利用DirectSound实现音频的实时捕捉和播放

对音频的实时捕捉和播放是网络多媒体传输的基础。本文研究了Visual C 6．0下的DirectSound类．并给出了基于此类在局域网不增加设备的情况下对音频进行实时捕捉和播放的实现。     

基于DirectSound的飞行模拟器音响系统实现

DirectSound可以实现复杂的混音播放。根据某型飞机飞行模拟器的研制需求,提出了利用DirectSound技术实现音响系统的方法．给出了两个混音播放类和音响系统的程序编制方法。应用实践证明,基于DirectSound技术的飞行模拟器音响系统,编程简单,模拟效果好,具有普遍的应用性。     

DirectSound封装技术及其音频捕捉应用

DirectX为多媒体应用程序的开发提供了统一的编程接口，但十分繁琐。基于代码重用的原则，本文提出了一种二次封装技术，并利用ActiveX控件实现了这种技术，现已在远程网络教育中成功得到应用。     

利用DirectSound实现音频的流式播放控制

Directsound是微软DirectX多媒体的重要组成部分,提供了大量的音频处理API函数.它提供了快速的混音、声音捕获、播放、硬件加速等功能,常被用来实现声音的播放功能.与其他实现声音播放的方式相比,DirectSound提供了独特的通知机制,利用该方法,可以在播放音频时,实现音频与其他事件的精确同步.对DirectSound的相关概念及播放机制进行了介绍,在此基础上,封装了一个声音的播放类,实现对流式数据的播放及消息触发,从而降低其开发难度和工作量.     

基于Python语言的音频捕获及频谱分析程序设计

Python编程语言具有极高的开发效率,DirectSound提供了强大高效的声音处理功能。文章介绍了基于Python编程语言和DirectSound的音频数据捕获技术,并对采集到的音频数据进行了相关分析处理,实时显示其波形和频谱。     

基于Python语言的音频捕获及频谱分析设计

Python编程语言具有极高的开发效率,DirectSound提供了强大高效的声音处理功能。文中利用Python编程语言和DirectSound的音频数据捕获技术,并对采集到的音频数据进行了相关分析处理,实时显示其波形和频谱。     

基于Python语言的音频捕获及频谱分析程序设计

Python编程语言具有极高的开发效率,DirectSound提供了强大高效的声音处理功能。论文介绍了基于Python编程语言和DirectSound的音频数据捕获技术,并对采集到的音频数据进行了相关分析处理,实时显示其波形和频谱。     

基于数字语音教室的多路混音算法及应用

在研究音频混音技术的基础上,结合基于嵌入式技术的数字语音教室的实际情况,提出了一种应用于数字语音教室的多路实时混音算法,并给出了在数字语音教室中利用多路混音算法实现多人实时音频交互的方法.     

基于数字语音教室的多路混音算法及应用

在研究音频混音技术的基础上,结合基于嵌入式技术的数字语音教室的实际情况,提出了一种应用于数字语音教室的多路实时混音算法,并给出了在数字语音教室中利用多路混音算法实现多人实时音频交互的方法。     

多路数字混音方案设计

混音技术是多媒体会议系统中的一个重要环节,极大影响了各用户之间的相互交流.通过对现有算法的总结分析,得出了参与混音的路数和各路语音强度大小是影响混音效果的主要因素.针对上述两个主要问题,提出了一种新的混音模型,在硬件设计上采用多路分组并行处理方式,限制最大混音的路数,并通过自动增益控制电路,控制输入语音信号强度;在软件设计上采用自动门限检测技术和改进的自对齐算法,减少了噪声的引入,避免了溢出检测问题,混音效果自然流畅.可以广泛应用于大规模多媒体会议系统中.     

基于DirectSound的三维声音的仿真设计及实现

三维声音仿真是虚拟环境的一个重要组成部分。分析了利用D irectSound实现在虚拟环境中三维声音实时仿真的关键技术,并通过V isual C++6.0平台编程实现三维声音效果的播放。     

基于DirectSound的实时音频采集与声阵列定位系统研究

DireetSotmd提供了强大的音频处理功能,使得开发者在不需要知道声卡硬件细节的前提下开发出高性能的多媒体程序.本文结合笔者的开发实例,详细阐述了使用DirectSound实现实时音频采集的方法,并设计实现了基于时延估计算法的声阵列定位系统.     

利用PC机集群实现声频工程中复杂音频处理

研究利用普通PC机（而不是DSP）构建数字音频虚拟仪器的技术,以满足声频工程中复杂的数字音频处理要求。提出了一个基于PC机集群的分布式解决方案,设计实现了相关的软件,并对关键技术进行了测试。     

嵌入式音频系统的设计与实现

音频系统在嵌入式多媒体系统中占有很重要的地位。主要是利用双核心的Blackfin 561处理器的特点,设计了一种适合于此处理器的音频系统硬件解决方案。同时结合对嵌入式音频系统主要问题的分析,得到了一种针对此硬件架构的软件整体解决方案。在软件方面通过设置合理的缓冲区以及采取静音检测、同步等措施,实现了该应用在嵌入式μCLinux系统下的驱动程序,为嵌入式μCLinux系统下的芯片驱动开发提供了很好的借鉴。     

The impact of status and audio conferencing technology on business meetings

This field study examining the effects of organizational status and multimedia audio communications technology on communication patterns in business meetings confirms that high status group members verbally dominate discussions and have more control over the flow of the proceedings. However, it reveals a new and surprising finding: multimedia communications technology can in fact exaggerate status constraints in contrast to findings that there is no effect or an equalizing effect on status inequalities of text-based conferencing technology. It appears that in audio conferences, the lack of non-verbal cues that can aid turn-taking combined with (1) the participants' knowledge of the group's status hierarchy and (2) the tendency to compare oneself unfavourably to those of higher status, makes it more difficult for lower status individuals to contribute verbally to discussions than in face-to-face interactions. Such status constraints may have both positive and negative impacts on group communication effectiveness, these and implications for the design and implementation of multimedia communications technology are discussed.     

基于残差网络和随机森林的音频识别方法

环境声音分类(ESC)是音频处理领域的重要分支之一,在未来多媒体应用中有重要的作用。音频识别是提取音频中特定的声学特性,将音频分类至样本对应的正确场景,有助于感知和理解周围环境。现阶段音频识别主要是通过信号处理技术和机器学习方法达成。随着人工智能飞速发展,传统的音频处理技术以及机器学习方法面临着巨大的挑战,ESC的识别准确性有待进一步提高。结合残差网络和随机森林两种方法,将一维时域信号的音频数据转换为二维数据形式的梅尔声谱图,预训练残差网络获得一个精度较高的网络模型作为特征提取器,利用该网络模型提取音频中的深层特征,再利用随机森林对深层特征进行分类。该方法在ESC任务上识别率提升了近10%,取得了较好的分类结果。     

From broadcast television to Internet audio/video: techniques and tools for VCR‐style interactivity

One of the new applications evolving in the Internet is streaming audio/video. A major reason for its growing popularity is interest in the compelling new services that become possible. Prototype services are being developed which are new to the Internet but offer the same look, feel, and functionality that have traditionally only been found in services delivered via other communication medium, e.g. broadcast television. In addition, the Internet is evolving to offer ‘value‐added’ services, like streaming audio/video with VCR‐style interactivity and embedded hyperlinks. We are poised both on seeing the development of new paradigms for interacting with audio/video, and on seeing the merging of broadcast television and Internet‐based broadcasts. Before this process can be considered successful, a number of technical challenges, derived from the various ways in which content is physically delivered, must be solved. In this paper, we focus on the value‐added service of VCR interactivity. VCR interactivity has long been a challenge for both broadcast television and streamed Internet audio/video. The challenge is how to provide individualized playout for content being streamed to a large group of users using one‐to‐many delivery. While some new companies are starting to offer devices which provide this kind of service for broadcast television, there are still numerous technical challenges for the Internet‐based version of a similar service. This paper has a three‐fold objective. First, we describe the types of services available in the traditional broadcast infrastructure and compare these to the types of services that are deployed or possible in Internet‐based services. Second, we describe our attempts to implement some of the more challenging and novel service types. In particular, we examine client‐based control of programs streamed over the Internet to tens, thousands, or even millions of users. Finally, we discuss the impact of these services on the protocols and applications used to support Internet‐based, multi‐party conferencing. Copyright © 2001 John Wiley & Sons, Ltd.     

Toward bandwidth-efficient and error-robust audio streaming over lossy packet networks

Bandwidth efficiency and error robustness are essential issues for different multimedia streaming applications. This paper presents strategies for high-quality audio streaming based on fragmenting perceptually coded audio frames and shuffling the data components among multiple packets for transportation. This is done to increase robustness against packet loss. We also address the delivery of audio data consisting of components with different proportional priorities. Our approach is rationalized with streaming tests using the MPEG AAC audio codec in a simulated network environment and formal listening tests to evaluate the resulting audio output. According to the results, the proposed schemes improve audio quality significantly with reasonable increase to network resource utilization compared to traditional error robustness measures.     

一种利用声音识别的森林盗伐检测方法

针对目前森林盗伐猖獗,且尚无实时检测盗伐行为方法问题,提出了一种基于声音识别的森林盗伐检测方法。通过对声音信号的频谱特征分析、相似度值及信噪比计算,检测是否存在链锯伐木行为。实验结果表明,提出的方法能够有效地排除干扰声音,准确、实时地识别链锯伐木声音。     

基于发音特征的音/视频双流语音识别模型

构建了一种基于发音特征的音/视频双流动态贝叶斯网络(dynamic Bayesian network, DBN)语音识别模型,定义了各节点的条件概率关系,以及发音特征之间的异步约束关系,最后在音/视频连接数字语音数据库上进行了语音识别实验,并与音频单流、视频单流DBN模型比较了在不同信噪比情况下的识别效果。结果表明,在低信噪比情况下,基于发音特征的音/视频双流语音识别模型表现出最好的识别性能,而且随着噪声的增加,其识别率下降的趋势比较平缓,表明该模型对噪声具有很强的鲁棒性,更适用于低信噪比环境下的语音识别