Digital coding of waveforms: Principles and applications to speech and video

Installing video cameras in public facilities for surveillance becomes more and more popular. This paper proposes a novel authentication scheme based on chaotic semi-fragile watermarking. The timing information of video frames is modulated into the parameters of a chaotic system. The system output, which is a noise-like signal, is used as a watermark and embedded into the block-based discrete cosine transform domain. The embedded information is demodulated by a maximum likelihood estimator. Temporal tampering can be detected by the mismatch between the extracted and the observed timing information. In addition, the deviation of the extracted watermark from the original one allows us to locate spatial tampering. It is shown that the proposed scheme can satisfy the peculiar requirements of authenticating a digital video surveillance system.

Chaotic Watermarking for Video Authentication in Surveillance Applications

Digital speech watermarking is a robust way to hide and thus secure data like audio and video from any intentional or unintentional manipulation through transmission. In terms of some signal characteristics including bandwidth, voice/non-voice and production model, digital speech signal is different from audio, music and other signals. Although, various review articles on image, audio and video watermarking are available, there are still few review papers on digital speech watermarking. Therefore this article presents an overview of digital speech watermarking including issues of robustness, capacity and imperceptibility. Other issues discussed are types of digital speech watermarking, application, models and masking methods. This article further highlights the related challenges in the real world, research opportunities and future works in this area, yet to be explored fully.

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An overview of digital speech watermarking

We present a blind speech watermarking algorithm that embeds the watermark data in the phase of non-voiced speech by replacing the excitation signal of an autoregressive speech signal representation. The watermark signal is embedded in a frequency subband, which facilitates robustness against bandpass filtering channels. We derive several sets of pulse shapes that prevent intersymbol interference and that allow the passband watermark signal to be created by simple filtering. A marker-based synchronization scheme robustly detects the location of the embedded watermark data without the occurrence of insertions or deletions. In light of a potential application to analog aeronautical voice radio communication, we present experimental results for embedding a watermark in narrowband speech at a bit-rate of 450 bit/s. The recursive least-squares (RLS) equalization-based watermark detector not only compensates for the vocal tract filtering, but also recovers the watermark data in the presence of nonlinear phase and bandpass filtering, amplitude modulation, and additive white Gaussian noise (AWGN), making the watermarking scheme highly robust.

Speech Watermarking for Analog Flat-Fading Bandpass Channels

Digital audio watermarking robust against digital-to-analog (D/A) and analog-to-digital (A/D) conversions is an important issue. In a number of watermark application scenarios, D/A and A/D conversions are involved. In this article, we first investigate the degradation due to DA/AD conversions via sound cards, which can be decomposed into volume change, additional noise, and time-scale modification (TSM). Then, we propose a solution for DA/AD conversions by considering the effect of the volume change, additional noise and TSM. For the volume change, we introduce relation-based watermarking method by modifying groups of the energy relation of three adjacent DWT coefficient sections. For the additional noise, we pick up the lowest-frequency coefficients for watermarking. For the TSM, the synchronization technique (with synchronization codes and an interpolation processing operation) is exploited. Simulation tests show the proposed audio watermarking algorithm provides a satisfactory performance to DA/AD conversions and those common audio processing manipulations.

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Audio watermarking robust against D/A and A/D conversions

Speech bandwidth extension can be defined as the deliberate process of expanding the frequency range (bandwidth) for speech transmission. Its significant advancement in recent years has led to the technology being adopted commercially in several areas including psychoacoustic bass enhancement of small loudspeakers and the high frequency enhancement of perceptually coded audio. In this paper, a data hiding method based on dither quantization is used for speech bandwidth extension. More specifically, the out-of-band information is encoded and embedded into the narrowband speech without degrading the quality of the bandlimited signal. At the receiver, when the out-of-band information is extracted from the hidden channel, it can be used to combine with the bandlimited signal, providing a signal with a wider bandwidth. To encode the out-of-band speech more efficiently, acoustic phonetic classification is employed to generate three linear prediction (LP) codebook. The simulation results show that compared with using non-classified codebook, the propose scheme have a better bandwidth extension performance in terms of log spectral distortion (LSD).

Speech Bandwidth Extension by Data Hiding and Phonetic Classification

The current public switched telephone network (PSTN) is only able to deliver analog signals in a relatively narrow frequency band, about 300-3400 Hz. Such a bandwidth is so small that the intelligibility of speech frequently suffers from poor subjective quality. In order to improve the voice quality and intelligibility, it is proposed to use a data hiding technique to extend the PSTN channel bandwidth artificially. That is, the higher frequency components beyond the PSTN bandwidth are encoded and imperceptibly embedded into the narrowband signal. When the hidden signal is extracted at the receiver, wideband speech can be reconstructed with better perceptual quality. The advantage of the proposed scheme lies in that it is fully compatible with conventional end-user equipment, e.g., a plain ordinary telephone set (POTS). Experimental results show that the proposed scheme has a better performance than conventional bandwidth extension methods with speaker-independent training.

Artificial bandwidth extension of telephony speech by data hiding

This paper presents a graphical model approach that fuses thermal infrared (IR) and visible spectrum video for human tracking. The proposed model uses unobserved variables to describe the data in terms of the process that generates them. It is thus able to capture and exploit the statistical structure of the IR and the visible data separately, as well as their mutual dependencies. Model parameters are learned form data using the expectation maximization (EM) algorithm. Automatic calibration is performed as part of this procedure. Tracking is done by Bayesian inference of the object location from the observed data. The effectiveness of the proposed method is demonstrated by the experimental results on the video clips captured in real world scenarios.

Thermo-visual video fusion using probabilistic graphical model for human tracking

This paper proposes a novel method to perform concurrent data transmission through the public switched telephone network (PSTN). More specifically, a hidden channel is created by removing the voice components below the perceptual masking threshold. The method of code division multiple access (CDMA) is then used to transmit data through the hidden channel. At the receiver, since digital data is imperceptibly embedded into the hidden channel, conventional telephone sets are able to access the voice signal without additional equipments. Meanwhile, a customized telephone set can extract the hidden data and provide concurrent services. Experimental results show that a data rate at hundreds bits per second can be achieved with a bit error rate (BER) of 10/sup -3/.

Concurrent data transmission through PSTN by CDMA

Most of the current speech transmission systems are only able to deliver speech signals in a narrow frequency band. This narrowband speech is characterized by a thin and muffled sound. In this paper, we propose a data hiding scheme to artificially extend the speech bandwidth so that the speech quality can be improved. The hardware perspectives of implementing the proposed scheme are also discussed. A Xilinx MicroBlaze soft processor is used in this study. To balance the trade off between logic resource consumption and processing speed, the data hiding scheme are mainly implemented in application software while fast Fourier transform (FFT) is designed into a hardware acceleration model. Experimental results demonstrate the effectiveness of the proposed data hiding scheme.

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Siyue Chen

Papers

Speech Bandwidth Extension by Data Hiding and Phonetic Classification

Artificial bandwidth extension of telephony speech by data hiding

Thermo-visual video fusion using probabilistic graphical model for human tracking

Concurrent data transmission through PSTN by CDMA

Data Hiding for Speech Bandwidth Extension and its Hardware Implementation