une étape approfondie de l'évaluation des performances de la technique de tatouage adoptée en terme de robustesse à la compression MPEG (MP3, AAC et aacPlus).The goal of this thesis is to explore the idea of watermark for sound enhancement. Classically, watermark schemes are oriented towards security applications or maximization of the transmitted bit rates. Our approach is completely different. Our goal is to study how an audio watermarking can improve the quality of the host audio signal by exploiting the information it conveys. The audio watermarking is considered as a memory that carries information about the original signal.The low bitrate compression of audio signals is one of the applications covered by this concept. In this context, two objectives are proposed: reducing the pre-echo and the attack softening, two phenomena introduced by the perceptual audio coders, particularly AAC and MP3 encoders ; preserving the harmonicity of audio signals, distorted by coders with bandwidth extension, especially HE-AAC encoder. These coders are limited in the reconstruction of the high-frequency spectrum mainly because of the potential unpredictability of the fine structure of the latter, as well as imperfect indicators of tonal to noise.The first part of this manuscript presents the basic principles of low rate coding systems and studies the various distortions introduced by the latter. Based on this study, two solutions are proposed. The first one, principally aimed at reducing the pre-echo, consist in correcting the time envelope of the signal after reception by exploiting the prior knowledge of the temporal envelope of the original signal, which is assumed transmitted by an auxiliary channel at low bitrates (<500 bps). The second solution is to correct the harmonicity generated by coders with bandwidth extension. This primarily affects strongly harmonic signals (e.g. violin) and is perceived as a dissonance. We propose then to preserve the harmonicity of audio signals by spectral translations. The parameters being passed again by an auxiliary channel at low bitrates.The second part of this document is dedicated to the integration of audio watermarking techniques in the solution presented in the first part. In this context, the audio watermarking replaces the previous auxiliary channel and is regarded as a memory of the original signal, carrying information necessary for the correction of harmonicity and the pre-echo reduction.PARIS5-Bibliotheque electronique (751069902) / SudocSudocFranceF

A novel pseudonoise sequence for time-spread echo based audio watermarking

This paper deals with the problem of digital audio watermarking using echo hiding. Compared to many other methods for audio watermarking, echo hiding techniques exhibit advantages in terms of relatively simple encoding and decoding, and robustness against common attacks. The low security issue existing in most echo hiding techniques is overcome in the timespread echo method by using pseudonoise (PN) sequence as a secret key. In this paper, we propose a novel sequence, in conjunction with a new decoding function, to improve the imperceptibility and the robustness of time-spread echo based audio watermarking. Theoretical analysis and simulation examples illustrate the effectiveness of the proposed sequence and decoding function.<br /