A Universal Audio Steganalysis Scheme Based on Multiscale Spectrograms and DeepResNet

We propose a frame loss concealment technique for decoders compatible with MPEG advanced audio coding (AAC). The spectral information of the lost frame is estimated in the modified discrete cosine transform (MDCT) domain via efficient techniques that are tailored to individual source signal components: in noise-like spectral bins the MDCT coefficients are obtained by shaped-noise insertion, while coefficients in tone-dominant bins are estimated by frame interpolation followed by a refinement procedure so as to optimize the fit of the concealed frames with neighboring frames. Experimental results demonstrate that the proposed technique offers performance superior to techniques adopted in commercial AAC decoders.

The modified discrete cosine transform (MDCT) is widely used in current perceptual audio coding schemes. This paper presents an integer approximation of this lapped transform, called integer MDCT, which is derived from the MDCT using lifting scheme, and a scalable lossless system based on MPEG AAC and Integer MDCT is presented providing a lossless enhancement of AAC coding scheme, which provides bit-exact reconstruction of the input signal independent of the implementation accuracy of the AAC core coder

The modified discrete cosine transform (MDCT) has been widely used in many transform audio coding algorithms such as MPEG-1/2 layer III (mp3), MPEG-2/4 AAC, Dolby AC2/AC3, and numerous experimental audio coding algorithms. In this paper, we study the probabilistic distribution properties of the MDCT coefficient for audio signals. It is shown that the generalized Gaussian function with distribution parameter r=0.5 or r=1 (Laplacian) provides a good approximation to the distributions of MDCT coefficients for a variety of audio signals. Results from our study also show that although the distribution of these coefficients is not strictly Laplacian, the divergence between them is in fact very small. Therefore, it leads to only marginal redundancy if these coefficients are simply coded with some low-complexity codes designed for Laplacian sources

In this paper, we describe the implementation of the multi-channel AAC encoder system for digital audio broadcasting. The encoder system is based on MPEG-2/4 advanced audio coding (AAC) and is capable of real-time encoding up to 5.1 channel audio. To give a flexible functionality, it consists of multiple DSPs, IEC61937 and TCP/IP interface and 6 channel audio input facilities. The reference AAC decoder was implemented for verification test of the encoder. The encoder system is also integrated with the AAC streaming system for interoperation test. Through these tests, the encoder system is verified to be a good solution for high quality audio broadcasting.

The perfect reconstruction cosine/sine-modulated filter banks belonging to the class of modulated filter banks have been studied extensively due to their attractive features (simple structure, analysis and synthesis filters are of equal length, low computational complexity), and consequently, they have received a great interest in audio coding applications. In fact, they are employed in the international speech and audio coding standards and proprietary audio compression algorithms. The oddly and evenly stacked modified discrete cosine transform (MDCT) and the corresponding modified discrete sine transform (MDST), the modulated lapped transform (MLT), the extended lapped transforms (ELTs), and their biorthogonal versions are real-valued cosine/sine-modulated filter banks satisfying the perfect reconstruction property. The modulated complex lapped transform (MCLT) is the complex-valued filter bank whose real part is the MLT or equivalently, the oddly stacked MDCT, and the imaginary part is the oddly stacked MDST. In this chapter, definitions, general properties, and matrix representations of the MDCT/MDST, MLT, ELT, and MCLT filter banks are presented. In order to an analysis/synthesis filter bank be perfect reconstruction, the necessary and sufficient conditions imposed on the analysis and synthesis windowing functions play an important role. Therefore, additionally the windowing procedure and perfect reconstruction (biorthogonal) conditions in the case of identical and (nonidentical) analysis and synthesis windowing functions, design of a windowing function including definitions of commonly windowing functions used in audio coding applications, adaptive switching of transform block sizes and windowing functions, and general perfect reconstruction conditions for the ELT filter bank with multiple overlapping factor both for the orthogonal and biorthogonal cases are derived and/or discussed in detail.

Audio compression systems exploit periodicity in signals to remove inter-frame redundancies via the long term prediction (LTP) tool. This simple tool capitalizes on the periodic component of the waveform by selecting a past segment as the basis for prediction of the current frame. However, most audio signals are polyphonic in nature, containing a mixture of several periodic components. While such polyphonic signals may themselves be periodic with overall period equaling the least common multiple of the individual component periods, the signal rarely remains sufficiently stationary over the extended period, rendering the LTP tool suboptimal. Instead of seeking a past segment that represents a “compromise” for incompatible component periods, we propose a more complex filter that predicts every periodic component of the signal from its immediate history, and this is achieved by cascading LTP filters, each corresponding to individual periodic component. We also propose a recursive “divide and conquer” technique to estimate parameters of all the LTP filters. We then demonstrate the effectiveness of cascaded LTP in two distinct settings of the ultra low delay Bluetooth Subband Codec and the MPEG Advanced Audio Coding (AAC) standard. In MPEG AAC, we specifically adapt the cascaded LTP parameter estimation to take into account the perceptual distortion criteria, and also propose a low decoder complexity variant. Objective and subjective results for all the settings validate the effectiveness of the proposal on a variety of polyphonic signals.

This paper addresses the extraction of frequency, amplitude, and phase of sinusoidal components directly from modified discrete cosine transform (MDCT) domain, which is broadly used in many digital audio applications. An efficient method is proposed for portable device based on the approximation of MDCT expression and the simple combinations of local maximum MDCT coefficients. By extracting the frequency and amplitude of a stationary sinusoid, the experimental results demonstrate that the proposed algorithm produces accurate results with low computational complexity.

Most existing audio steganographic methods embed secret messages according to a pseudorandom number generator, thus some auditory sensitive parts in cover audio, such as mute or near-mute segments, will be contaminated, which would lead to poor perceptual quality and may introduce some detectable artifacts for steganalysis. In this paper, we propose a novel adaptive audio steganography in the time domain based on the advanced audio coding (AAC) and the Syndrome-Trellis coding (STC). The proposed method firstly compresses a given wave signal into AAC compressed file with a high bitrate, and then obtains a residual signal by comparing the signal before and after AAC compression. According to the quantity and sign of the residual signal, \(\pm 1\) embedding costs are assigned to the audio samples. Finally, the STC is used to create the stego audio. The extensive results evaluated on 10,000 music and 10,000 speech audio clips have shown that our method can significantly outperform the conventional \(\pm 1\) LSB based steganography in terms of security and audio quality.

Enhanced aacPlus is an audio codec which is composed of advanced audio coding (AAC), spectral band replication (SBR), and parametric stereo (PS) for efficient audio coding at low bit rates. We propose a new coding scheme for lossless bit rate reduction of PS in enhanced aacPlus. We first determine the optimal contexts for context-based coding of quantized stereo parameter indexes in PS. Then we propose a coding scheme which is based on context-based coding using the determined optimal contexts in conjunction with arithmetic coding. The proposed scheme has normal and memory-reduced versions, both of which are shown to guarantee significant and consistent lossless bit rate reduction of PS. The saved bits by the proposed scheme can be used for the other modules of enhanced aacPlus, AAC and SBR.

AAC(Advanced Audio Coding) is the most popular audio compression standard and used widely in recent years. The steganography schemes of AAC emerged gradually. This paper presents a novel steganalysis method to attack the steganography of Huffman codebook, which hide information by modifying the codebook of each scale factor band(SFB), and have good imperceptivity and security. Based on the correlation of neighboring SFBs' codebook, the paper proposes to extract the Markov transition probability of adjacent SFBs' codebook as steganalysis feature, and adopt calibration to improve the accuracy. Extensive experiments demonstrate the effectiveness of the proposed methods. To the best of our knowledge, this piece of work is the first one to detect AAC steganography of Huffman codebook.

The Dolby Digital (AC-3) and the Dolby Digital Plus or Enhanced AC-3 (E-AC-3) systems are currently the key enabling technologies for high-quality compression of digital audio signals. For the time-to-frequency transformation of an audio data block and vice versa, both systems have adopted a long transform being the modified discrete cosine transform (MDCT). The AC-3 additionally defines two variants of cosine-modulated filter banks called the first and second short transforms. A fast conversion algorithm is presented to convert the frequency coefficients of the long (MDCT) transform to those of two short transforms and vice versa, directly in the frequency domain. It is based on a block sparse matrix factorization of a conversion matrix. The fast conversion algorithm is efficient in terms of the structural simplicity, arithmetic complexity and memory requirements compared to the obvious conversion methods. Moreover, the existing AC-3 fast computational modules may be simply re-used in the conversion procedures. Consequently, the E-AC-3 to AC-3 bit stream conversion and the AC-3 to E-AC-3 bit stream transcoding can be realized in a simplified and efficient way, thus minimizing the amount of partial decoding/encoding and memory requirements during the conversion and transcoding processes.

This study presents several optimization approaches for the MPEG-2/4 Audio Advanced Coding (AAC) Low Complexity (LC) encoding and decoding processes. Considering the power consumption and the peripherals required for consumer electronics, this study adopts the TI OMAP5912 platform for portable devices. An important optimization issue for implementing AAC codec on embedded and mobile devices is to reduce computational complexity and memory consumption. Due to power saving issues, most embedded and mobile systems can only provide very limited computational power and memory resources for the coding process. As a result, modifying and simplifying only one or two blocks is insufficient for optimizing the AAC encoder and enabling it to work well on embedded systems. It is therefore necessary to enhance the computational efficiency of other important modules in the encoding algorithm. This study focuses on optimizing the Temporal Noise Shaping (TNS), Mid/Side (M/S) Stereo, Modified Discrete Cosine Transform (MDCT) and Inverse Quantization (IQ) modules in the encoder and decoder. Furthermore, we also propose an efficient memory reduction approach that provides a satisfactory balance between the reduction of memory usage and the expansion of the encoded files. In the proposed design, both the AAC encoder and decoder are built with fixed-point arithmetic operations and implemented on a DSP processor combined with an ARM-core for peripheral controlling. Experimental results demonstrate that the proposed AAC codec is computationally effective, has low memory consumption, and is suitable for low-cost embedded and mobile applications.

In this paper, we propose an effective audio steganalysis scheme based on deep residual convolutional networks in the temporal domain. Firstly, considering the weak difference between cover and stego, a high pass filter is adopted in the proposed network which is used to calculate the residual map of the audio signal. Then, comparing with convolutional neural networks (CNNs) based audio steganalysis in recent studies, the deeper network structure and complicated convolutional modules are considered to capture the complex statistical characteristic of steganography. Finally, batch normalization layers and shortcut connections are applied to decrease the dangers of over-fitting and accelerate the convergence of back-propagation. In the experiments, we compared the proposed scheme with CNNs based and hand-crafted features based audio steganalysis methods to detect the various steganographic algorithms on speech and music audio clips respectively. The experimental results demonstrate that the proposed scheme is able to detect multiple state-of-the-art audio steganographic schemes with different payloads effectively and outperforms several recently proposed audio steganalysis methods.

In this paper, we develop a 3D audio reproduction scheme for the purpose of delivering audio over IP networks. In this scheme, audio streams constructed at a server are composed of the TCP/IP header followed by multi-channel audio data compressed by MPEG advanced audio coding (AAC), and the decoded audio signals are played out on a stereo loudspeaker system at the client. Since the audio source is recorded by a multi-channel microphone but the playout is dedicated to stereo speakers, the quality mismatch between the multi-channel and the stereo system should be overcome. As a potential solution, we first investigate the effect of 3D audio processing on the audio quality at the client by applying a head-related transfer function (HRTF). Next, a crosstalk cancellation process is applied to the audio with 3D effects in order to improve the immersion of the processed 3D effects on a stereo loudspeaker system. Finally, we evaluate the performance of the 3D audio reproduction system in terms of the identification of an audio source and quality comparison before and after applying the crosstalk cancellation technique.

This paper presents a novel parametric vector quantization (PVQ) scheme as the secondary encoding to code perceptually salient percussive sounds such as drums in the time domain. It is deployed to improve the quality of service in the case of packet losses during percussive events. As a generalization and an improvement of our earlier system, the new scheme can achieve a better balance between bandwidth efficiency and error robustness. The proposed coding technique has been implemented with the MPEG-2 AAC (advanced audio coding) frame structure. Experimental results with music samples have shown the effectiveness of the proposed scheme.