Director11．5制作基于lingo语言的嵌入式音频播放器

Director11．5中音频的播放一般不能直接控制，不能满足用户的多种需要。本文通过对Director11．5中音频播放的深入分析和研究，运用lingo语句进行脚本编写，制作了嵌入式音频播放器，实现了多媒体作品内部直接自主控制音频播放，实现音频播放的多种效果，具有较强的参考价值和启发性。     

PSP音乐播放请外援

用PSP播放音乐还不简单,将MP3文件一股脑拷贝到记忆卡里不就可以播放吗？相信这是很多PSP玩家播放音乐时的习惯动作,但某些情况下用这个办法播放音频并不合适,比如要严格按照一定次序来播放音频,要想玩转PSP音频播放,请外援是必不可少的.     

让RealPlayer连续播放多个文件

使用RealPlayer播放音频或者视频文件时，总是要一个一个的选择播放，非常麻烦，那么，有没有方法让音频或者视频文件连续播放呢？答案是肯定的，方法如下：     

安卓应用中无声音频的收集与检测

在安卓系统中，一些安卓应用为了避免被系统杀死，会通过各种方式在后台占用系统的CPU，内存等资源，实现后台保活.这类行为会加速安卓系统的电量消耗.其中一种后台保活的方式是在后台持有Audiomix锁并播放无声音频.针对这种行为，本文设计了相应的方案来检测这个问题.通过对安卓源码进行修改，收集到安卓应用正在播放的音频数据，再通过检测脚本对音频进行实时检测，来判断安卓应用是否在后台播放无声音频来实现保活.实验分析了50个安卓应用，结果表明该方法可以有效检测此类行为.     

完美播放——视频媒体全接触

对下越来越盛行的网络媒体——音频与视频，给大家带来了不少欢渝；一首好歌，一部好的电影，不仅可以释放我们的身心，还给我们更多的启示……但在播放各种媒体文件时经常会有这样或那样的问题——音频、视频不同步；不能播放等问题，这期的专题中，我们将会为大家介绍一些解决媒体播放过程中各种常见问题的技巧。     

P2P即时音频通信系统的java实现

根据sun公司开发的JavaSound开发包,利用其中的各种多媒体音频类和接口,实现了音频通信。系统采用固定延时播放来保证播放的连续性,并通过发送音频数据前增加分组的冗余信息,来避免分组丢失。本系统建立在sun公司的p2p平台jxta上。     

Q＆A——RealPlayer播放器没有播放列表

如何利用RealPlayer连续播放音频或者视频的文件啊？一个一个点击播放太浪费时间和精力了，希望能教我—个好方法，使其具有类似Winamp的列表播放功能？     

利用Delphi的流对象对多媒体文件进行操作

本文提出了多媒体文件(特别是音频和视频文件)的存取与播放的操作问题，为了解决该问题，首先介绍Delphi中的流对象的基本概念和知识；然后分别借用MSA Access数据库和SQL Server数据库利用数据漉对象来实现音频和视频文件的存储和读取；最后通过MediaPlayer组件演示从数据库中读取的音频和视频数据。     

音频天下,谁是英雄--四款音频播放器软件大比拼

电脑休闲，除了玩网游，聊天以外，当然还少不了放点儿音乐来让自己的耳朵享受一番。古语有云“工欲善其事，必先利其器”，选择一款适合自己的音频播放器是至关重要的，如今的音频播放器已不再像当年的Winamp一枝独秀，几乎每个星期都有新的播放器诞生在本文中，精心挑选了四款音频播放软件来进行评测，看完本文后．大家就可自行权衡，从而选择适合自己的播放器软件了。     

亲密接触MP3播放机

音乐一直是人们生活中必备的“调味品”。从１９７９年日本索尼公司推出世界上第一台便携式磁带播放机Walkman开始，便携式音乐播放机就成为音乐爱好者的宠儿。１９９６年，基于MP３的音频压缩技术问世，第一台具有影响力的MP３播放机是美国Diamond（帝盟）公司在１９９８年年底推出的Rio３００。从此，MP３播放机这个主要以闪存卡为存储介质的便携音频播放机便紧紧锁住了人们的视线。MP３播放机正在以它独有的魅力席卷全球，如果说CD播放机使人们听到的音乐由模拟音频转为数字音频，那么MP３播放机则使大家真正体会到了移动音乐的乐…     

Design and Experimental Evaluation of an Adaptive Playout Delay Control Mechanism for Packetized Audio for Use over the Internet

We describe the design and the experimental evaluation of a playout delay control mechanism we have developed in order to support unicast, voice-based audio communications over the Internet. The proposed mechanism was designed to dynamically adjust the talkspurt playout delays to the traffic conditions of the underlying network without assuming either the existence of an external mechanism for maintaining an accurate clock synchronization between the sender and the receiver during the audio communication, or a specific distribution of the audio packet transmission delays. Performance figures derived from several experiments are reported that illustrate the adequacy of the proposed mechanism in dynamically adjusting the audio packet playout delay to the network traffic conditions while maintaining a small percentage of packet loss.     

Packet audio playout delay adjustment: performance bounds and algorithms

In packet audio applications, packets are buffered at a receiving site and their playout delayed in order to compensate for variable network delays. In this paper, we consider the problem of adaptively adjusting the playout delay in order to keep this delay as small as possible, while at the same time avoiding excessive “loss” due to the arrival of packets at the receiver after their playout time has already passed. The contributions of this paper are twofold. First, given a trace of packet audio receptions at a receiver, we present efficient algorithms for computing a bound on the achievable performance of any playout delay adjustment algorithm. More precisely, we compute upper and lower bounds (which are shown to be tight for the range of loss and delay values of interest) on the optimum (minimum) average playout delay for a given number of packet losses (due to late arrivals) at the receiver for that trace. Second, we present a new adaptive delay adjustment algorithm that tracks the network delay of recently received packets and efficiently maintains delay percentile information. This information, together with a “delay spike” detection algorithm based on (but extending) our earlier work, is used to dynamically adjust talkspurt playout delay. We show that this algorithm outperforms existing delay adjustment algorithms over a number of measured audio delay traces and performs close to the theoretical optimum over a range of parameter values of interest.     

VOIP动态缓出时延算法研究分析

在以包为单位进行数据传输合、语音应用程序(VOIP,Voice Over Internet Protocol)中,为了补偿数据包在网络传输中不可预知的网络传输延迟,在接收端首先必须把接收到的数据包缓存起来,缓存一定的时间再播放出来,以减少通话的抖动,得到比较满意的通话质量。文章主要研究动态缓出时延算法,力求使这个缓出时延尽可能小,同时尽可能减少包的丢失率。文章提出了一个有效动态缓出时延算法,该算法主要跟踪最近到达的数据包的网络传输时延求出其近似分布函数,并利用这些信息和延迟峰的侦测算法预测下一个语音峰的缓出时延。实验结果表明利用该算法可以在缓出时延和包丢失率之间达到最佳平衡,是一种理想、有效的算法。     

HMM delay prediction technique for VoIP

This paper proposes a new algorithm for predicting audio packet playout delay for voice conferencing applications that use silence suppression. The proposed algorithm uses a hidden Markov model (HMM) to predict the playout delay. Several existing algorithms are reviewed to show that the HMM technique is based on a combination of various desirable features of other algorithms. Voice over Internet protocol (VoIP) applications produce packets at a deterministic rate but various queuing delays are added to the packets by the network causing packet interarrival jitter. Playout delay prediction techniques schedule audio packets for playout and attempt to make a reasonable compromise between the number of lost packets, the one-way delay and the delay variation since these criteria cannot be optimized simultaneously. In particular, this paper will show that the proposed HMM technique makes a good compromise between the mean end-to-end delay, end-to-end delay standard deviation and average packet loss rate.     

Adaptive playout scheduling and loss concealment for voice communication over IP networks

The quality of service limitation of today's Internet is a major challenge for real-time voice communications. Excessive delay, packet loss, and high delay jitter all impair the communication quality. A new receiver-based playout scheduling scheme is proposed to improve the tradeoff between buffering delay and late loss for real-time voice communication over IP networks. In this scheme the network delay is estimated from past statistics and the playout time of the voice packets is adaptively adjusted. In contrast to previous work, the adjustment is not only performed between talkspurts, but also within talkspurts in a highly dynamic way. Proper reconstruction of continuous playout speech is achieved by scaling individual voice packets using a time-scale modification technique based on the Waveform Similarity Overlap-Add (WSOLA) algorithm. Results of subjective listening tests show that this operation does not impair audio quality, since the adaptation process requires infrequent scaling of the voice packets and low playout jitter is perceptually tolerable. The same time-scale modification technique is also used to conceal packet loss at very low delay, i.e., one packet time. Simulation results based on Internet measurements show that the tradeoff between buffering delay and late loss can be improved significantly. The overall audio quality is investigated based on subjective listening tests, showing typical gains of 1 on a 5-point scale of the Mean Opinion Score.     

An empirical study of delay jitter management policies

This paper presents an empirical study of several policies for managing the effect of delay jitter on the playout of audio and video in computer-based conferences. The problem addressed is that of managing the fundamental trade-off between display with low latency and display with few gaps. We describe a particular policy calledqueue monitoring which observes delay jitter over time and dynamically adjusts display latency in order to support low-latency conferences with an acceptable gap rate. Queue monitoring is evaluated by comparing it with two policies from the literature in a study based on measurements from a computer-based conferencing system. Our results show that queue monitoring performs as well or better than the other policies over the range of observed network loads. More importantly, we show that queue monitoring performs better on those network loads for which the other policies exhibit poor performance.This work was supported by the National Science Foundation (grant numbers CCR-9110938 and ICI-9015443), and by the IBM and Intel Corporations     

基于分组网络的实时语音自适应同步算法

首先介绍了多媒体通信中的流同步问题以及影响同步的各种因素,然后提出一种基于播放时间的自适应同步算法并加以推导证明,最后介绍了它在已设计的语音传输系统中的具体实现。     

Dynamic playout scheduling algorithms for continuous multimedia streams

In this paper, we investigate a playout scheduling framework for supporting the continuous and synchronized presentations of multimedia streams in a distributed multimedia presentation system. We assume a situation in which the server and network transmissions provide sufficient support for the delivery of media objects. In this context, major issues regarding the enforcement of the smooth presentation of multimedia streams at client sites must be addressed to deal with rate variance of stream presentations and delay variance of networks. We develop various playout-scheduling algorithms that are adaptable to quality-of-service parameters. The proposed algorithms permit the local adjustment of unsynchronized presentations by gradually accelerating or retarding presentation components, rather than abruptly skipping or pausing the presentation materials. A comprehensive experimental analysis of the proposed algorithms demonstrates that our algorithms can effectively avoid playout gaps (or hiccups) in the presentations. This scheduling framework can be readily used to support customized multimedia presentations.     

A Multilayered Audiovisual Streaming System Using the Network Bandwidth Adaptation and the Two-Phase Synchronization

Synchronous audiovisual streaming and playout are two of the major issues in the multimedia communication network. However, the past corresponding researches of media synchronization mainly focused on the mono-quality and single-layer (nonscalable) audiovisual data. To overcome challenges of ubiquitous multimedia streaming, a scalable audiovisual coder that can provide flexible scalabilities and adaptive streaming control to adapt to complicated network situations are both required. This paper proposes a multilayered audiovisual streaming scheme to deliver layered audiovisual data synchronously, which is called ML-AVSS. Fine-granular scalability (FGS) and bit-sliced arithmetic coding (BSAC) techniques are used to segment video and audio data into one base-layer and multiple enhancement-layer bitstreams. With advantages of audiovisual layer coding, a de-jitter procedure, a conditional retransmission mechanism and a playout synchronization mechanism are designed to transmit hybrid multilayered audiovisual bitstreams in consideration of the result of a network bandwidth adaptation and the distinct decoding time-complexity. Experimental results show that the proposed ML-AVSS is a feasible streaming scheme to overcome challenges of ubiquitous multimedia streaming, e.g., constrained channel bandwidth, quality degradation, unsmooth playout, etc.     

基于感知质量的多媒体会议系统的软件实现

论述了多媒体视频会议系统中多点控制单元(MCU)服务器、客户端的软件设计及实现。在客户端设计中,引入了基于用户感知质量的播放缓存算法,以提高动态网络环境下多媒体会议的音频性能。