一种基于信号子空间和听觉掩蔽效应的语音增强方法

提文推导了基于离散余弦变换(DCT)的子空间分解法对有色噪声背景下的语音进行增强的公式,用基于听觉掩蔽效应的感智滤波器对增强后的信号频谱进行平滑以抑制背景噪声。几种噪声背景下对增强语音的客观测试表明,本文提出的方法可以有效地减少语音信号的失真度。     

An Approach to the Implementation of a Discrete Cosine Transform

An approach to the implementation of a discrete cosine transform (DCT) for application to coding speech is described. The approach is oriented toward single speech channel encoding. In addition, a detailed computer simulation of an adaptive transform coder is described. The purpose of the computer simulation is to determine the internal precision at various points in the implementation required to avoid subjective degradation. Specific recommmendations are made on the required internal precision in the implementation of the discrete cosine transform. A breadboard implementation of the DCT using SSI and MSI TTL logic based on the results of the computer simulation is reported.     

An objective measure for predicting subjective quality of speechcoders

A perceptually motivated objective measure for evaluating speech quality is presented. The measure, computed from the original and coded versions of an utterance, exhibits statistically a monotonic relationship with the mean opinion score, a widely used criterion for speech coder assessment. For each 10-ms segment of an utterance, a weighted spectral vector is computed via 15 critical band filters for telephone bandwidth speech. The overall distortion, called Bark spectral distortion (BSD), is the average squared Euclidean distance between spectral vectors of the original and coded utterances. The BSD takes into account auditory frequency warping, critical band integration, amplitude sensitivity variations with frequency, and subjective loudness     

Discrete cosine harmonic wavelet transform and its application to signal compression and subband spectral estimation using modified group delay

This paper proposes a new harmonic wavelet transform (HWT) based on discrete cosine transform (DCTHWT) and its application for signal or image compression and subband spectral estimation using modified group delay (MGD). Further, the existing DFTHWT has also been explored for image compression. The DCTHWT provides better quality decomposed decimated signals, which enable improved compression and MGD processing. For signal/image compression, compared to the HWT based on DFT (DFTHWT), the DCTHWT reduces the reconstruction error. Compared to DFTHWT for the speech signal considered for a compression factor of 0.62, the DCTWHT provides a 30% reduction in reconstruction error. For an image, the DCTHWT algorithm due to its real nature, is computationally simple and more accurate than the DFTHWT. Further compared to Cohen–Daubechies–Feauveau 9/7 biorthogonal symmetric wavelet, the DCTHWT, with its computational advantage, gives a better or comparable performance. For an image with 6.25% coefficients, the reconstructed image by DFTHWT is significantly inferior in appearance to that by DCTHWT which is reflected in the error index as its values are 3.0 and 2.65%, respectively. For spectral estimation, DCTHWT reduces the bias both in frequency (frequency resolution) and spectral magnitude. The reduction in magnitude bias in turn improves the signal detectability. In DCTHWT, the improvement in frequency resolution and the signal detectability is not only due to good quality DCT subband signals but also due to their stretching (decimation) in the wavelet transform. The MGD reduces the variance while preserving the frequency resolution achieved by DCT and decimation. In view of these, the new spectral estimator facilitates a significant improvement both in magnitude and frequency bias, variance and signal detection ability; compared to those of MGD processing of both DFT and DCT fullband and DFT subband signals.     

一种基于非线性特征的应力影响下变异语音识别方法

考虑到变异语音产生的非线性特点,本文提出了一种基于TEO能量算子倒谱特征的应力影响下变异语音识别方法。先将语音信号分割成21个不同频带的信号,然后计算TEO能量,最后进行对数运算和离散余弦变换。对航空模拟飞行器中采集的小词表特定人的识别实验,采用非线性分析的基于TEO能量算子倒谱特征的方法,能有效地提高变异语音的识别性能,比传统的基于MFCC特征的方法识别率提高了11.3％。     

Design and cryptanalysis of transform-based analog speechscramblers

Four discrete orthogonal transforms have been evaluated for their suitability for use in transform-based analog speech encryption. Subjective as well as objective tests were conducted to compare the residual intelligibility and the recovered speech quality under channel conditions. The cryptanalytic strengths of the schemes were then compared by applying a novel cryptanalytic attack which exploits the redundancy of speech using a spectral vector codebook. The results indicate that the discrete cosine transform (DCT) is the best transform to use in transform-based encryption. A modification of the DCT-based scheme which significantly improves the security of the scrambler is proposed     

Multilayer subband-based video coding

A motion compensated interframe subband coding algorithm suitable for a wide range of video coding applications is described. In this approach the spectrum of each frame of video signal is first decomposed into smaller frequency bands where each can then be coded accordingly. For the best performance a combination of hybrid DCT/DPCM (discrete cosine transform/differential pulse code modulation), interframe DPCM, and intraframe PCM was considered. To preserve its hierarchical structure each band is coded independently of higher frequency bands but can share information with the lower bands. A simulation was carried out for HDTV sequences     

Optimized quadtree for Karhunen-Loeve transform in multispectralimage coding

A new multispectral image compression technique based on the Karhunen-Loeve transform (KLT) and the discrete cosine transform (DCT) is proposed. The quadtree for determining the transform block size and the quantizer for encoding the transform coefficients are jointly optimized in a rate-distortion sense. The problem is solved by a Lagrange multiplier approach. After a quadtree is determined by this approach, a one-dimensional (1-D) KLT is applied to the spectral axis for each block before the DCT is applied on the spatial domain. The eigenvectors of the autocovariance matrix, the quantization scale, and the quantized transform coefficients for each block are the output of the encoder. The overhead information required in this scheme is the bits for the quadtree, KLT, and quantizer representation.     

800/920/1200bps改进型多带激励语音编码器的DSP实现

该文介绍了基于多带激励(MBE) 语音模型的改进型800/920/1200bps 语音编码器。该编码器采用二级离散余弦变换编码方案对逐帧变化的谱幅度参数进行自适应动态量化编码,大大减小了编码比特率,使得800~1200bps速率的MBE语音编码器的实现成为可能。该文重点介绍二级离散余弦变换编码方案及DSP硬件实现。     

Algorithm-based low-power transform coding architectures: themultirate approach

In most low-power VLSI designs, the supply voltage is usually reduced to lower the total power consumption. However, the device speed will be degraded as the supply voltage goes down. In this paper, we propose new algorithmic-level techniques to compensate the increased delays based on the multirate approach. We apply the technique of polyphase decomposition to design low-power transform coding architectures, in which the transform coefficients are computed through decimated low-speed input sequences. Since the operating frequency is M-times slower than the original design while the system throughput rate is still maintained, the speed penalty can be compensated at the architectural level. We start with the design of low-power multirate discrete cosine transform (DCT)/inverse discrete cosine transform (IDCT) VLSI architectures. Then the multirate low-power design is extended to the modulated lapped transform (MLT), extended lapped transform (ELT), and a unified low-power transform coding architecture. Finally, we perform finite-precision analysis for the multirate DCT architectures. The analytical results can help us to choose the optimal wordlength for each DCT channel under required signal-to-noise ratio (SNR) constraint, which can further reduce the power consumption at the circuit level. The proposed multirate architectures can also be applied to very high-speed block discrete transforms in which only low-speed operators are required     

Algorithmes de transformations de Fourier et en cosinus mono et bidimensionnels

This paper presents a fast algorithm for the computation of the discrete Fourier and cosine transform, and this for transform lengths which are powers of 2. This approach achieves the lowest known number of operations (multiplications and additions) for the discrete Fourier transform of real, complex, symmetrical and antisymmetrical sequences, for the odd discrete Fourier transform and for the discrete cosine transform. The extension to the two-dimensional Fourier and cosine transform is presented as well.     

Compression of multispectral images by spectral classification andtransform coding

This paper presents a new technique for the compression of multispectral images, which relies on the segmentation of the image into regions of approximately homogeneous land cover. The rationale behind this approach is that, within regions of the same land cover, the pixels have stationary statistics and are characterized by mostly linear dependency, contrary to what usually happens for unsegmented images. Therefore, by applying conventional transform coding techniques to homogeneous groups of pixels, the proposed algorithm is able to effectively exploit the statistical redundancy of the image, thereby improving the rate distortion performance. The proposed coding strategy consists of three main steps. First, each pixel is classified by vector quantizing its spectral response vector, so that both a reliable classification and a minimum distortion encoding of each vector are obtained. Then, the classification map is entropy encoded and sent as side information, Finally, the residual vectors are grouped according to their classes and undergo Karhunen-Loeve transforming in the spectral domain and discrete cosine transforming in the spatial domain. Numerical experiments on a six-band thematic mapper image show that the proposed technique outperforms the conventional transform coding technique by 1 to 2 dB at all rates of interest.     

用于多媒体广播的正交频分复用模块集成系统

在０．１８μｍ工艺或更小的线宽下,可把各单个子系统有效地集成在同一块芯片上。通过讨论新工艺条件下具有语音视频图像处理功能的高速编码正效频分复用调制（ＯＦＤＭ）集成系统设计的可能方案,并剖析实例,对其离散余弦变换、快速傅里叶变换及前向纠错编码进行了分析。     

Psychophysically derived quantisation model for efficient DWT imagecoding

An efficient method for the optimum quantisation of the luminance and chrominance components for discrete wavelet transform (DWT) still image coding, based on current psychophysical evidence, is described. Its practical application to image compression is presented. This new approach is further supported by experimental work which compares perceived DWT image quality with conventional JPEG-encoding, that is block-based discrete cosine transform (DCT) coding, on the basis of quantisation distortion     

Analysis and application of digital spectral warping in analog and mixed-signal testing

Spectral warping is a digital signal processing transform which shifts the frequencies contained within a signal along the frequency axis. The Fourier transform coefficients of a warped signal correspond to frequency-domain 'samples' of the original signal which are unevenly spaced along the frequency axis. This property allows the technique to be efficiently used for DSP-based analog and mixed-signal testing. The analysis and application of spectral warping for test signal generation, response analysis, filter design, frequency response evaluation, etc. are discussed in this paper along with examples of the software and hardware implementation.     

Blind compensation of nonlinear distortion for bandlimited signals

Nonlinear distortion of bandlimited signals results in spectral spreading. This paper develops a blind nonlinear compensation method for bandlimited signals by suppressing the spectral content of the distorted signal above the original signal bandwidth by means of adaptive nonlinear filtering. The nonlinear compensator is constructed using a power series filter with adaptive coefficients. The adaptive coefficients are identified blindly by applying a least-squares criterion to the out-of-band spectral content of the nonlinear compensator output. The extraction of the out-of-band signal is efficiently performed by the discrete cosine transform. The effectiveness of the blind nonlinear compensation method is demonstrated by way of simulation examples involving periodic, colored noise, and bandlimited speech signals.     

变换编码中直流偏移的快速处理

根据变换编辑码的持点,对变换编码中直流偏移处理提出一种快速算法。该算法通过交流常规的直流偏移和变换的顺序,可大大减少直流偏移的运算次数,余弦变换编码器的每个编码块的直流偏移运算次数降低至一次,小波变换变换块后的真流运算降低至分析增益为1的子带大小。例如对8×的DCT块直流偏移次数只需1次,对采用MALLAT方式的L层分解的二维图像,直流偏移次数只有原来的1/4L.     

基于Matlab的数字水印系统的GUI设计

数字水印技术为数字信息安全和版权提供了一种有效的保护措施。本文通过小波变换和离散余弦变换的方法,首先对载体彩色图像进行小波变换,然后对变换后载体图像的低频子带再进行离散余弦变换,最后将彩色水印图像嵌入到离散变换的直流系数上。结果证明,在抗噪声攻击、JPEG压缩、小幅度剪切方面利用Matlab软件构建的数字水印系统可达到效果良好,稳健性强。     

基于DCT的自适应多重彩色图像盲水印算法

针对现有图像水印算法的一些不足,提出了一种基于离散余弦变换的自适应多重彩色图像盲水印算法。该算法根据彩色载体图像内容特点,将预处理后的不同二值图像水印自适应地嵌入到其绿色分量的离散余弦变换低频系数和中频系数,实现了不同水印在彩色载体图像中的嵌入及盲提取。实验结果表明,该算法在保证不可觉察性的前提下能够较好地抵抗各类常规攻击,具有良好的稳健性。     

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.