常见3G流媒体业务投诉的分析与排查

3G流媒体业务,是移动通信发展的方向.随着3G多媒体手机的大规模上市,以手机为载体的相关产业也将随之得到空前发展,通过音频、视频工具提供的服务将占主导地位.但运营商要发展流媒体业务还需解决诸多技术问题,如文件制作技术、文件定位调度管理技术、节点选择技术、同步技术、容错技术、终端能力以及安全机制等.结合日常流媒体投诉处理的一些经验,总结出几种常见流媒体业务投诉的处理和分析方法.     

基于ARM-Linux嵌入式系统的流媒体服务器实现

简要介绍了嵌入式开发的软硬件平台,针对H.263格式视频流和G.729格式音频流的多媒体数据实时传输要求,设计并实现了ARM-Linux嵌入式平台下基于RTP/RTCP协议的实时流媒体服务器.     

WMA流媒体音质测试与分析

WMA流媒体是一种目前电台及多媒体网站使用最为广泛的音频直播、点播及下载的主要音频传输格式,通过对不同码率WMA音频的主观评价,并针对波形统计、光谱图对比及频谱分析三种客观数据对比,对主观评价与客观数据之间的关系进行分析论述,并对WMA流媒体的应用进行指导。     

点击3G关键词——流媒体业务

随着3G时代的愈行愈近，其所提供的丰富多彩的业务内容为人们所期待．以流媒体为代表的视讯业务更是频频亮相预热．无疑已经成为一道亮丽的风景线。流媒体是把数据(通常是音频和视频)实时从服务端传送到客户端(手机终端)．客户端收到数据后．对其进行译码及播放。流媒体是未来移动通信的主流业务之一。     

移动流媒体关键技术及其平台实现

随着移动通信的迅速发展,移动流媒体成为移动通信发展的新热点,第三代移动通信(3G)标准的制订使得通过无线信道传输视频信息成为可能.介绍了移动流媒体技术的基本特征,阐述了移动流媒体的关键技术,包括网络协议、系统结构、QoS控制等,最后介绍了基于RealSystem的移动流媒体平台的实现.     

流媒体与移动流媒体技术的研究

流媒体在移动通信中的应用将是3G和3G后移动通信增值的主要方式。文章介绍了流媒体与移动流媒体的基本原理,阐述了移动流媒体的主要实现技术——高容错性、H．264视频压缩标准和传输协议,简要分析了移动流媒体业务发展的制约因素。     

移动流媒体业务的技术与应用

电信市场正在经历从基础服务到娱乐经济的历史转折期,移动通信的未来充满了机遇.在3G的发展进程中,移动流媒体业务是其中最重要的一项.介绍流媒体技术,通过了解移动流媒体业务系统和移动流媒体业务中的关键技术,分析移动流媒体业务发展中存在的问题.     

无线实时数据通信对等网络构架方案

基于移动通信网络的P2P流媒体共享系统，能实现网络中任意用户可以搜索到并且快速共享其所需要的流媒体资源，这对于网络信息交流和高效合理地利用网络资源有着重要的意义。文章提出了一种基于移动通信网络的对等流媒体覆盖网络方案，阐述了对等流媒体覆盖网络的可实现性和有效运行的一些必要条件，最后提出了基于提高服务质量的系统优化思路。     

基于嵌入式流媒体服务器的视频点播系统

移动通信网络流媒体服务资费较高,限制了人们在没有有线电视、宽带互联网连接、无线WiFi互联网连接的地方用手机收看视音频节目.针对这一现状,提出了一种基于嵌入式流媒体服务器的多功能视频点播系统.以ARM板和WiFi网络适配器作为流媒体服务器,设计一款Android手机播放器作为客户端,实现在智能手机上、多个客户端同时免流量接收服务器本地或外接设备推送的高清、超高清视频.该系统适用于公交、广场、电梯、商场、景区等场合,人们用手机观看视音频,实现不用耗费自己的流量,自由快速地选择自己喜欢的节目.     

流媒体及其在移动通信中的应用

本文介绍了流媒体技术的概念和特点，对流媒体应用于移动通信领域的可行性进行了详细分析，讨论了一个基于EGPRS网络的流媒体系统的结构、功能和工作过程，最后探讨了流媒体技术在移动通信中的发展现状和前景。     

Extended AMR-WB for high-quality audio on mobile devices

This article presents the architecture, performance, and application scenarios of the AMR-WB+ (extended AMR-WB) audio codec, which provides high quality at exceptionally low rates, and consistent quality over all audio types. This codec was recently selected by 3GPP and DVB to support low-bit-rate audio and audiovisual applications on mobile networks.     

低比特率音频编解码在嵌入式平台中的移植与优化

本文描述了AMR—WB＋音频编解码的结构框架,复杂性,移植和优化。AMR—WB＋编解码在低码率范围内表现出相当优秀的品质,并且在各种音频类型上都有一致的高性能。因此被3GPP和DVB选定为移动网络中低码率音频指定编解码算法。     

基于信号规整和稀疏变换的语音与音频分层编码方法

基于语音和音频信号的固有周期性特征,本文构建了一种适合语音和音频信号的统一分析/合成模型,并分别在24kbps和32kbps码率下,实现了对宽带语音和音频信号的高质量分层编码.首先,本文将具有时变周期的输入信号规整为具有固定周期的信号,并对规整后的周期信号构建规整矩阵;其次,对规整矩阵的行和列分别进行调制叠接变换(MLT)和离散余弦变换(DCT),完成规整矩阵的稀疏化;最后,利用分带量化和矢量哈夫曼编码完成稀疏矩阵元素的量化和编码.主客观测试结果表明,本文所提方法的语音、音频及其混合信号的编码质量均优于同等速率下的ITU-T G.722.1和AMR-WB编码器.     

The adaptive multirate wideband speech codec: system characteristics, quality advances, and deployment strategies

Wideband speech is the major differentiation and attraction of third-generation network services in both the circuit and packet switched domain. Increased audio bandwidth introduces a significant leap in perceived quality of service compared to currently utilized narrowband telephony in second-generation mobile communications and the PSTN. The adaptive multirate wideband (AMR-WB) speech codec is the service enabler for improved user experience. It is an established 3GPP and ITU-T wideband speech codec standard and represents the state-of-the-art in speech quality as well as robustness in error prone radio channels. It is also the first codec algorithm standardized for wideband speech for mobile communications.     

G.718: A New Embedded Speech and Audio Coding Standard with High Resilience to Error-Prone Transmission Channels

This article is an overview of the standardization, architecture, and performance of the new ITU-T Recommendation G.718. G.718 is an embedded variable bit rate codec providing a scalable solution for compression of 8 and 16 kHz sampled speech and audio signals at rates between 8 kb/s and 32 kb/s. It comprises five layers where higher-layer bitstreams can be discarded without affecting the lower layersiquest decoding. The codec also has an optional core layer interoperable with ITU-T G.722.2 (3GPP AMR-WB) at 12.65 kb/s. G.718 was designed to provide high speech quality at low bit rates and to be robust to significant rates of frame erasures or packet losses. It is also targeting good quality for generic audio at higher rates.     

Challenges of 16 kHz in acoustic pre- and post-processing for terminals

The standardization of the AMR-WB speech codec by 3GPP opens a new area on audio pre-and post-processing integrated in telecommunication terminals. It appears that the transition from narrowband to wideband signals is not a trivial task, but can rather be seen as a challenge, as acoustic quality requirements, scalability and computational limitations have to be faced at the same time. This article presents key examples to address the transition issue as well as proposals for low-complexity implementations of acoustic echo cancellation and noise reduction.     

基于Volterra级数预测的音频频带扩展

 本文采用非线性分析方法,基于Volterra级数提出了一种宽带音频信号的频带扩展方法,并利用高斯混合模型(Gaussian Mixture Model,GMM)和码本映射技术对扩展后的音频信号进行了谱包络和能量增益调整.实验表明,所提算法的性能要好于已有的非线性频带扩展算法,当用本文的方法替代ITU-T G.722.1C编码器中的噪声填充技术时,在24kbps得到了提升的超宽带音频质量.     

AMR-WB算法仿真及其在TMS320VC5509的优化实现

自适应多速率宽带语音编码标准(AMR-WB)是第一个同时用于有线和无线通用标准的宽带语音编码器.介绍了AMR-WB算法井进行了AMR-WB算法定点C代码的仿真实现以及将该算法移植到TMS320VC5509DSP芯片的实现过程,具体包括C5509存储器的分配,C5509与PC机兼容性问题的处理;并对该算法进行初步优化,基...     

Real-Time Software Video Codec with a Fast Adaptive Motion Vector Search

A PC-based real-time software MPEG-4 video codec with a fast adaptive motion vector search is presented. In a fast adaptive motion estimation (ME) technique, the search order is dynamically changed in according with the motion of objects. This technique suppresses load fluctuation in the ME and contributes to the stable real-time work of the software codec. MMX instructions are used to increase the codec speed. On a portable PC, the software video codec supports satisfactory mobile visual communication at 64 kbps and 128 kbps, for example, at QCIF 15 fps. The codec on a 450 MHz PentiumII processor can encode and decode 30 CIF frames in real-time.     

Results from subjective testing of the HD codec at 16-96 kbps

As continuing support of National Public Radio's Tomorrow Radio project, this paper presents results from a consumer study conducted with iBiquity's HD Radio codec. Forty participants judged the quality of the HD codec (HDC) at various bit rates, ranging from 16 kbps to 96 kbps. A range of musical and speech genres were tested. This study provides in-depth information intended to help broadcasters select optimal bit rates when HD Radio's 96 kbps data stream is shared between primary and supplemental channels.