数字音频压缩自适应变换编码算法的研究

本文介绍了一种建立在心理声学基础上的适用于宽频带数字音频信号的数据压缩技术。我们利用计算机对整个编解码算法的全部过程进行了模拟，并采用Motorola公司的数字信号处理芯片DSP56009作为硬件平台，实现了高质量的音频信号压缩编码。试验效果压缩比达到1／10以下，没有明显的音质衰减。     

数字音频压缩中的变换编码算法

变换编码是音频压缩中的一个重要部分,文中叙述ＭＰＥＧ音频编码标准中的变换编码技术,包括改进余弦变换和反变换（ＭＤＣＴ和ＩＭＤＣＴ）时域混叠抵消与自适应窗选择,详细推导了ＭＤＣＴ和ＩＭＤＣＴ的快速算法。     

基于MPEG-AAC编码器的压缩域音频增强方法

本文基于MPEG-AAC音频编解码器,提出了一种压缩域的音频增强方法.首先,对含噪音频信号的比特流进行解码,得到含噪音频信号的MDCT系数;然后,利用修正的加权递归平均（Modified Weighted Recursive Averaging,MWRA）方法估计噪声功率;再者,利用基于听觉掩蔽原理的自适应β-阶双曲余弦（COSH）统计模型,对含噪音频的MDCT系数进行增强处理;最后,将增强后的MDCT系数重新量化编码,得到用于解码的增强比特流实验结果表明,本文提出的方法能有效去除AAC解码音频信号中的多种背景噪声,其性能明显优于参考方法.     

A survey of efficient MDCT implementations in MP3 audio coding standard: Retrospective and state-of-the-art

This tutorial paper describes various efficient implementations (published and new unpublished) of the forward and backward modified discrete cosine transform (MDCT) in the MPEG layer III (MP3) audio coding standard developed in the time period 1990-2010, including the efficient implementation of polyphase filter banks for completeness. The efficient MDCT implementations are discussed in the context of (fast) complete analysis/synthesis MDCT filter banks in the MP3 encoder and decoder. In general, for each efficient forward/backward MDCT block transforms implementation are presented: complete formulas or sparse matrix factorizations of the algorithm, the corresponding signal flow graph for the short audio block and the total arithmetic complexity as well as the useful comments related to improving the arithmetic complexity and a possible structural simplification of the algorithm. Finally, all efficient forward/backward MDCT implementations are compared both in terms of the arithmetic complexity and structural simplicity. It is important to note that almost all presented algorithms can be also used for the 2ⁿ-length data blocks in others MPEG audio coding standards and proprietary audio compression algorithms.     

一种结合正弦参数提取和MDCT滤波器组的频谱优化方法

通过研究信号中正弦成分对改进的离散余弦变换(MDCT)的频谱产生的影响，针对使用MDCT滤波器组的音频编码系统，结合正弦参数提取，提出了一种对MDCT频谱进行优化的方法。这种方法利用MDCT和离散傅里里变换(DFT)之间的关系，在提取正弦参数的基础之上，通过简单线性变换求得MDCT谱，并在频域分离信号中的部分准静态正弦．从而达到优化MDCT频谱的目的。     

一种符合ITU-T指标要求的嵌入式立体声语音频编码方法

 基于国际电信联盟标准化组织(ITU-T)编码标准G.729.1,本文提出了一种嵌入式变速率立体声语音与音频编码方法.本算法利用G.729.1和改进的调制叠接变换(Modulated Lapped Transform,MLT)编码技术对输入信号的中值与边带信息进行分层编码,形成具有嵌入式结构的码流.编码器可处理宽带和超宽带的立体声信号,宽带立体声信号编码的最大码率为48kb/s,超宽带立体声信号编码的最大速率为64kb/s.实现结果表明,本编码器的编码质量均达到了ITU-T对G.EV-VBR立体声编码的指标要求.     

一种基于WLP-MDCT混合音频编码算法

介绍了一种卷曲线性预测(Warped Linear Prediction,WLP)与改进型离散余弦变换(Modified Discrete Cosine Transform,MDCT)混合音频编码的算法。WLP技术用来构造一对前滤波器和后滤波器,其中前滤波器用来降低MDCT编码过程中前回声的产生,后滤波器可以对量化噪声进行整形,从而进一步提高重建音频的主观听觉质量。实验结果表明该算法确实有效可行。     

Transform Coding Based on Source Filter Model in the MDCT Domain

State‐of‐the‐art voice codecs have been developed to extend the input bandwidth to enhance quality while maintaining interoperability with a legacy codec. Most of them employ a modified discrete cosine transform (MDCT) for coding their extended band. We propose a source filter model‐based coding algorithm of MDCT spectral coefficients, apply it to the ITU‐T G.711.1 super wideband (SWB) extension codec, and subjectively test it to validate the model. A subjective test shows a better quality over the standardized SWB codec.     

一种基于正弦模型的音频压缩算法

提出一种适用于无线移动通信的低码率音频压缩算法。该算法基于正弦模型，而且针对极低码率的应用做了修正，提高了重建音频的质量。这些修正包括：自适应变换的分析长度，用于匹配跟踪算法和参数量化的心理声学模型以及频域的无相位音频重建算法。主观测试表明，在0.5bit/抽样的码率要求下，重建信号达到并超过了调幅广播的音频质量。     

Generalized fast mixed-radix algorithm for the computation of forward and inverse MDCTs

This paper presents a generalized mixed-radix decimation-in-time (DIT) fast algorithm for computing the modified discrete cosine transform (MDCT) of the composite lengths N=2×q^m, m≥2, where q is an odd positive integer. The proposed algorithm not only has the merits of parallelism and numerical stability, but also needs less multiplications than that of type-IV discrete cosine transform (DCT-IV) and type-II discrete cosine transform (DCT-II) based MDCT algorithms due to the optimized efficient length-(N/q) modules. The computation of MDCT for composite lengths N=q^m×2ⁿ, m≥2, n≥2, can then be realized by combining the proposed algorithm with fast radix-2 MDCT algorithm developed for N=2ⁿ. The combined algorithm can be used for the computation of length-12/36 MDCT used in MPEG-1/-2 layer III audio coding as well as the recently established wideband speech and audio coding standards such as G.729.1, where length-640 MDCT is used. The realization of the inverse MDCT (IMDCT) can be obtained by transposing the signal flow graph of the MDCT.     

New fast computational structures for an efficient implementation of the forward/backward MDCT in MP3 audio coding standard

New fast computational structures identical for an efficient implementation of both the forward and backward modified discrete cosine transform (MDCT) in MPEG-1/2 Layer III (MP3) audio coding standard are described. They are based on a new proposed universal fast rotation-based MDCT computational structure [V. Britanak, New universal rotation-based fast computational structures for an efficient implementation of the DCT-IV/DST-IV and analysis/synthesis MDCT/MDST filter banks, Signal Processing 89 (11) (November 2009) 2213–2232]. New fast computational structures are derived in the form of a linear code and they are particularly suitable for high-performance programmable DSP processors. For the short audio block it is shown that our efficient MDCT implementation in MP3 can be modified to achieve the same minimal multiplicative complexity compared to that of Dai and Wagh [An MDCT hardware accelerator for MP3 audio, in: Proceedings of the IEEE Symposium on Application Specific Processors (SASP’2008), Anaheim, CA, June 2008, pp. 121–125].     

Low noise reversible MDCT (RMDCT) and its application in progressive-to-lossless embedded audio coding

A reversible transform converts an integer input to an integer output, while retaining the ability to reconstruct the exact input from the output sequence. It is one of the key components for lossless and progressive-to-lossless audio codecs. In this work, we investigate the desired characteristics of a high-performance reversible transform. Specifically, we show that the smaller the quantization noise of the reversible modified discrete cosine transform (RMDCT), the better the compression performance of the lossless and progressive-to-lossless codec that utilizes the transform. Armed with this knowledge, we develop a number of RMDCT solutions. The first RMDCT solution is implemented by turning every rotation module of a float MDCT (FMDCT) into a reversible rotation, which uses multiple factorizations to further reduce the quantization noise. The second and third solutions use the matrix lifting to implement a reversible fast Fourier transform (FFT) and a reversible fractional-shifted FFT, respectively, which are further combined with the reversible rotations to form the RMDCT. With the matrix lifting, we can design the RMDCT that has less quantization noise and can still be computed efficiently. A progressive-to-lossless embedded audio codec (PLEAC) employing the RMDCT is implemented with superior results for both lossless and lossy audio compression.     

A Low‐Delay MDCT/IMDCT

This letter presents an algorithm for selecting a low delay for the modified discrete cosine transform (MDCT) and inverse MDCT (IMDCT). The implementation of conventional MDCT and IMDCT requires a 50% overlap‐add (OLA) for a perfect reconstruction. In the OLA process, an algorithmic delay in the frame length is employed. A reduced overlap window and MDCT/IMDCT phase shifting is used to reduce the algorithmic delay. The performance of the proposed algorithm is evaluated by applying the low‐delay MDCT to the G. 729.1 speech codec.     

High-quality audio transform coding at 64 kbps

This paper presents a transform coding algorithm devoted to high quality audio coding at a bit rate of 64 kbps per monophonic channel. It enables the transmission of a high quality stereo sound through the basic access (2B channels) of ISDN. Although a complete system including framing, synchronization and error correction has been developed, only the bit rate compression algorithm is described here. A detailed analysis of the signal processing techniques such as the time/frequency transformation, the pre-echo reduction by adaptive filtering, the fast algorithm computations, etc., is provided. The use of psychoacoustical properties is also precisely reported. Finally, some subjective evaluation results and one real time implementation of the coder using the ATT DSP32C digital signal processor are presented     

Fast mode decision algorithm for intra prediction in H.264/AVC with integer transform and adaptive threshold

Compared with other existing video coding standards, H.264/AVC can achieve a significant improvement in compression performances. A robust criterion named the rate distortion optimization (RDO) is employed to select the optimal coding modes and motion vectors for each macroblock (MB), which achieves a high compression ratio while leading to a great increase in the complexity and computational load unfortunately. In this paper, a fast mode decision algorithm for H.264/AVC intra prediction based on integer transform and adaptive threshold is proposed. Before the intra prediction, integer transform operations on the original image are executed to find the directions of local textures. According to this direction, only a small part of the possible intra prediction modes are tested for RDO calculation at the first step. If the minimum mean absolute error (MMAE) of the reconstructed block corresponding to the best mode is smaller than an adaptive threshold which depends on the quantization parameter (QP), the RDO calculation is terminated. Otherwise, more possible modes need to be tested. The adaptive threshold aims to balance the compression performance and the computational load. Simulation results with various video sequences show that the fast mode decision algorithm proposed in this paper can accelerate the encoding speed significantly only with negligible PSNR loss or bit rate increment. This work is supported in part by China National Natural Science Foundation (CNSF) under Project No.60572045, the Ministry of Education of China Ph.D. Program Foundation under Project No.20050698033, and by a Cooperation Project (2005.7– 2007.7) with Microsoft Research Asia.     

Low bit rate transparent audio compression using adapted wavelets

Describes a novel wavelet based audio synthesis and coding method. The method uses optimal adaptive wavelet selection and wavelet coefficients quantization procedures together with a dynamic dictionary approach. The adaptive wavelet transform selection and transform coefficient bit allocation procedures are designed to take advantage of the masking effect in human hearing. They minimize the number of bits required to represent each frame of audio material at a fixed distortion level. The dynamic dictionary greatly reduces statistical redundancies in the audio source. Experiments indicate that the proposed adaptive wavelet selection procedure by itself can achieve almost transparent coding of monophonic compact disk (CD) quality signals (sampled at 44.1 kHz) at bit rates of 64-70 kilobits per second (kb/s). The combined adaptive wavelet selection and dynamic dictionary coding procedures achieve almost transparent coding of monophonic CD quality signals at bit rates of 48-66 kb/s     

基于小波变换和音质模型的音频编码算法研究

音频编码要解决的问题是以最小感知失真用低速率表达音频信号.本文设计了一种基于正交小波变换和音质模型的自适应比特分配音频编码算法,它可以将1411.2kbit/s的双声道立体声高保真音频信号压缩成低至32kbit/s的速率,并保持很好的音频质量.     

AAC编码算法的快速实现

Advanced Audio Coding(AAC)是一种高质量的音频编码标准,其编码算法的运算复杂度很高。本文针对AAC的心理声学模型谱估计采用SDFT'谱代替标准建议的DFI'谱,对其比特分配模块则提出了一种自适应的最佳全局量化阶搜索方法,此外,还采用基于FFT的MDC'I'快速算法,大大降低了AAC编码算法的运算复杂度。     

High-quality audio compression using an adaptive wavelet packetdecomposition and psychoacoustic modeling

This paper presents a technique to incorporate psychoacoustic models into an adaptive wavelet packet scheme to achieve perceptually transparent compression of high-quality (34.1 kHz) audio signals at about 45 kb/s. The filter bank structure adapts according to psychoacoustic criteria and according to the computational complexity that is available at the decoder. This permits software implementations that can perform according to the computational power available in order to achieve real time coding/decoding. The bit allocation scheme is an adapted zero-tree algorithm that also takes input from the psychoacoustic model. The measure of performance is a quantity called subband perceptual rate, which the filter bank structure adapts to approach the perceptual entropy (PE) as closely as possible. In addition, this method is also amenable to progressive transmission, that is, it can achieve the best quality of reconstruction possible considering the size of the bit stream available at the encoder. The result is a variable-rate compression scheme for high-quality audio that takes into account the allowed computational complexity, the available bit-budget, and the psychoacoustic criteria for transparent coding. This paper thus provides a novel scheme to marry the results in wavelet packets and perceptual coding to construct an algorithm that is well suited to high-quality audio transfer for Internet and storage applications     

Mixed-Radix Algorithm for the Computation of Forward and Inverse MDCTs

The modified discrete cosine transform (MDCT) and inverse MDCT (IMDCT) are two of the most computationally intensive operations in MPEG audio coding standards. A new mixed-radix algorithm for efficiently computing the MDCT/IMDCT is presented. The proposed mixed-radix MDCT algorithm is composed of two recursive algorithms. The first algorithm, called the radix-2 decimation-in-frequency algorithm, is obtained by decomposing an N-point MDCT into two MDCTs with the length N/2. The second algorithm, called the radix-3 decimation-in-time algorithm, is obtained by decomposing an N -point MDCT into three MDCTs with the length N/3. Since the proposed MDCT algorithm is also expressed in the form of a simple sparse matrix factorization, the corresponding IMDCT algorithm can be easily derived by simply transposing the matrix factorization. Comparison of the proposed algorithm with some existing ones shows that our proposed algorithm is more suitable for parallel implementation and particularly suitable for the layer III of MPEG-1 and MPEG-2 audio encoding and decoding. Moreover, the proposed algorithm can be easily extended to the multidimensional case by using the vector-radix method.