音频流

在提出上述框架的基础上，本文还概括了3G移动网络中的音频流业务的提供。

Supported by this proposed framework , provision of audio streaming services on 3G mobile networks is also outlined .

这些无线技术，通过支持多种互联网应用，如web浏览，电子邮件，视频和音频流具有不同的QoS要求，在这种不同的情况下，对于时延和丢报率的要求各不相同。

These wireless technologies support multiple Internet applications like web browsing , emails , video and audio streaming with varying QoS requirements . During different scenarios , some are tolerant to packet delays and packet losses while others are not .

通过音频流索引，在Web中实现基于例子和关键词缩放搜索能力的多媒体数据的检索。

Realize from the Web multimedia data retrieval based on example and key words zoom search capability .

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。

Another , a source filter , is used in the receiver side to receive audio stream from the network .

P2P网络中IP音频流穿越NAT的设计与实现

Design and Implementation for IP Audio Stream to Traverse NAT in P2P Network

在IP视频会议系统中，音频流需要保持其连续性（媒体内同步），而视频流的变化应与音频信息保持一致（媒体间同步）。

Audio stream needs to keep continuous , and video stream movement should be consistent with audio stream in Video Conference System over IP .

音频流识别实验表明，三音素HMM模型词识别率明显高于单音素HMM；

Audio experiment results show that tri-phone HMM has higher word recognition rates than monophone HMM.

基于RTP的音频流多播系统的JMF实现

The Implementation of JMF of Audio Steam Multicast System Based on RTP

提出了基于MPEG压缩域音频流的足球比赛精彩场景自动分析与提取算法首先直接提取出压缩域音频特征；

This paper presents an algorithm to automatically analyze and extract soccer program highlights scene based on MPEG compressed audio-track analysis .

研究了P2P流、音频流和视频流的共存问题，提出了网络控制和用户控制2种P2P带宽控制的方法。

This pape studies the coexistence of P2P traffic with audio and video applications and analyzes two ways for controlling the P2P bandwidth consumption : network-controlled and user-controlled ways .

第一种方案是改进网络中的路由器，使其支持一些QOS策略，为网络音频流（以及其他各种媒体流）提供一定的QoS保障机制，从而保证音频流的QoS质量。

One solution is to improve the routers in the Internet which support certain QoS strategy so that the QoS of audio ( and any other multimedia ) applications is guaranteed .

分析并提出基于音频流的多媒体数据检索方法和关键技术，在此基础上提出软件实现的框架：自动从Web中引出各种包含音频流的多媒体数据并为其中的音频流建立索引；

Retrieval method and technology of the multimedia data based on audio stream are analyzed and proposed . The frame of their software realization is proposed : derive multimedia data that contain audio stream from the Web and establish index for the audio stream ;

采用了基于RTP／RTCP协议并结合Winsock技术实现了实时视频音频流数据的网络传输，并在实际编程设计中对数据传输的抖动、延时、同步等具体问题作了处理。

The related network protocols selecting RTP / RTCP and using Winsock tech in the paper . Meanwhile , some implementation problems were discussed and treated .

本文研究在PC机Windows操作系统的平台上，如何建立一个基于TCP／IP网络协议的支持视频、音频流的实时、双向传输和通用数据传输（如支持白板功能）的桌面视频会议系统。

This paper discusses how to establish a desktop videoconferencing system on Windows operating system , which supports audio / video full-duplex realtime transfer and other data exchange ( i.e. the white-board ) based on TCP / IP network protocol .

另一方面，接收单元仅接收Vorbis编码的音频流。

The sink pad , on the other hand , simply accepts Vorbis-encoded audio stream .

给出了一个如何使用DirectSound技术实现音频流的捕获并实现网络上多路声音混合的方案。

An implementation of audio streaming capture and mixing multiple audio streaming on network based on DirectSound tech-nology .

NyquistLISP功能强大、高效且可扩展，可为您提供一个良好的预定义数据结构集，用这些结构来表示音频流的各个可参数化的方面。

Nyquist LISP is powerful , efficient , and extensible , providing you with a good set of predefined data structures that represent various parametric aspects of the audio stream .

象视频和音频流、在线游戏、视频会议，IP语音（VoIP）和文件传输协议这些应用（FTP）需要范围广泛的服务质量要求，例如带宽和延时。

Applications such as video and audio streaming , online gaming , video conferencing , Voice over IP ( VoIP ) and File Transfer Protocol ( FTP ) demand a wide range of QoS requirements such as bandwidth and delay .

音视频的传输则采用基于UDP的RTP/RTCP协议实现，保证了视音频流传输的时实性和连续性，避免了分组丢失，确保了分组的有序传输。

Transmission of audio and video realized by RTP / RTCP protocol based on UDP , ensured real-time and continuity of flow transmission of the audio and video , avoided the packet loss and ensured sequential transmission .

出于对产业化和市场等各方面因素的考虑，AVS标准采纳基于MPEG-2系统层对AVS视频流和音频流进行复用的方案，实现与现有MPEG-2系统的兼容性。

In Consideration of multi-factor like industrial realization and market influence , AVS standard decided to adopt the plan of audio and video stream multiplex which based on MPEG-2 systems standard to keep the compatibility of AVS with MPEG-2 systems standard .

介绍了音频流多播系统结构设计及协议选择，探讨了服务器端的JMF实现，并论述了客户端实现方法。

This paper introduces the structure design and protocol selection of audio steam multicast system , probes into the JMF implementation of the server side , and expounds the implementing method of the client side .

针对分布式虚拟战场环境中音频流多点传输这一仿真应用，将音频流传输作为一个联邦成员，定义了音频联邦成员的仿真对象模型，并通过RTI的API函数实现联邦成员交互的过程。

Aiming at the application of audio transmission in distributed virtual battlefield environment , the audio stream is regarded as a federate . The simulation object model is defined according to the HLA / RTI standard , and the interaction of federates is realized through API of RTI .

文章提出一种对MPEG-1静态文件进行索引的方案。该方案把MPEG-1静态文件分解成独立的系统信息、音频流和视频流，并建立索引树。

Through the analysis of the structure and characteristics of MPEG-1 video and audio data , described a method to separate MPEG-1 files into system stream , video stream and audio stream , and construct the index tree of the three streams .

视频被分解为纯视频流和音频流两种模态，每种模态均可依次提取和综合出低层内容和中层内容。

Video is parsed into pure video stream and audio stream .

多流-支持发送不同的音频流到不同的设备。

Multistreaming-Support for sending different audio streams to different devices .

音频流分割和多模式控制的设计与实现

Audio - Stream Segment and Multi - Pattern Control System

实时音频流分割与控制系统

System of dividing and controlling real - time audio frequency

基于模式的实时音频流分割与控制系统

Pattern based audio - stream segment and control system

它支持分裂的一部分，音频流。

It supports splitting a part of audio stream .

只需少量代码即可对音频流产生明显效果。

A small amount of code can have dramatic effects on the audio stream .