audio streaming

• Mbns will soon offer high - speed internet access , video and audio streaming , as well as stock trading , home banking and distance learning applications
  稍后mbns将供应高速互联网上网服务、视讯和音讯码流、股票交易、家居银行及遥距课程。
• Furthermore , on the analysis of the psychoacoustic models , some simplification can be made in the psychoacoustic models . the purpose of this simplification is to reduce the bit rate of audio stream
  此外，基于对心理声学模型的分析，对心理声学模型作若干近似，简化的目的是为了用较少的比特编码音频数据。
• Multimedia stream , such as video , audio stream on internet have three main characteristics : first , need of higher bandwidth ; second , a higher demand on real - time ; third , usually bursting
  在网络上传输多媒体数据，像语音和视频流，具有三个方面的主要特征，一是需要很高的带宽，二是对实时性要求很高，三是数据往往都是突发的。
• In an oversimplified description , the speech recognition system processes and cleans up the audio and mathematically processes and matches the audio with language and acoustic models to turn the audio stream into a set of phonemes
  简而言之，语音识别系统进行处理并整理音频，然后进行数字处理并把音频与语言和声音模型匹配起来以便把音频流转为一组音素。
• Firstly , the international standards and key techniques of video conference are analyzed , as well as the advantages and disadvantages of present products . secondly , mpvc , an ip - based multipoint videoconference system model , is given by theories and technologies of videoconference . finally , the thesis presents complete design scheme and technique route , and implements multipoint real - time transportation of video / audio stream
  论文选择设计、实现全互连的多媒体视频会议系统为研究对象，通过对视频会议国际标准、关键技术、产品的研究分析以后，在原有课题“多媒体合成作战指挥支持系统”的基础上，提出了一种基于ip网络的多点视频会议系统mpvc ( multipointvideoconference )模型，给出了完整的设计方案和技术途径，实现了视频/音频流多点实时传输。
• It also designs the mechanism of realizing audio / video conference . in java runtime environment , a better method can be taken , this is to say , we expand the conception of jmfs " multiplexior " , getting the " audio mixer " object to manage the audio stream and getting the " video mixer " object and the " token click and preempt " control mechanism object to manage the video stream . at last this solution method gives afford the facts below : the quantities of persons who join the audio collaborate do not been restricted and people do not need a voice authority before they speak
  本文还对音频视频协作会议内部实现机制进行了重新设计改进，给出一种针对java运行环境的优化实现方法，它对jmf中的复用器对象的概念进行扩展，引入“混音器”处理音频流、视频复用器和“令牌控制机制”处理视频流，提供了不限用户数量和是否拥有发言权条件限制的音频协作手段，也提供了用户数量及发送权受限的视频交流手段。
• The teaching system adopts a way of video and audio streaming to support the distance teaching in the circummstance of network . lt applys the solution of realnetworks company ' s stream media , uses media player to play courseware and uses smil and realone player ' s performance to combine different kinds of courseware
  采用java反射机制可以把可能变化的部分分离出来，由程序动态加载那部分实现。这样大大减少了程序的耦合性，从而提高软件的复用性，有利于教学系统以后的功能扩展。教学支持系统采用视音频流的方式解决网络环境下的教学问题。
• Using rtf , with the help of directshow ' s frame , two filters are programmed to send and receive multicasting data in the application research . one filter , a tenderer filter , is used in the sender side to send audio file or real - time audio to the network . another , a source filter , is used in the receiver side to receive audio stream from the network
  使用rtp协议，借助于directshow的框架，在应用研究中建立了两个用于发送和接收组播媒体流的过滤器( filter ) ，一个是呈现过滤器( rendererfilter ) ，用在发送端，用于向网络中发送音频文件和实时音频；另一个是源过滤器( sourcefilter ) ，用在接收端，用于接收从网络中发送过来的音频流。
• At first the article puts emphasis on analyzing those current network ip technology , various audio codec algorithms , realtime stream medium transmit technology and those process mechanism of realtime low bandwidth audio stream medium , etc . in allusion to high requirement of system , resulted from so many terminations of attending a lecture , rate of flow , bad situation of network and realtime interactive voice , a new algorithm and the relevant project of processing low bit - rate audio stream was brought forward
  本文着重分析了当前网络方面的ip技术，各种音频编码算法，实时流媒体传输技术，实时低带宽音频流媒体的处理机制等等。针对网络实时应用中诸如客户端众多，各种多媒体数据流量大，网络状况差，语音交互实时性等方面较高的要求，提出一种新的实时低带宽音频流处理算法及相应的处理方案。
• The scheme adjusts the audio frame length without impairing audio quality according to the occupied level of audio playout buffer and changes the playout frame rate of video according to the occupied level of video playout buffer to achieve intra - media synchronization of audio stream and video stream
  算法分别根据音频和视频播放缓冲区的占用水平，在不影响音频播放质量的情况下调整音频帧的长度，实现音频流媒体内同步；通过改变视频流的播放帧率达到视频流媒体内同步。