The performance of adaptive acoustic echo cancelers (AEC) is sensitive to the nonstationarity and correlation of speech signals. In this paper, we explore a new approach based on an adaptive AEC driven by data hidden in speech, to enhance the AEC robustness. We propose a two-stage AEC, where the first stage is a classical NLMS-based AEC driven by the far-end speech. In the signal, we embed-in an extended conception of data hiding-an imperceptible white and stationary signal, i.e., a watermark. The goal of the second stage AEC is to identify the misalignment of the first stage. It is driven by the watermark solely and takes advantage of its appropriate properties (stationary and white) to improve the robustness of the two-stage AEC to the nonstationarity and correlation of speech, and thus reduce the overall system misadjustment. We test two kinds of implementations: in the first implementation, referred to as adaptive watermark driven AEC (A-WdAEC), the watermark is a white stationary Gaussian noise. Driven by this signal, the second stage converges faster than the classical AEC and provides better performance in steady state. In the second implementation, referred to as maximum length sequences WdAEC (MLS-WdAEC), the watermark is built from MLS. Thus, the second stage performs a block identification of the first stage misalignment, given by the circular correlation watermark/preprocessed version of the first stage residual echo. The advantage of this implementation lies in its robustness against noise and undermodeling. Simulation results show the relevance of the “WdAEC” approach, compared to the classical “error-driven AEC.”

Nonlinear audio system identification generally relies on Gaussianity, whiteness and stationarity hypothesis on the input signal, although audio signals are non-Gaussian, highly correlated and non-stationary. However, since the physical behavior of nonlinear audio systems is input-dependent, they should be identified using natural audio signals (speech or music) as input, instead of artificial signals (sweeps or noise) as usually done. We propose an identification scheme that conditions audio signals to fit the desired properties for an efficient identification. The identification system consists in (1) a Gaussianization step that makes the signal near-Gaussian under a perceptual constraint; (2) a predictor filterbank that whitens the signal; (3) an orthonormalization step that enhances the statistical properties of the input vector of the last step, under a Gaussianity hypothesis; (4) an adaptive nonlinear model. The proposed scheme enhances the convergence rate of the identification and reduces the steady state identification error, compared to other schemes, for example the classical adaptive nonlinear identification.

The harmonic plus noise model (HNM) is widely used for spectral modeling of mixed harmonic/noise speech sounds. In this paper, we present an analysis/synthesis system based on a long-term two-band HNM. “Long-term ” means that the time-trajectories of the HNM parameters are modeled using “smooth ” (discrete cosine) functions depending on a small set of parameters. The goal is to capture and exploit the longterm correlation of spectral components on time segments of up to several hundreds of ms. The proposed long-term HNM enables joint compact representation of signals (thus a potential for low bit-rate coding) and easy signal transformation (e.g. time stretching) directly from the long-term parameters. Experiments show that it can be compared favourably with the shortterm version in terms of parameter rates and signal quality. Index Terms: speech analysis/synthesis, harmonic + noise model, long-term processing.

The paper presents a new detection method for a spread spectrum and perceptual watermarking system, viewed as a hidden data transmission system, where the generic detection operation is achieved by a Wiener deconvolution filter. We point out the insufficiencies of the generic reception scheme concerning truncation errors and an ill-conditioned deconvolution operation, and we propose a cascade realization of the reception filter, which significantly improves the detection performances (to a multiplicative factor of 20) at higher bit rates, even in the presence of MPEG compression.

The paper presents an analysis of the perceptual impact of watermarking on the stationarity of audio signals. Indeed, the embedded watermark is piecewise stationary, so it modifies the stationarity of the original audio signal. This study is based on stationarity indices, which represent a measure of the variations in the spectral characteristics of signals, using time frequency representations. Simulation results with two kinds of signals, test signals and audio signals (speech and music) are presented. Comparison stationarity indices between watermarked and original audio signals show a significant stationarity enhancement of the watermarked signal, especially for transient attacks.