A Transcoding System for Audio Standards

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.

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.

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.

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.

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.

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.

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.

In this paper, the relationship between the features of the input audio and its corresponding transparent bitrate is studied. This relationship can be used to pre-estimate the transparent bitrate required for an input audio file, so that this bitrate can be used for encoding and the transparent quality can be achieved with the minimum compressed file size. Two features are selected and tested. Firstly, the perceptual feature is extracted from the test sequences using MPEG-4 advanced audio coding. The results of the signal-to-mask ratio and the transparent bitrate are compared. Secondly, the relationship between the music genre of the audio and its transparent bitrate is explored.

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.

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.

MPEG-4 scalable lossless (SLS) coding is a unified solution for demands in high compression perceptual audio and high quality lossless audio. It provides a fine-grain scalable extension to the MPEG-4 advanced audio coding (AAC) perceptual audio coder up to fully lossless reconstruction. It is observed that the quality of SLS coded audio is still far from optimal when the core bitrate is low. With this observation, an adaptive bit-plane scanning (ABPS) method is proposed in this paper. In ABPS, the bit-plane scanning order is adaptive to the energy distribution of the signal to be coded. The results show that with ABPS, the perceptual quality of the audio under low core bitrate scenario is significantly improved without any extra payload.

In this paper, we provide some insights into the IEEE Standard for Systems of Advanced Audio and Video Coding (IEEE 1857.3). Specifically, the standard defines file format and real-time transport protocol (RTP) payload format for IEEE 1857 video and IEEE 1857.2 audio. The storage of video and audio not only uses the existing capabilities of the ISO base media file format (BMFF), but also specifies extensions to support specific features of the IEEE 1857 video codec and IEEE 1857.2 audio codec. The target applications and services of the standard include but are not limited to Internet media streaming, IPTV, video conference, video telephony, and video-on-demand.

Scalable-to-Lossless (SLS) audio compression, as standardized by MPEG, provides a lossy base layer compatible with the Advanced Audio Coding (AAC) format, ensuring state-of-the-art quality in the base layer, and additional fine grained enhancements that eventually provide a lossless compressed version of the signal. While SLS offers highly efficient lossless compression, the perceptual quality of its intermediate lossy layers has been observed to be suboptimal. This paper proposes a modified SLS audio coding scheme that provides enhanced perceptual quality at an intermediate bit-rate, at the expense of an additional parameter per frequency band as side-information. This scheme when coupled with a trellis-based optimization algorithm is demonstrated to outperform, in terms of quality at the intermediate bit-rate, both standard SLS and a recent perceptually enhanced variant, with minimal degradation in lossless coding performance.

Opus, Advanced Audio Coding (AAC), and mp3 are well known audio codecs in the market. The goal of this research is to compare the quality of above three audio codecs at different bitrates and to determine which codec can maintain the best quality overall. An objective evaluation approach is presented and preliminary results are summarized.

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