标签：flash actionscript 播放 推流 rtmp
本文记录一些基于Flash的流媒体处理的例子。Flash平台最常见的流媒体协议是RTMP。此前记录的一些基于C/C++的RTMP播放器/推流器，但是没有记录过基于Flash中的ActionScript的RTMP播放器/推流器。其实基于Flash的RTMP播放器/推流器才能算得上是RTMP技术中的“正规军”。RTMP本身设计出来就是用于Flash平台之间通信的，而且RTMP最大的优势——“无插件直播”，也是得益于广泛安装在客户端的Flash Player。因此本文分别记录一个基于ActionScript的RTMP播放器和基于ActionScript的RTMP推流器。
基于C/C++的RTMP流媒体处理的例子可以参考下面几个。
发布
最简单的基于librtmp的示例：发布H.264（H.264通过RTMP发布）
最简单的基于librtmp的示例：发布（FLV通过RTMP发布）
接收
最简单的基于librtmp的示例：接收（RTMP保存为FLV）
最简单的基于FFMPEG+SDL的视频播放器 ver2 （采用SDL2.0）
相比于使用C/C++处理RTMP而言，使用ActionScript处理RTMP非常的简单。RTMP建立连接的方法都已经封装好了，只需要调用现成的接口函数就可以了。但是使用ActionScript处理RTMP的劣势也十分明显——可供自己开发的地方很少。由于Flash本身不开源，所以我们无法得到它的底层代码，因而也不能对编解码底层的参数进行调整。总而言之，ActionScript处理RTMP可以概括为几个字：“简单但是不灵活”。
ActionScript播放RTMP流媒体的流程如下图所示。
从图中可以看出，流程可以分成两部分：播放和显示。
播放分成3步：
（1）建立NetConnection前2步分别建立了RTMP规范中的两个逻辑结构：NetConnection和NetStream。NetConnection代表服务器端应用程序和客户端之间基础的连通关系。NetStream代表了发送多媒体数据的通道。服务器和客户端之间只能建立一个NetConnection，但是基于该连接可以创建很多NetStream。这两个结构的结构如下图所示。
显示部分将播放的视频显示在“舞台”上。这一部分通过创建一个Video对象实现。
ActionScript推送RTMP流媒体的流程如下图所示。
本文附件中包含以下2个ActionScript工程：
simplest as3 rtmp player，最简单的RTMP播放器，其中包含3个独立的子工程：
simplest_as3_rtmp_player：最简单的RTMP播放器。simplest_as3_local_player：最简单的本地文件播放器。simplest_as3_rtmp_player_multiscreen：最简单的RTMP多屏播放器。
simplest_as3_rtmp_streamer，最简单的RTMP推流器下面看一下上述几个工程的源代码。
simplest_as3_rtmp_player
simplest_as3_rtmp_player是最简单的RTMP播放器，代码如下所示。
/**
* 最简单的基于ActionScript的RTMP播放器
* Simplest AS3 RTMP Player
*
* 雷霄骅 Lei Xiaohua
* leixiaohua1020@126.com
* 中国传媒大学/数字电视技术
* Communication University of China / Digital TV Technology
* http://blog.csdn.net/leixiaohua1020
*
* 本程序使用ActionScript3语言完成，播放RTMP服务器上的流媒体
* 是最简单的基于ActionScript3的播放器。
*
* This software is written in Actionscript3, it plays stream
* on RTMP server
* It‘s the simplest RTMP player based on ActionScript3.
*
*/
package {
import flash.display.Sprite;
import flash.net.NetConnection;
import flash.events.NetStatusEvent;
import flash.events.AsyncErrorEvent;
import flash.net.NetStream;
import flash.media.Video;
public class simplest_as3_rtmp_player extends Sprite
{
var nc:NetConnection;
var ns:NetStream;
var video:Video;
public function simplest_as3_rtmp_player()
{
nc = new NetConnection();
nc.addEventListener(NetStatusEvent.NET_STATUS, netStatusHandler);
nc.connect("rtmp://localhost/live");
}
private function netStatusHandler(event:NetStatusEvent):void
{
trace("event.info.level: " + event.info.level + "\n", "event.info.code: " + event.info.code);
switch (event.info.code)
{
case "NetConnection.Connect.Success":
doVideo(nc);
break;
case "NetConnection.Connect.Failed":
break;
case "NetConnection.Connect.Rejected":
break;
case "NetStream.Play.Stop":
break;
case "NetStream.Play.StreamNotFound":
break;
}
}
// play a recorded stream on the server
private function doVideo(nc:NetConnection):void {
ns = new NetStream(nc);
ns.addEventListener(NetStatusEvent.NET_STATUS, netStatusHandler);
video = new Video(640,480);
video.attachNetStream(ns);
ns.play("myCamera");
addChild(video);
}
// create a playlist on the server
/*
private function doPlaylist(nc:NetConnection):void {
ns = new NetStream(nc);
ns.addEventListener(NetStatusEvent.NET_STATUS, netStatusHandler);
video = new Video();
video.attachNetStream(ns);
// Play the first 3 seconds of the video
ns.play( "bikes", 0, 3, true );
// Play from 20 seconds on
ns.play( "bikes", 20, -1, false);
// End on frame 5
ns.play( "bikes", 5, 0, false );
addChild(video);
}
*/
}
}simplest_as3_local_player
simplest_as3_local_player用于播放本地FLV文件。ActionScript中播放本地视频（*.flv）和播放RTMP流程是一样的：先创建NetConnection，再创建NetStream。它们最大的不同在于，播放本地文件建立NetConnection的时候，是不传地址的。例如播放RTMP的时候代码如下：
nc.connect("rtmp://localhost/live");播放本地文件的时候代码如下：nc.connect(null);调用play()的时候，RTMP传递服务器上的路径，如下所示。
ns.play("myCamera");本地文件直接传递本地路径，如下所示。ns.play("sintel.flv");simplest_as3_rtmp_player_multiscreen
simplest_as3_rtmp_player_multiscreen是一个多屏播放的简单例子。实现了2x2网格播放4路视频。不再过多记录。
simplest_as3_rtmp_streamer
simplest_as3_rtmp_player是最简单的RTMP推流器，代码如下所示。
/**
* 最简单的基于ActionScript的RTMP推流器
* Simplest AS3 RTMP Streamer
*
* 雷霄骅 Lei Xiaohua
* leixiaohua1020@126.com
* 中国传媒大学/数字电视技术
* Communication University of China / Digital TV Technology
* http://blog.csdn.net/leixiaohua1020
*
* 本程序使用ActionScript3语言完成，推送本地摄像头的数据至RTMP流媒体服务器，
* 是最简单的基于ActionScript3的推流器。
*
* This software is written in Actionscript3, it streams camera‘s video to
* RTMP server.
* It‘s the simplest RTMP streamer based on ActionScript3.
*
*/
package {
import flash.display.MovieClip;
import flash.net.NetConnection;
import flash.events.NetStatusEvent;
import flash.net.NetStream;
import flash.media.Video;
import flash.media.Camera;
import flash.media.Microphone;
//import flash.media.H264Profile;
//import flash.media.H264VideoStreamSettings;
public class simplest_as3_rtmp_streamer extends MovieClip
{
var nc:NetConnection;
var ns:NetStream;
var nsPlayer:NetStream;
var vid:Video;
var vidPlayer:Video;
var cam:Camera;
var mic:Microphone;
var screen_w:int=320;
var screen_h:int=240;
public function simplest_as3_rtmp_streamer()
{
nc = new NetConnection();
nc.addEventListener(NetStatusEvent.NET_STATUS, onNetStatus);
nc.connect("rtmp://localhost/live");
}
private function onNetStatus(event:NetStatusEvent):void{
trace(event.info.code);
if(event.info.code == "NetConnection.Connect.Success"){
publishCamera();
displayPublishingVideo();
displayPlaybackVideo();
}
}
private function publishCamera() {
//Cam
cam = Camera.getCamera();
/**
* public function setMode(width:int, height:int, fps:Number, favorArea:Boolean = true):void
* width:int — The requested capture width, in pixels. The default value is 160.
* height:int — The requested capture height, in pixels. The default value is 120.
* fps:Number — The requested capture frame rate, in frames per second. The default value is 15.
*/
cam.setMode(640, 480, 15);
/**
* public function setKeyFrameInterval(keyFrameInterval:int):void
* The number of video frames transmitted in full (called keyframes) instead of being interpolated by the video compression algorithm.
* The default value is 15, which means that every 15th frame is a keyframe. A value of 1 means that every frame is a keyframe.
* The allowed values are 1 through 300.
*/
cam.setKeyFrameInterval(25);
/**
* public function setQuality(bandwidth:int, quality:int):void
* bandwidth:int — Specifies the maximum amount of bandwidth that the current outgoing video feed can use, in bytes per second (bps).
* To specify that the video can use as much bandwidth as needed to maintain the value of quality, pass 0 for bandwidth.
* The default value is 16384.
* quality:int — An integer that specifies the required level of picture quality, as determined by the amount of compression
* being applied to each video frame. Acceptable values range from 1 (lowest quality, maximum compression) to 100
* (highest quality, no compression). To specify that picture quality can vary as needed to avoid exceeding bandwidth,
* pass 0 for quality.
*/
cam.setQuality(200000, 90);
/**
* public function setProfileLevel(profile:String, level:String):void
* Set profile and level for video encoding.
* Possible values for profile are H264Profile.BASELINE and H264Profile.MAIN. Default value is H264Profile.BASELINE.
* Other values are ignored and results in an error.
* Supported levels are 1, 1b, 1.1, 1.2, 1.3, 2, 2.1, 2.2, 3, 3.1, 3.2, 4, 4.1, 4.2, 5, and 5.1.
* Level may be increased if required by resolution and frame rate.
*/
//var h264setting:H264VideoStreamSettings = new H264VideoStreamSettings();
// h264setting.setProfileLevel(H264Profile.MAIN, 4);
//Mic
mic = Microphone.getMicrophone();
/*
* The encoded speech quality when using the Speex codec. Possible values are from 0 to 10. The default value is 6.
* Higher numbers represent higher quality but require more bandwidth, as shown in the following table.
* The bit rate values that are listed represent net bit rates and do not include packetization overhead.
* ------------------------------------------
* Quality value | Required bit rate (kbps)
*-------------------------------------------
* 0 | 3.95
* 1 | 5.75
* 2 | 7.75
* 3 | 9.80
* 4 | 12.8
* 5 | 16.8
* 6 | 20.6
* 7 | 23.8
* 8 | 27.8
* 9 | 34.2
* 10 | 42.2
*-------------------------------------------
*/
mic.encodeQuality = 9;
/* The rate at which the microphone is capturing sound, in kHz. Acceptable values are 5, 8, 11, 22, and 44. The default value is 8 kHz
* if your sound capture device supports this value. Otherwise, the default value is the next available capture level above 8 kHz that
* your sound capture device supports, usually 11 kHz.
*
*/
mic.rate = 44;
ns = new NetStream(nc);
//H.264 Setting
//ns.videoStreamSettings = h264setting;
ns.attachCamera(cam);
ns.attachAudio(mic);
ns.publish("myCamera", "live");
}
private function displayPublishingVideo():void {
vid = new Video(screen_w, screen_h);
vid.x = 10;
vid.y = 10;
vid.attachCamera(cam);
addChild(vid);
}
private function displayPlaybackVideo():void{
nsPlayer = new NetStream(nc);
nsPlayer.play("myCamera");
vidPlayer = new Video(screen_w, screen_h);
vidPlayer.x = screen_w + 20;
vidPlayer.y = 10;
vidPlayer.attachNetStream(nsPlayer);
addChild(vidPlayer);
}
}
}程序运行后的结果如下图所示。
运行结果如下图所示。
运行结果如下图所示。
运行结果如下图所示。
SourceForge：https://sourceforge.net/projects/simplestflashmediaexample/
Github：https://github.com/leixiaohua1020/simplest_flashmedia_example
开源中国：http://git.oschina.net/leixiaohua1020/simplest_flashmedia_example
CSDN下载：http://download.csdn.net/detail/leixiaohua1020/8456441
最简单的基于Flash的流媒体示例：RTMP推送和接收（ActionScript）
标签：flash actionscript 播放 推流 rtmp
原文地址：http://blog.csdn.net/leixiaohua1020/article/details/43936141
