An MDCT Domain Frame-Loss Concealment Technique for MPEG Advanced Audio Coding

Speech is decomposed into three different components, based on the idea of Daudet and Torresani (Signal Processing, vol. 82, no. 11, pp. 1595, 2002), as signal = tonal + transient + residual. The tonal and transient components are identified using a small number of coefficients of the modified discrete cosine transform (MDCT) and the wavelet transform, respectively. Determinations of the significant MDCT and wavelet coefficients in the algorithm of Daudet and Torresani, referred as the D&T algorithm, are achieved by thresholds. All MDCT coefficients are assumed to be independent as well as wavelet coefficients. However, the MDCT coefficients probably have statistical dependencies, namely the clustering and persistence properties, and so do the wavelet coefficients. We propose a modification to the D&T algorithm, that can capture statistical dependencies by utilizing the hidden Markov model. The Viterbi and the maximum a posteriori (MAP) algorithms, used to find the optimal state distribution, are applied to determine the significant MDCT and wavelet coefficients automatically. The modified algorithm was used to encode 43 monosyllabic consonant-vowel-consonant (CVC) words and 3 sentences. Results showed that the modified algorithm improves the coding efficiency by 37% compared with the threshold method of D&T algorithm when equal numbers of significant coefficients are used.

Given the popularity of audio and video applications, compressed audio has become an important carrier of covert communication on the Internet. Many novel compressed audio steganography schemes have emerged that offer good hiding capability and aural concealment. In this paper, a universal steganalysis scheme called MultiSpecNet is proposed to detect steganography based on multiple embedding domains (advanced audio coding (AAC) and MPEG-1 Audio Layer III (MP3)), which are currently the two most popular compressed audio standards. The basic idea is that modification of either domain by a steganography scheme will change the time-frequency relationship of the audio signal after decoding. The proposed approach adopts the spectrogram as the input feature to extract richer information. DeepResNet is used to learn the distinguishing feature representations, and multiscale spectrograms are used to enrich the feature diversity. The experimental results show that the proposed scheme is effective at detecting different steganography schemes based on the AAC and MP3 embedding domains. The detection accuracy of the proposed scheme is higher than that achieved by other state-of-the-art schemes. Using spectrograms as the input, DeepResNet achieves better performance than schemes using quantized modified discrete cosine transform (MDCT) coefficients and mel-spectrogram, although the quantized MDCT coefficient is the parameter modified by the steganography schemes directly and mel-spectrogram is very popular and effective for general audio signal analysis. To the best of our knowledge, this work is the first audio steganalysis scheme that can detect multiple steganography schemes in both the MP3 and AAC embedding domains. The method proposed in this paper can be extended to audio steganalysis for other codecs or for audio forensics purposes.

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

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.

In this paper, we present a novel audio denoising algorithm for MP3 (MPEGl-layer3) compressed audio, based on Modified Discrete Cosine Transform (MDCT). The proposed algorithm works directly on MP3 frames. Firstly, we get MDCT coefficients from MP3 audio semi-decoder. And then, the normal inverse Gaussian (NIG) probability density function is performed to modelling the statistical distribution of MDCT coefficient, based on its sparse features. In the denoising process, a maximum a posterior (MAP) estimator based on NIG density is exploited to remove noise from MP3 audio. The experimental results show that the proposed algorithm can remove noise directly from noisy MP3 audio, and the signal-to-noise ratio (SNR) of the compressed audio is improved obviously.

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

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.

We describe an efficient implementation for converting MPEG-2/MPEG-4 Advanced Audio Coding (AAC) encoded data to Dolby Digital AC-3. We describe many techniques to exploit the information in the AAC bitstream to simplify the AC-3 encoder. These techniques can be straightforwardly used in other transcoding schemes between different multimedia standards.

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 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.

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.

This paper proposes a new perceptual hashing algorithm for speech content identification with compressed domain based on MDCT (Modified Discrete Cosine Transform) Spectrum Entropy. It aims primarily to solve problems of large computational complexity and poor real-time performance that appear when applying traditional identification methods to the compressed speeches. The process begins by extracting the MDCT coefficients, which are the intermediately decoded results of compressed speeches in MP3 format. In order to reduce the computational complexity, these coefficients are divided into sub-bands and the energy of MDCT spectrum is then calculated. Subbands of MDCT spectrum energy are then mapped to a similar mass function in information entropy theory. The function will be used as a perceptual feature and set to extract binary hash values. Experimental results show that the proposed algorithm keeps greater robustness to content-preserving operations while also maintaining efficiency. As a result of the partial decoding process, the real-time performance can meet the requirements of applications in real-time communication terminals.

MPEG-1 layer-3, popularly known as MP3, has revolutionized the digital music domain. MP3 makes use of psychoacoustic modeling to achieve compression through the removal of perceptually irrelevant components of digital audio. The psychoacoustic model is the key element of perceptual coding and requires intensive FFT computation for calculating the frequency spectrum. This spectrum is used to compute masking thresholds. Thus, the original MP3 algorithm computes modified discrete cosine transform (MDCT) and FFT parallelly. The proposed algorithm is an alternative to this. We make use of complex modified discrete cosine transform (CMDCT) of the filter-bank outputs for generating MDCT coefficients as well as the frequency spectrum. This method requires fewer computations than the original method. A novel method of window switching, based on filter-bank output is used to simplify the overall algorithm. The proposed algorithm reduces the amount of computations for MP3 encoder while retaining the audio quality.

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.

A novel approach is presented to detect double compressed MP3 audio by frequency vibration. With the analysis of double compression effect on MDCT (Modified Discrete Cosine Transform) coefficients in MP3 audio, we propose a simple feature called FVV (frequency vibration value) to measure the vibration caused by double compression. The experimental results on challenging dataset show that our method outperforms most of the existing methods in double MP3 compression detection, especially with a second bitrate higher than the first one. Besides, we can also estimate the original bit-rate for a double compressed MP3 by this technique.