This paper presents the complete design of an audio data hiding system destined to transmit a binary information via the audio communication channel for audio annotation applications. The proposed system is based on an innovative embedding strategy. It consists of 1) a new inaudibility control procedure that locally regulates the watermark transparency, 2) an informed embedding function that maximizes system robustness to additive channel perturbation by using a new criterion of robustness - that is by maintaining the error probability at a fixed value, and 3) an efficient and low computational cost synchronization mechanism. System performance in terms of inaudibility of the watermark, transmission reliability with respect to various perturbations, and computational cost is evaluated on real audio signals to determine the efficiency of the proposed embedding strategy

A particular application of audio watermarking systems consists in using the audio signal as a transmission channel for binary information. In this context, reliability criteria with respect to transmission rate, such as bit error rate (BER), is a major issue that defines system performance. Current research has already proven the advantage of using informed embedding rather than blind embedding. In this paper, we present a new informed watermarking system based on an embedding strategy that maximizes robustness to perturbations such as compression and analog transmission. The proposed system performance is then compared to the equivalent blind embedding system performance. Experimental results permit us to assess the achieved efficiency. For transmission rates, up to 500 bits/s, BER are divided by 10 when channel perturbation is compression and divided by 4 in the case of analog transmission.

A 64 kbps coder of wideband (15 kHz) monophonic audio signals is described. Its structure is based on the transform coded excitation scheme, adopted to 7 kHz band signals. Significant modifications are proposed, that yield reduction of delay while keeping an almost transparent quality of speech and music, equivalent to that provided by the MPEG-1, layer II audio standard at the same bit rate. Algorithmic delay has been reduced to 17 ms-approximately 1/3 the delay of the MPEG coder.

The authors investigate several algorithms that construct the input for the synthesis filter in the CELP coder, they present them under the same formalism, and compare their performances. They model the excitation vector by a linear combination of K signals, which are issued from K codebooks and multiplied by K associated gains. They demonstrate that this generalized form incorporates several particular coders such as code excited linear predictive coders, multipulse coders, self excited vocoders, etc. The least squares minimization problem is presented afterwards. In the case of orthogonal codebooks, they show that the optimal solution of this least squares problem is equivalent to orthogonal transform coding. They use the Karhunen-Loeve transform to design the corresponding orthogonal codebooks. In the case of nonorthogonal codebooks, they are restricted to suboptimal iterative algorithms for index selection and gain computation. They present some new algorithms based on orthogonalization procedures and QR factorizations that attempt to reduce this suboptimality. In a particular case, when the excitation is modeled using one gain coefficient (for example, ternary excitation or concatenation of short codebook vectors), an iterative angle minimization algorithm is proposed for index selection. The different extraction algorithms are compared with regard to the resulting coder complexity and synthetic speech quality. They find a particularly attractive method that consists of modeling the excitation with one unique gain.>

The authors investigate three algorithms that orthogonalize codebooks in a multi-stage code excited linear prediction (CELP) speech coder. They carry out the same processing, a locally optimal modeling of the perceptual signal, but the computational costs differ. The authors show that the recursive modified Gram-Schmidt algorithm extra computational cost is less than the other two. An orthogonal codebook is defined a priori and the authors observe an equivalence to orthogonal transform coding. Three methods based on the Karhunen-Loeve transform for designing this codebook are compared. A partitioned shape-gain VQ is applied in the transform domain.>

A generalized coder that includes all types of excitation is presented. In this analysis-by-synthesis scheme, maximization formulae are the same regardless of the kind of excitation. The total excitation is expressed as a linear combination of excitation vectors. Given the number of excitation vectors or, equivalently, the bit rate of the coder, finding the vectors and their corresponding gains is a specific least-squares problem. The standard way this problem is usually solved is given, and three alternative mathematical approaches are proposed. These approaches are a Gram-Schmidt orthogonalization, a Choleski decomposition, a Householder transform. All these procedure have the same geometrical interpretation and lead to the same floating point simulation results.>