AC—2广播与存贮的高质量数字音频编码器

本文介绍了美国杜比实验室研制的适用于广播的宽带数字音频变换编码器系统。编码器的比特率每信道192～64kbit/s,它能提供不同的比特率、编码延时及音频带宽三者折衷的方案。还重点介绍了新一代音频编码器,这种编码器在比特率压缩过程中应用了人耳在频域和时域的掩蔽效应,使用了新型滤波器簇,能根据信号的时间特性和频率特性,动态调整频率与时间分辨率。还介绍了编码器系统现在和将来的应用。     

行车噪声环境下基于人耳频率选择特性的声学特征提取方法

本文提出了一种基于加权Mel滤波器组的声学特征提取方法。该方法通过提取音频信号中的共振峰信息,使用动态自适应方法对中高频部分的Mel滤波器组进行加权,从而模仿人耳覆膜的频率选择映射。相比较于传统的MFCC特征,更适用于行车噪声环境下的快速声学事件检测任务;弥补了传统的Mel滤波器组高频部分分辨率低,从而导致对噪声鲁棒性较差的问题。实验结果表明:在信噪比较低的行车环境中,该特征有助于提高声学事件的检出率。     

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。     

噪声环境中的双声道音频回波抵消模型构建

为提升远端说话人员的分辨率，保证真实空间化听觉感受效果，构建一种噪声环境中的双声道音频回波抵消模型。该模型以子带自适应滤波器组为核心分解音频信号，使其形成若干个子频带信号；在子带域内计算噪声干扰阈值，依据阈值的计算结果并结合人耳听觉特性，叠加独立分布白噪声，实现各个子带的回波抵消。在此基础上，引入步长补偿因子和反馈控制理念改进最小均方算法，优化滤波器组参数，提升双声道音频回波抵消效果。测试结果表明：所设计模型可有效处理音频信号中的噪声信号，抵消后的音频信号信噪比均在3.5 dB以上，音频信号的振幅误差均在0.2 V以下，能够提升远端人员说话的分辨率，保证音频质量。     

用于音频功放的数字上采样滤波器设计

设计了一种用于音频功放的数字上采样滤波器,通过可配置的多级插值系统将多种采样率(44.1 kHz、48.0 kHz、88.2 kHz、96.0 kHz、176.4 kH、192 kHz)的音频信号采样频率提高到6.144/5.644 8 MHz,解决了目前功放芯片只能兼容某种单一采样频率信号的问题,极大提高了音频功放的性能和音频功放芯片的灵活性。通过对半带滤波器算法结构研究,依据半带滤波器的特点,在不损失性能的前提下改进滤波器结构,使半带滤波器的资源消耗减少3/4。最终经Matlab和Modelism仿真验证分析,上采样率器带内纹波损失低于0.03 dB,各级滤波器阻带衰减-80 dB左右,滤波器最大群延时为2.052 ms,输出波形光滑,满足实际使用需求。     

一种新的宽带信号正交分量的解调方法

在D通道均匀DFT分析滤波器组基础上给出了一种新的信号正交分量的解调结构 ,为解调宽带信号 (B fc)的正交分量 ,系统采用两组D通道均匀DFT分析滤波器组通带互补方法 ,以克服单组滤波器中滤波器的非理想性所造成的信号失真。在频域中当信号频谱处于两个或更多个滤波器通道时 ,通过频谱组合办法直接得到实信号的正半谱或负半谱。在载波频率已恢复的情况下给出了IQ正交分量的解调算法。仿真证明该系统结构和算法的正确性和高效性     

采用数字滤波器的音频系统的频域均衡(二)--均衡实例

4.2FIR部分数字反折叠和重现滤波器 -- 根据实测频响进行设计 部分数字反折叠和重现FIR滤波器被设计成用于时域和频域中的线性系统参数建模应用的音频数据采集和测试信号发生的组成部分.     

环移S＆G自适应滤波器及其在交叠的周期性生物电信号分离中的应用

本文提出了一种适用于交叠周期信号的周期检测和信号分离的环移Ｓｔｏｐ＆Ｇｏ自适应数字滤波器，该滤波器对周期信号在频域的交叠程序没有任何要求，除假设书籍要分离的周期信号的周期的大致范围外，不需其他先验知识，因而它提供了一种分离时域和频域都互相交叠的周期信号的有效可行的方法，文章从理论讨论了该滤波器的收敛条件和各参数的选取要求，并提出了一种较实用的粗精度周期搜索和通道自动关闭算法，减少了计算量，实验证明     

数字式单边带调制器

本装置可用于产生数字化信息音并将瞬时信息音频转换到适用于通讯的频谱位置上。为响应输入的数字数据,产生了分配频率和持续时间明确的不连续音频,并经济地组合为无假信号的单边带抑制载波信号。该信号的频率位置可方便地采用常用的混频方法确定。     

IBM PC通用音频数据采集系统

本文介绍一种适用于IBM PC及其兼容机的通用音频数据采集系统。该系统具有8个A/D通道和一个D/A通道,分辨率为12bit,带有一个可编程的7阶椭圆抗混叠滤波器。系统具有功能强,通用性好,电路简单,不占用PC机中断和DMA资源等优点。能广泛用于音频及其他测量与控制领域。     

An Improved Detection Technique for Spread Spectrum Audio Watermarking with a Spectral Envelope Filter

We propose an improved algorithm for detecting audio watermarks based on a spread spectrum in the spectral domain. Since the energy of a watermark is much smaller than that of the cover audio data, pre‐processing to reduce the effect of the cover data is needed to reliably extract watermarks. We introduce a spectral envelope filter as a preprocess that enhances detecting performance by filtering out the intrinsic spectral character of cover data. The proposed watermarking structure can be easily included in the compression system and can extract watermarks from partially decompressed spectral data. Our experimental results demonstrate that with a bit error rate of around 10 dB against general attacks, the proposed detecting scheme works better than detectors without the spectral filter.     

Perceptual Audio Coding Using Sinusoidal/Optimum Wavelet Representation

A perceptual audio coder, in which each audio segment is adaptively analyzed using either a sinusoidal or an optimum wavelet basis according to the time-varying characteristics of the audio signals, has been constructed. The basis optimization is achieved by a novel switched filter bank scheme, which switches between a uniform filter bank structure (discrete cosine transform) and a non-uniform filter bank structure (discrete wavelet transform). A major artifact of the International ISO/Moving Pictures Experts Group (MPEG) audio coding standard (MPEG-I layers 1 and 2) known as pre-echo distortion which uses a uniform filter bank structure for audio signal analysis, is almost eliminated in the proposed coder. A perceptual masking model implemented using a high-resolution wavelet packet filter bank with 27 subbands, closely mimicking the critical bands of the human auditory system, is employed in this audio coder. The resulting scheme is a variable bit-rate audio coder, which provides compression ratios comparable to MPEG-I layers 1 and 2 with almost transparent quality.     

基于频谱减法的语音去噪算法研究

语音增强技术是音频信号处理中的重要部分,频谱减法是目前在语音增强技术中最常用的方法之一。针对传统频谱减法会产生音乐噪声并无法消除音乐噪声的不足之处及高频噪声干扰比较严重的情况下频谱减法效果差的情况,采用了在频谱减法之后进行LMS滤波以降低音乐噪声对语音质量的影响和低通滤波以滤除脉冲干扰。根据仿真结果表明,改进扩展频谱减法能够有效降低音乐噪声和尖锐的高频兹兹声,从而提高信噪比,达到语音增强的目的。     

基于奇异值分解的DCT域音频数字水印算法

提出了一种基于DCT直流分量的音频数字水印算法。算法根据音频信号在DCT域能量较大处可以隐藏较多信息的水印嵌入原则,结合音频信号在DCT域能量分布特征,首先把原音频信号先进行分段处理,然后对每段做DCT变换,找出水印嵌入点,实现了在一段音频信号中隐藏一幅二值图像的奇异值向量。实验结果表明,该算法对MP3压缩、低通滤波、重采样等恶性攻击具有较强的鲁棒性和不可感知性。     

基于傅里叶变换的双音多频信号识别实验设计

傅里叶变换是建立信号时间域与频率域之间的一种变换关系。针对傅里叶变换的学习难点,分析了不同类型信号的傅里叶变换关系。结合工程应用,设计了电话拨号的双音多频音频信号按键识别综合实验,通过信号分段、滤波处理和快速傅里叶变换算法提取信号的特征频率,实现了双音多频音频信号的按键号码识别功能。     

Single channel nonstationary stochastic signal separation using linear time-varying filters

Separability of signal mixtures given only one mixture observation is defined as the identification of the accuracy to which the signals can be separated. The paper shows that when signals are separated using the generalized Wiener filter, the degree of separability can be deduced from the signal structure. To identify this structure, the processes are represented on an general spectral domain, and a sufficient solution to the Wiener filter is obtained. The filter is composed of a term independent of the signal values, corresponding to regions in the spectral domain where the desired signal components are not distorted by interfering noise components, and a term dependent on the signal correlations, corresponding to the region where components overlap. An example of determining perfect separability of modulated random signals is given with application in radar and speech processing.     

分频段数字音频动态范围处理器的设计、仿真与测试

本文提出一种分频段数字音频动态范围处理器的设计方法。传统的分频段方法主要采用滤波器组,而本文则利用扬声系统中的LR分割器来分割频段。传统的动态范围处理器通常利用前馈包络法和低通滤波器来平滑增益曲线。但面对快速变化的信号或噪声,此类方法得到的结果可能会有较大的失真。针对此问题,本文提出了一种创新的时间幅度并行控制法,在时间和幅度域内同时对信号进行控制,能很好地降低噪声并改善输出信号的失真度。最后,本文给出了MATLAB仿真和基于ADSP21364的实测结果,验证了本方法的性能。      

On the synthesis of digital transmit filters

A simple method for calculating the coefficients of a digital transversal filter when cascaded with analog filters is proposed. The optimization criterion takes into account both the spectral characteristics of the signal and the residual amount of intersymbol interference at the decision device input. In contrast with standard methods no a priori constraints are applied to the filter in the time or in the frequency domain     

基于匹配滤波的脆弱数字音频水印盲检测算法

根据原始水印构造匹配滤波器,并结合小波估计方法,提出脆弱音频水印的盲检测算法。在假定嵌入水印为噪声的情况下,利用小波不需要原始信号辅助信息即可估计该信号的特点,首先根据给定的估计门限估计了加入水印后的原始信号,从而实现了水印与音频信号分离。然后将水印与估计产生的噪声构成的混合信号送入构造的匹配滤波器作相关运算,根据滤波器的输出响应判断水印及载体是否受到攻击。实验结果表明,所提出方法对噪声、有损压缩、滤波、随机插入与删除等操作具有较高的敏感性。     

Reconfigurable filter bank structures for low complexity digital channelizer using fractional interpolation and MFIR filters with cosine modulation

A low complexity digital reconfigurable filter bank structure which is suitable for digital channelizers in software defined radio is proposed in this paper. In order to get a sharp transition width with low hardware complexity, multiplicative finite impulse response (MFIR) filter is used as the prototype filter in the proposed structure. A masking stage and a channel generation stage constitute the two stages of the proposed structure. In the masking stage, 5 sub-bands are generated by performing cosine modulation technique on the prototype filter. In the channel generation stage, different channels are generated by doing spectral addition/subtraction operations on the interpolated versions of the MFIR filter. Hence, the basic techniques used in the proposed structure are, cosine modulation and fractional interpolation technique. By adjusting a 3-bit control signal, different regions and channels, generated by the two stages, can be obtained. The proposed structure has a hardware complexity, which is much lesser than that of the state-of-the art techniques used for SDR channelizers. The number of channels generated using the proposed structure is also more. The main advantage of the proposed structure is that the hardware complexity of the proposed structure remains the same even if the number of channels is increased.