数字信号基于小波变换的码元符号率检测

数字信号在码元边界上由于幅度、频率或相位的变化而产生变化。小波变换对信号的跳变点和奇异点有很好的检测能力。利用小波变换的模极大值方法可以对码元的跳变点进行检测,通过分析码元的跳变点进而可分析出码元的符号率。计算机仿真结果表明,该方法对码元的跳变点检测具有很高的精确性,能够比较准确地识别出码元符号率。     

带限MPSK信号码元宽度的盲估计

理想MPSK信号在码元跳变处表现为相位的不连续性。小波变换能有效检测出这些信号细节,从而为估计码元宽度提供了依据。实际系统常采用脉冲成形技术减小发射带宽,改变了信号包络,从而影响了小波变换的检测性能。本文对截获接收机输出的信号进行预处理后再利用基于Haar小波变换的算法提取码元宽度信息,理论分析和实际信号处理表明,这种算法能有效检测码元跳变,从而估计出码元宽度。     

一种基于小波分析的码元速率估计方法

针对实际系统中MPSK信号在码元跳变处包络幅度降低,相位跳变不明显,常规算法检测效果不佳的问题。文中提出了一种基于小波变换对MPSK信号进行码元速率估计的新方法。对截获信号进行预处理以减弱跳变处的影响并提高估计的精度。理论分析和实验结果表明,这种算法能有效估计出码元速率,有很高的工程应用价值。     

QPSK信号码元速率估计

接收端码元速率的估计,有助于获取信号调制规律,便于信号正确解调。以π／4-QPSK为例,在解决小波变换尺度优化选取和基带快速提取的难点的同时,旨在运用小波变换估计信号码元速率。具体方法是利用QPSK相位跳变发生在码元周期交替处的特点和Haar小波的相位特点,使小波变换系数极大模值出现在码元速率整数倍处;根据极大模值构造出同频率冲激序列,从其功率谱图上估计码元速率。经Matlab仿真,的确能估计出信号码元速率为500B。该方法所得估计值受噪声变化影响甚微,且在一定范围内小波变换尺度越大,分析效果越好。     

QAM信号的波特率估计

将传统的信号处理工具———傅里叶变换与小波变换相结合,提出了一种QAM信号波特率提取的新方法。对QAM信号作2次连续小波变换,其小波变换的结果在码元跳变点处出现明显的峰值,其他处都为零。然后再作FFT,则一次谐波谱线(即第2根峰值谱线)的位置恰好位于f=fb处,因此,通过检测第2根峰值谱线位置即可估计出QAM信号的波特率。对高斯白噪声中QAM信号的波特率提取进行了仿真并给出了部分实验结果。     

基于小波变换的MSK信号码速率盲估计

针对最小频移键控调制信号的码速率估计问题,提出一种基于Haar小波变换的MSK信号码速率盲估计方法。首先对接收信号作傅里叶变换得到信号频谱,对频谱频点分析粗估计信号的码速率,接着通过粗估计的码速率选取短时傅里叶变换窗函数长度和3个小波尺度,利用短时傅里叶变换得到信号瞬时频率变化,再利用小波的边缘检测特性对信号瞬时频率序列相位跳变点检测,最后对检测结果作频谱分析,估计频率得到MSK信号的码速率。仿真结果表明,高于信噪比门限时本算法可以对MSK信号码速率有效估计。     

基于小波变换的跳频信号参数盲估计

本文提出了一种基于小波变换的未知跳频信号参数盲估计方法.该方法利用小波变换在时频域上良好的局部化性质和检测突变点能力,在未知任何先验参数的情况下,能够准确估计出跳频信号的跳变时刻、跳频频率和跳频周期等参数.文中阐述了应用小波变换对跳频信号分析的基本原理,给出了估计跳频信号参数的具体算法步骤,在计算机仿真的基础上对结果进行了性能分析,得到了较为准确的估计结果.     

一种基于小波变换的码元速率盲估计方法

小波变换对暂态信号具有较强的检测能力，根据这一性质可以通过提取数字信号码元变化处的暂态信息来估计码元速率。目前的算法存在尺度盲点、抗噪声性能低等弱点，本文针对这些问题作了分析和改进，提出了一种通过对信号基带形式进行二次小波变换来提取码元速率的估计方法，不仅能克服了目前算法的不足，而且所需要先验知识少，能达到码元速率的盲估计。仿真实验表明了该算法较其它算法具有较高的精确度和抗噪性能。     

基于小波变换的时间间隔检测

本文提出一种基于小波变换的检测数字信号码元间隔的方法。该方法利用基小波的伸缩与时移特性获得检测所需的时、频分辨率,通过对信号做多尺度的连续小波变换,可以在较低信噪比下很好地检测到信号频率并得到边沿过渡较快的相应小波变换时谱。根据小波分析与奇异性信号检测的理论,本文还提出对小波变换时谱的二次处理方法。理论分析和计算机模拟结果表明,本方法是有效的,是提高时间间隔检测性能的一种新途径。     

基于小波变换的时间间隔检测

本文提出一种基于小波变换的检测数字信号码元间隔的方法。该方法利用基小波的伸缩与时移特性获得检测所需的时、频分辨率,通过对信号做多尺度的连续小波变换,可以在较低信噪比下很好地检测到信号频率并得到边沿过渡较快的相应小波变换时谱。根据小波分析与奇异性信号检测的理论,本文还提出对小波变换时谱的二次处理方法。理论分析和计算机模拟结果表明,本方法是有效的,是提高时间间隔检测性能的一种新途径。     

Comparison Between DPCM and Hadamard Transform Coding in the Composite Coding of the NTSC Color TV Signal

Since differential-pulse-code modulation (DPCM) and orthogonal transform coding (OTC) are the most fundamental methods of high-efficiency coding (bit-reduction method), it is important to clarify the basic coding characteristics of these methods and the difference between them in order to utilize the bit-reduction method effectively. This paper theoretically as well as experimentally compares the coding efficiency of a DPCM having a two-dimensional predictor with that of a two-dimensional Hadamard transform coding method (HTC) in the intrafield coding of the NTSC composite signal. The comparison evidenced that the distinctive difference in coding characteristics between DPCM and HTC depends greatly on the power level of carrier chrominance signals. That is, it is confirmed theoretically and experimentally that the coding efficiency of the HTC is far lower than that of the DPCM in the case of a signal having a high power level carrier chrominance signal such as a color-bar signal.     

Theoretical Comparison Between DPCM and Transform Coding Regarding the Robustness of Coding Performance for Variation of Picture Statistics

Since DPCM and transform coding are two fundamental approaches to high-efficiency (bit reduction) coding, it is important to clarify the basic coding characteristics of these approaches and the differences between them in order to utilize the high-efficiency coding method effectively. It is important to compare them not only from the standpoint of coding performance as optimized coding schemes based on the statistics of the input picture signal, but also from that of the robustness of coding performance for the variation of picture statistics to be coded. This paper theoretically compares the robustness of the coding performance of DPCM having a two-dimensional predictor with that of a two-dimensional Hadamard transform coding in an intrafield coding method of the NTSC composite signal. The comparison provides theoretical evidence that transform coding is more stable than DPCM, and this tendency is marked at lower bit rates such as 1 or 2 bits/pel, while DPCM has a higher coding performance for pictures with high autocorrelation.     

多小波图像编码中前置滤波器的设计

本文研究多小波图像编码中前置滤波器的设计,多小波变换矢得滤波,普通的标量信号要通过一个前置滤波器转化为一信号,才能进行我小波变换。本文根据图像信号的特点,结合现有的二种前置滤波器,提出了一种新的置滤波方程。实验数据表明,这种新的前置滤波器优于现有的二种前置滤波器,因而为提高图像压缩比例创造了条件。     

Transform methods for seismic data compression

The authors consider the development and evaluation of transform coding algorithms for the storage of seismic signals. Transform coding algorithms are developed using the discrete Fourier transform (DFT), the discrete cosine transform (DCT), the Walsh-Hadamard transform (WHT), and the Karhunen-Loeve transform (KLT). These are evaluated and compared to a linear predictive coding algorithm for data rates ranging from 150 to 550 bit/s. The results reveal that sinusoidal transforms are well-suited for robust, low-rate seismic signal representation. In particular, it is shown that a DCT coding scheme reproduces faithfully the seismic waveform at approximately one-third of the original rate     

Hybrid picture coding with wavelet transform and overlappedmotion-compensated interframe prediction coding

A novel video coding scheme using an orthonormal wavelet transform is proposed. The wavelet transform is used in a motion compensated interframe coder in which a blockless motion compensation technique is employed to increase efficiency of wavelet transform coding. A new scanning method for wavelet coefficients is also proposed which is rather different from subband coding. Simulation work is carried out to evaluate the proposed coding method. Significant improvement in subjective quality is obtained over that obtained with conventional hybrid coding methods that use blockwise motion compensation and DCT. Some improvement has also been realized in the signal to noise ratio. Although wavelet coding is still in its early stages of development, it appears to hold great promise for motion picture coding     

A 128 Kb/s Hi-Fi audio codec based on adaptive transform codingwith adaptive block size MDCT

A Hi-Fi audio codec with an improved adaptive transform coding (ATC) algorithm is presented using digital signal processors (DSPs). An audio signal with a 20 kHz bandwidth sampled at 48 kHz is coded at a rate of 128 kb/s. The algorithm utilizes adaptive block size selection, which is effective for preecho suppression. A modified discrete cosine transform (MDCT) with a simple window set is employed to reduce block boundary noise without decreasing the performance of transform coding. In addition, a fast MDCT calculation algorithm, based on a fast Fourier transform, is adopted. Weighted bit allocation is employed to quantize the transformed coefficients. The codec was realized by a multiprocessor system composed of newly developed DSP boards. Subjective tests with the codec show that the coding quality is comparable to that of compact disc signals     

Multichannel ECG data compression by multirate signal processingand transform domain coding techniques

A multilead electrocardiography (ECG) data compression method is presented. First, a linear transform is applied to the standard ECG lead signals, which are highly correlated with each other. In this way a set of uncorrelated transform domain signals is obtained. Then, the resulting transform domain signals are compressed using various coding methods, including multirate signal processing and transform domain coding techniques     

On adaptive vector transform quantization for speech coding

Adaptive vector transform quantization (AVTQ) as a coding system is discussed. The optimal bit assignment is derived based on vector quantization asymptotic theory for different PDFs (probability density functions) of the transform coefficients. Strategies for shaping the quantization noise spectrum and for adapting the bit assignment to the changes in the speech statistics are discussed. A good estimate of the efficiency of any coding system is given by the system coding gain over scalar PCM (pulse code modulation). Based on the optimal bit allocation, the coding gain of the vector transform quantization (VTQ) system operating on a stationary input signal is derived. The VTQ coding gain demonstrates a significant advantage of vector quantization over scalar quantization within the framework of transform coding. System simulation results are presented for a first-order Gauss-Markov process and for typical speech waveforms. The results of fixed and adaptive systems are compared for speech input. Also, the AVTQ results are compared to known scalar speech coding systems     

小波包变换在水声信号压缩编码中的应用

应用小波包变换及Huffman编码技术相结合的方法对水声信号进行了压缩编码。先对水声信号进行小波包分解;然后对分解系数进行阈值处理,最后对阈值后的系数进行Huffman编码。使用了两种阈值方案,通过仿真比较．其中不同频段不同阈值方案的零率比全局阈值的零率高。尽管全局阈值的零率稍低一些,但其Huffman编码效率较高且硬件实现较为简单,因此最后选取全局阈值进行闽值量化。仿真结果表明,本算法对水声信号压缩编码效果理想。