让FLVPlayer 组件如何能够识别非flv后缀的视频

[来源：蓝色理想 | 作者：mirycat | 时间：2008-04-04 | 点击： | 收藏本文【大中小】]

在本地测试播放成功的 flv 视频在服务器上无法播放, 并且直接访问 flv 文件被告知找不到文件, 大家一定碰到过这种问题.这是因为 IIS 6 的默认配置中并没有 flv 后缀的 mime 映射, 所以会产生 404 找不到文件的错误. 通常的做法是修改 IIS 配置, 添加 flv 的 mime 映射, 问题是解决了, 但如果因为某些条件限制无法修改配置怎么办, 第一个想到的是修改成让他可以被识别的后缀名, 可惜的是自带的组件默认情况下并不允许你那么做, 否则会报告错误

player.contentPath = "water.swf"; // player 是舞台中 FLVPlayer 组件实例名, 这里我把后缀名改成 swf, 以便可以让 IIS 识别

QUOTE:

报错
1005: Invalid xml: URL: "water.swf" No root node found; if file is an flv it must have .flv extension

解决 : 使用自定义的 NCManager 替代默认的 NCManager
判断 flv 后缀的代码在 NCManager 的 connectToURL 方法中, 我修改了他的实现从而跳过了这个步骤
NCManager.as 262 行处
原 :

提示：您可以先修改部分代码再运行

现 :

提示：您可以先修改部分代码再运行

我修改代码跳过了这个判断, 这样即使不是 flv 后缀同样能够播放, 当然带来的副作用就是无法使用 SMIL 了.

最后我把修改了的代码另存成 HttpConnectionManager.as, 同时类名也改成 HttpConnectionManager, 放到 fla 所在目录中

//****************************************************************************
//Copyright (C) 2004-2005 Macromedia, Inc. All Rights Reserved.
//The following is Sample Code and is subject to all restrictions on
//such code as contained in the End User License Agreement accompanying
//this product.
//****************************************************************************
import mx.video.*;
/**
* <p>Creates <code>NetConnection</code> for <code>VideoPlayer</code>, a
* helper class for that user facing class.</p>
*
* <p>NCManager supports a subset of SMIL to handle multiple streams
* for multiple bandwidths.  NCManager assumes any URL that does not
* being with "rtmp://" and does not end with ".flv" is a SMIL url.
* See SMILParser for more on SMIL support.</p>
*
* @see SMILParser
*/
class HttpConnectionManager implements INCManager {
// my VideoPlayer
private var _owner:VideoPlayer;
// server connection info
private var _contentPath:String;
private var _protocol:String;
private var _serverName:String;
private var _portNumber:String;
private var _wrappedURL:String;
private var _appName:String;
private var _streamName:String;
private var _streamLength:Number;
private var _streamWidth:Number;
private var _streamHeight:Number;
private var _streams:Array;
private var _isRTMP:Boolean;
private var _smilMgr:SMILManager;
private var _bitrate:Number;
/**
* <p>fallbackServerName is exposed in two ways:</p>
*
* <p>User can supply second <meta base> in smil and that base
* attr will be taken as the fallbackServerName (note that only
* the server name will be taken from this and not the application
* name or anything else).</p>
*
* <p>The second way is the user can directly set this by
* accessing the ncMgr property in FLVPlayback or VideoPlayer and
* set fallbackServerName property directly.</p>
*/
public var fallbackServerName:String;
// interval for xn timeout
private var _timeoutIntervalId:Number;
private var _timeout:Number;
/**
* Default connection timeout in milliseconds.
*
* @see #getTimeout()
* @see #setTimeout()
*/
public var DEFAULT_TIMEOUT:Number = 60000;
// bandwidth detection stuff
public var _payload:Number;
private var _autoSenseBW:Boolean;
// info on successful xn
private var _nc:NetConnection;
private var _ncUri:String;
// info on mult xns we try
public var _tryNC:Array;
private var _tryNCIntervalId:Number;
// List of connections tried by connectRTMP, in order tried
private static var RTMP_CONN:Array = [
{ protocol: "rtmp:/", port:"1935" }
, { protocol: "rtmp:/", port:"443" }
, { protocol: "rtmpt:/", port:"80" }
, { protocol: "rtmps:/", port:"443" }
];
// Counter that tracks the next type to try in RTMP_CONN.
private var _connTypeCounter:Number;
public function NCManager()	{
initNCInfo();
initOtherInfo();
// intervals
_timeoutIntervalId = 0;
_tryNCIntervalId = 0;
// actually calls setter
_timeout = DEFAULT_TIMEOUT;
}
private function initNCInfo():Void {
_isRTMP = undefined;
_serverName = undefined;
_wrappedURL = undefined;
_portNumber = undefined;
_appName = undefined;
}
private function initOtherInfo():Void {
_contentPath = undefined;
_streamName = undefined;
_streamLength = undefined;
_streamWidth = undefined;
_streamHeight = undefined;
_streams = undefined;
_autoSenseBW = false;
_payload = 0;
_connTypeCounter = 0;
cleanConns();
}
/*
* @see INCManager#getTimeout()
*/
public function getTimeout():Number {
return _timeout;
}
/*
* @see INCManager#setTimeout()
*/
public function setTimeout(t:Number):Void {
_timeout = t;
if (_timeoutIntervalId != 0) {
clearInterval(_timeoutIntervalId);
_timeoutIntervalId = setInterval(this, "_onFCSConnectTimeOut", _timeout);
}
}
/**
* For RTMP streams, returns value calculated from autodetection,
* not value set via setBitrate().
*
* @see INCManager#getBitrate()
*/
public function getBitrate():Number {
return _bitrate;
}
/**
* This value is only used with progressive download (HTTP), with
* RTMP streaming uses autodetection.
*
* @see INCManager#getBitrate()
*/
public function setBitrate(b:Number):Void {
if (_isRTMP == undefined || !_isRTMP) {
_bitrate = b;
}
}
/**
* @see INCManager#getVideoPlayer()
*/
public function getVideoPlayer():VideoPlayer {
return _owner;
}
/**
* @see INCManager#setVideoPlayer()
*/
public function setVideoPlayer(v:VideoPlayer):Void {
_owner = v;
}
/**
* @see INCManager#getNetConnection()
*/
public function getNetConnection():NetConnection {
return _nc;
}
/**
* @see INCManager#getStreamName()
*/
public function getStreamName():String {
return _streamName;
}
/**
* @see INCManager#isRTMP()
*/
public function isRTMP():Boolean {
return _isRTMP;
}
/**
* @see INCManager#getStreamLength()
*/
public function getStreamLength():Number {
return _streamLength;
}
/**
* @see INCManager#getStreamWidth()
*/
public function getStreamWidth():Number {
return _streamWidth;
}
/**
* @see INCManager#getStreamHeight()
*/
public function getStreamHeight():Number {
return _streamHeight;
}
/**
* @see INCManager#connectToURL()
*/
public function connectToURL(url:String):Boolean {
//ifdef DEBUG
//debugTrace("connectToURL(" + url + ")");
//endif
// init
initOtherInfo();
_contentPath = url;
if (_contentPath == null || _contentPath == undefined || _contentPath == "") {
throw new VideoError(VideoError.INVALID_CONTENT_PATH);
}
// parse URL to determine what to do with it
var parseResults:Object = parseURL(_contentPath);
if (parseResults.streamName == undefined || parseResults.streamName == "") {
throw new VideoError(VideoError.INVALID_CONTENT_PATH, url);
}
// connect to either rtmp or http or download and parse smil
if (parseResults.isRTMP) {
var canReuse:Boolean = canReuseOldConnection(parseResults);
_isRTMP = true;
_protocol = parseResults.protocol;
_streamName = parseResults.streamName;
_serverName = parseResults.serverName;
_wrappedURL = parseResults.wrappedURL;
_portNumber = parseResults.portNumber;
_appName = parseResults.appName;
if ( _appName == undefined || _appName == "" ||
_streamName == undefined || _streamName == "" ) {
throw new VideoError(VideoError.INVALID_CONTENT_PATH, url);
}
_autoSenseBW = (_streamName.indexOf(",") != -1);
return (canReuse || connectRTMP());
} else {
var canReuse:Boolean = canReuseOldConnection(parseResults);
_isRTMP = false;
_streamName = parseResults.streamName;
return (canReuse || connectHTTP());
}
}
/**
* @see INCManager#connectAgain()
*/
public function connectAgain():Boolean
{
//ifdef DEBUG
//debugTrace("connectAgain()");
//endif
var slashIndex:Number = _appName.indexOf("/");
if (slashIndex < 0) {
// return the appName and streamName back to original form
// so we can start this process all over again with the
// fallback server if necessary
slashIndex = _streamName.indexOf("/");
if (slashIndex >= 0) {
_appName += "/";
_appName += _streamName.slice(0, slashIndex);
_streamName = _streamName.slice(slashIndex + 1);
}
return false;
}
var newStreamName = _appName.slice(slashIndex + 1);
newStreamName += "/";
newStreamName += _streamName;
_streamName = newStreamName;
_appName = _appName.slice(0, slashIndex);
close();
_payload = 0;
_connTypeCounter = 0;
cleanConns();
connectRTMP();
return true;
}
/**
* @see INCManager#reconnect()
*/
public function reconnect():Void
{
//ifdef DEBUG
//debugTrace("reconnect()");
//endif
if (!_isRTMP) {
throw new Error("Cannot call reconnect on an http connection");
}
_nc.onStatus = function(info:Object):Void { this.mc.reconnectOnStatus(this, info); };
_nc.onBWDone = function():Void { this.mc.onReconnected(); };
//ifdef DEBUG
//debugTrace("_ncUri = " + _ncUri);
//endif
_nc.connect(_ncUri, false);
}
/**
* dispatches reconnect event, called by
* <code>NetConnection.onBWDone</code>
*
* @private
*/
public function onReconnected():Void {
delete _nc.onStatus;
delete _nc.onBWDone;
_owner.ncReconnected();
}
/**
* @see INCManager#close
*/
public function close():Void {
if (_nc) {
_nc.close();
}
}
/**
* Called by <code>SMILManager</code> when done.
*
* @see INCManager#helperDone()
*/
public function helperDone(helper:Object, success:Boolean) {
if (helper != _smilMgr) return;
if (!success) {
_nc = undefined;
_owner.ncConnected();
delete _smilMgr;
return;
}
// success!
// grab width and height
_streamWidth = _smilMgr.width;
_streamHeight = _smilMgr.height;
// get correct streamname
var parseResults:Object;
var url:String = _smilMgr.baseURLAttr[0];
if (url != undefined && url != "") {
parseResults = parseURL(url);
_isRTMP = parseResults.isRTMP;
_streamName = parseResults.streamName;
if (_isRTMP) {
_protocol = parseResults.protocol;
_serverName = parseResults.serverName;
_portNumber = parseResults.portNumber;
_wrappedURL = parseResults.wrappedURL;
_appName = parseResults.appName;
if (_appName == undefined || _appName == "") {
throw new VideoError(VideoError.INVALID_XML, "Base RTMP URL must include application name: " + url);
}
if (_smilMgr.baseURLAttr.length > 1) {
var parseResults:Object = parseURL(_smilMgr.baseURLAttr[1]);
if (parseResults.serverName != undefined) {
fallbackServerName = parseResults.serverName;
}
}
}
}
_streams = _smilMgr.videoTags;
for (var i:Number = 0; i < _streams.length; i++) {
url = _streams[i].src;
parseResults = parseURL(url);
if (_isRTMP == undefined) {
_isRTMP = parseResults.isRTMP;
if (_isRTMP) {
_protocol = parseResults.protocol;
if (_streams.length > 1) {
throw new VideoError(VideoError.INVALID_XML, "Cannot switch between multiple absolute RTMP URLs, must use meta tag base attribute.");
}
_serverName = parseResults.serverName;
_portNumber = parseResults.portNumber;
_wrappedURL = parseResults.wrappedURL;
_appName = parseResults.appName;
if (_appName == undefined || _appName == "") {
throw new VideoError(VideoError.INVALID_XML, "Base RTMP URL must include application name: " + url);
}
}
} else if ( _streamName != undefined && _streamName != "" &&
!parseResults.isRelative && _streams.length > 1 ) {
throw new VideoError(VideoError.INVALID_XML, "When using meta tag base attribute, cannot use absolute URLs for video or ref tag src attributes.");
}
_streams[i].parseResults = parseResults;
}
_autoSenseBW = _streams.length > 1;
if (!_autoSenseBW) {
if (_streamName != undefined) {
_streamName += _streams[0].parseResults.streamName;
} else {
_streamName = _streams[0].parseResults.streamName;
}
_streamLength = _streams[0].dur;
}
if (_isRTMP) {
connectRTMP();
} else {
if (_autoSenseBW) bitrateMatch();
connectHTTP();
_owner.ncConnected();
}
}
/**
* matches bitrate with stream
*
* @private
*/
private function bitrateMatch():Void {
var whichStream:Number;
if (isNaN(_bitrate)) {
whichStream = 0;
} else {
for (var j:Number = 0; j < _streams.length; j++) {
if (isNaN(_streams[j].bitrate) || _bitrate <= _streams[j].bitrate) {
whichStream = j;
break;
}
}
}
if (isNaN(whichStream)) {
throw new VideoError(VideoError.NO_BITRATE_MATCH);
}
if (_streamName != undefined) {
_streamName += _streams[whichStream].src;
} else {
_streamName = _streams[whichStream].src;
}
_streamLength = _streams[whichStream].dur;
}
/**
* <p>Parses URL to determine if it is http or rtmp.  If it is rtmp,
* breaks it into pieces to extract server URL and port, application
* name and stream name.  If .flv is at the end of an rtmp URL, it
* will be stripped off.</p>
*
* @private
*/
private function parseURL(url:String):Object {
//ifdef DEBUG
//debugTrace("parseURL()");
//endif
var parseResults = new Object();
// get protocol
var startIndex:Number = 0;
var endIndex:Number = url.indexOf(":/", startIndex);
if (endIndex >= 0) {
endIndex += 2;
parseResults.protocol = url.slice(startIndex, endIndex);
parseResults.isRelative = false;
} else {
parseResults.isRelative = true;
}
if ( parseResults.protocol != undefined &&
( parseResults.protocol == "rtmp:/" ||
parseResults.protocol == "rtmpt:/" ||
parseResults.protocol == "rtmps:/" ) ) {
parseResults.isRTMP = true;
startIndex = endIndex;
if (url.charAt(startIndex) == '/') {
startIndex++;
// get server (and maybe port)
var colonIndex:Number = url.indexOf(":", startIndex);
var slashIndex:Number = url.indexOf("/", startIndex);
if (slashIndex < 0) {
if (colonIndex < 0) {
parseResults.serverName = url.slice(startIndex);
} else {
endIndex = colonIndex;
parseResults.portNumber = url.slice(startIndex, endIndex);
startIndex = endIndex + 1;
parseResults.serverName = url.slice(startIndex);
}
return parseResults;
}
if (colonIndex >= 0 && colonIndex < slashIndex) {
endIndex = colonIndex;
parseResults.serverName = url.slice(startIndex, endIndex);
startIndex = endIndex + 1;
endIndex = slashIndex;
parseResults.portNumber = url.slice(startIndex, endIndex);
} else {
endIndex = slashIndex;
parseResults.serverName = url.slice(startIndex, endIndex);
}
startIndex = endIndex + 1;
}
// handle wrapped RTMP servers bit recursively, if it is there
if (url.charAt(startIndex) == '?') {
var subURL = url.slice(startIndex + 1);
var subParseResults = parseURL(subURL);
if (subParseResults.protocol == undefined || !subParseResults.isRTMP) {
throw new VideoError(VideoError.INVALID_CONTENT_PATH, url);
}
parseResults.wrappedURL = "?";
parseResults.wrappedURL += subParseResults.protocol;
if (subParseResults.server != undefined) {
parseResults.wrappedURL += "/";
parseResults.wrappedURL +=  subParseResults.server;
}
if (subParseResults.wrappedURL != undefined) {
parseResults.wrappedURL += "/?";
parseResults.wrappedURL +=  subParseResults.wrappedURL;
}
parseResults.appName = subParseResults.appName;
parseResults.streamName = subParseResults.streamName;
return parseResults;
}
// get application name
endIndex = url.indexOf("/", startIndex);
if (endIndex < 0) {
parseResults.appName = url.slice(startIndex);
return parseResults;
}
parseResults.appName = url.slice(startIndex, endIndex);
startIndex = endIndex + 1;
// check for instance name to be added to application name
endIndex = url.indexOf("/", startIndex);
if (endIndex < 0) {
parseResults.streamName = url.slice(startIndex);
return parseResults;
}
parseResults.appName += "/";
parseResults.appName += url.slice(startIndex, endIndex);
startIndex = endIndex + 1;
// get flv name
parseResults.streamName = url.slice(startIndex);
} else {
// is http, just return the full url received as streamName
parseResults.isRTMP = false;
parseResults.streamName = url;
}
return parseResults;
}
/**
* <p>Compares connection info with previous NetConnection,
* will reuse existing connection if possible.
*/
private function canReuseOldConnection(parseResults:Object):Boolean {
// no reuse if no prior connection
if (_nc == undefined || _nc == null) return false;
// http connection
if (!parseResults.isRTMP) {
// can reuse if prev connection was http
if (!_isRTMP) return true;
// cannot reuse if was rtmp--close
_owner.close();
_nc = null;
initNCInfo();
return false;
}
// rtmp connection
if (_isRTMP) {
if ( parseResults.serverName == _serverName && parseResults.appName == _appName &&
parseResults.protocol == _protocol && parseResults.portNumber == _portNumber &&
parseResults.wrappedURL == _wrappedURL ) {
return true;
}
// cannot reuse this rtmp--close
_owner.close();
_nc = null;
}
initNCInfo();
return false;
}
/**
* <p>Handles creating <code>NetConnection</code> instance for
* progressive download of FLV via http.</p>
*
* @private
*/
private function connectHTTP():Boolean {
//ifdef DEBUG
//debugTrace("connectHTTP()");
//endif
_nc = new NetConnection();
_nc.connect(null);
return true;
}
/**
* <p>Top level function for creating <code>NetConnection</code>
* instance for progressive download of FLV via rtmp.  Actually
* tries to create several different connections using different
* protocols and ports in a pipeline, so multiple connection
* attempts may be occurring simultaneously, and will use the
* first one that connects successfully.</p>
*
* @private
*/
private function connectRTMP():Boolean {
//ifdef DEBUG
//debugTrace("connectRTMP()");
//endif
// setup timeout
clearInterval(_timeoutIntervalId);
_timeoutIntervalId = setInterval(this, "_onFCSConnectTimeOut", _timeout);
_tryNC = new Array();
for (var i:Number = 0; i < RTMP_CONN.length; i++) {
//ifdef DEBUG
//debugTrace("Creating connection " + i);
//endif
_tryNC[i] = new NetConnection();
_tryNC[i].mc = this;
_tryNC[i].pending = false;
_tryNC[i].connIndex = i;
_tryNC[i].onBWDone = function(p_bw:Number):Void {
this.mc.onConnected(this, p_bw);
}
_tryNC[i].onBWCheck = function():Number {
return ++this.mc._payload;
}
_tryNC[i].onStatus = function(info:Object):Void { this.mc.connectOnStatus(this, info); };
}
nextConnect();
return false;
}
/**
* <p>Does work of trying to open rtmp connections.  Called either
* by <code>connectRTMP</code> or on an interval set up in
* that method.</p>
*
* <p>For creating rtmp connections.</p>
*
* @see #connectRTMP()
* @private
*/
private function nextConnect():Void {
//ifdef DEBUG
//debugTrace("nextConnect()");
//endif
clearInterval(_tryNCIntervalId);
_tryNCIntervalId = 0;
var protocol:String;
var port:String;
if (_connTypeCounter == 0) {
protocol = _protocol;
if (_portNumber != undefined) {
port = _portNumber;
} else {
for (var i:Number = 0; i < RTMP_CONN.length; i++) {
if (protocol == RTMP_CONN[i].protocol) {
port = RTMP_CONN[i].port;
break;
}
}
}
} else {
protocol = RTMP_CONN[_connTypeCounter].protocol;
port = RTMP_CONN[_connTypeCounter].port;
}
var xnURL:String = protocol + ((_serverName == undefined) ? "" : "/" + _serverName + ":" + port + "/") + ((_wrappedURL == undefined) ? "" : _wrappedURL + "/") + _appName;
//ifdef DEBUG
//debugTrace( "_tryNC[" + _connTypeCounter + "] connecting to room: " + xnURL );
//endif
_tryNC[_connTypeCounter].pending = true;
_tryNC[_connTypeCounter].connect( xnURL, _autoSenseBW);
if (_connTypeCounter < (RTMP_CONN.length-1)) {
_connTypeCounter++;
_tryNCIntervalId = setInterval(this, "nextConnect", 1500);
}
}
/**
* <p>Stops all intervals, closes all unneeded connections, and other
* cleanup related to the <code>connectRTMP</code> strategy of
* pipelining connection attempts to different protocols and
* ports.</p>
*
* <p>For creating rtmp connections.</p>
*
* @see #connectRTMP()
* @private
*/
public function cleanConns() {
//ifdef DEBUG
//debugTrace("cleanConns()");
//endif
clearInterval(_tryNCIntervalId);
_tryNCIntervalId = 0;
if (_tryNC != undefined) {
for (var i:Number = 0; i < _tryNC.length; i++) {
if (_tryNC[i] != undefined) {
//ifdef DEBUG
//debugTrace("_tryNC[" + i + "] = " + _tryNC[i]);
//endif
delete _tryNC[i].onStatus;
if (_tryNC[i].pending) {
_tryNC[i].onStatus = function(info:Object):Void { this.mc.disconnectOnStatus(this, info); };
} else {
delete _tryNC[i].onStatus;
_tryNC[i].close();
}
}
delete _tryNC[i];
}
delete _tryNC;
}
}
/**
* <p>Starts another pipelined connection attempt with
* <code>connectRTMP</code> with the fallback server.</p>
*
* <p>For creating rtmp connections.</p>
*
* @see #connectRTMP()
* @private
*/
private function tryFallBack():Void {
//ifdef DEBUG
//debugTrace("tryFallBack()");
//endif
if (_serverName == fallbackServerName || fallbackServerName == undefined || fallbackServerName == null) {
//ifdef DEBUG
//debugTrace("Already tried to fall back!");
//endif
//it's not connected
delete _nc;
_nc = undefined;
_owner.ncConnected();
} else {
_connTypeCounter = 0;
cleanConns();
_serverName = fallbackServerName;
//ifdef DEBUG
//debugTrace("connect: " + _serverName);
//endif
connectRTMP();
}
}
/**
* <p>Starts another pipelined connection attempt with
* <code>connectRTMP</code> with the fallback server.</p>
*
* <p>For creating rtmp connections.</p>
*
* @see #connectRTMP()
* @private
*/
public function onConnected(p_nc:NetConnection, p_bw:Number):Void
{
//ifdef DEBUG
//debugTrace("onConnected()");
//endif
// avoid timeout
clearInterval(_timeoutIntervalId);
_timeoutIntervalId = 0;
// ditch these now unneeded functions and listeners
delete p_nc.onBWDone;
delete p_nc.onBWCheck;
delete p_nc.onStatus;
// store pointers to the successful xn and uri
_nc = p_nc;
_ncUri = _nc.uri;
if (_autoSenseBW) {
_bitrate = p_bw * 1024;
if (_streams != undefined) {
bitrateMatch();
} else if (_streamName.indexOf(",") != -1) {
var sSplit:Array = _streamName.split(",");
// remove leading and trailing whitespace from string
for (var i:Number = 0; i < sSplit.length; i+=2) {
var sName = stripFrontAndBackWhiteSpace(sSplit[i]);
if (i + 1 < sSplit.length) {
// If we have less bw than the next threshold or if
// there isn't another threshold (last string)
if (p_bw <= Number(sSplit[i+1])) {
_streamName = sName;
break;
}
} else {
_streamName = sName;
break;
}
} // for
}
}
// strip off .flv if included
if (_streamName.slice(-4).toLowerCase() == ".flv") {
_streamName = _streamName.slice(0, -4);
}
// if we need to get the stream length from the server, do it here
if (!_owner.isLive && _streamLength == undefined) {
var res:Object = new Object();
res.mc = this;
res.onResult = function(length:Number) { this.mc.getStreamLengthResult(length); };
_nc.call("getStreamLength", res, _streamName);
} else {
_owner.ncConnected();
}
}
/**
* netStatus event listener when connecting
*
* @private
*/
public function connectOnStatus(target:NetConnection, info:Object):Void {
//ifdef DEBUG
//debugTrace("_tryNC["+target.connIndex+"].onStatus: " + info.code);
//var stuff;
//for (stuff in info) {
//	debugTrace("info[" + stuff + "] = " + info[stuff]);
//}
//endif
target.pending = false;
if (info.code == "NetConnection.Connect.Success") {
//ifdef DEBUG
//debugTrace( "Connection " + RTMP_CONN[target.connIndex].protocol +
//			":" + RTMP_CONN[target.connIndex].port + " succeeded!" );
//endif
_nc = _tryNC[target.connIndex];
_tryNC[target.connIndex] = undefined;
cleanConns();
} else if ( ( (info.code == "NetConnection.Connect.Failed") ||
(info.code == "NetConnection.Connect.Rejected") ) &&
( target.connIndex == (RTMP_CONN.length - 1) ) ) {
// Try rearranging the app URL, then the fallbackServer
if (!connectAgain()) {
tryFallBack();
}
} else {
//ifdef DEBUG
//debugTrace( RTMP_CONN[target.connIndex].protocol + ":" +
//			RTMP_CONN[target.connIndex].port +
//			" onStatus:" + info.code);
//endif
}
}
/**
* netStatus event listener when reconnecting
*
* @private
*/
public function reconnectOnStatus(target:NetConnection, info:Object):Void
{
//ifdef DEBUG
//debugTrace("reconnectOnStatus: " + info.code);
//endif
if ( (info.code == "NetConnection.Connect.Failed") ||
(info.code == "NetConnection.Connect.Rejected") ) {
// Try the fallbackServer
delete _nc;
_nc = undefined;
_owner.ncReconnected();
}
}
/**
* netStatus event listener for disconnecting extra
* NetConnections that were opened in parallel
*
* @private
*/
public function disconnectOnStatus(target:NetConnection, info:Object):Void
{
//ifdef DEBUG
//debugTrace("disconnectOnStatus: " + info.code);
//endif
if (info.code == "NetConnection.Connect.Success") {
delete target.onStatus;
//ifdef DEBUG
//debugTrace("Closing myself");
//endif
target.close();
}
}
/**
* Responder function to receive streamLength result from
* server after making rpc
*
* @private
*/
public function getStreamLengthResult(length:Number):Void {
//ifdef DEBUG
//debugTrace("getStreamLengthResult(" + length + ")");
//endif
_streamLength = length;
_owner.ncConnected();
}
/**
* <p>Called on interval to timeout all connection attempts.</p>
*
* <p>For creating rtmp connections.</p>
*
* @see #connectRTMP()
* @private
*/
public function _onFCSConnectTimeOut():Void
{
//ifdef DEBUG
//debugTrace("_onFCSConnectTimeOut()");
//endif
cleanConns();
_nc = undefined;
if (!connectAgain()) {
_owner.ncConnected();
}
}
private static function stripFrontAndBackWhiteSpace(p_str:String):String
{
var i:Number;
var l:Number = p_str.length;
var startIndex:Number = 0
var endIndex:Number = l;
for (i = 0; i < l; i++) {
switch (p_str.charCodeAt(i)) {
case 9: // tab
case 10: // new line
case 13: // carriage return
case 32: // space
continue;
}
startIndex = i;
break;
}
for (i = l; i >= 0; i--) {
switch (p_str.charCodeAt(i)) {
case 9: // tab
case 10: // new line
case 13: // carriage return
case 32: // space
continue;
}
endIndex = i + 1;
break;
}
if (endIndex <= startIndex) {
return "";
}
return p_str.slice(startIndex, endIndex);
}
//ifdef DEBUG
//public function debugTrace(s:String):Void
//{
//	if (_owner != undefined) {
//		_owner.debugTrace("#NCManager# " + s);
//	}
//}
//endif
} // class mx.video.NCManager</td>
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教程分类

	基础操作		动画特效
	应用开发		组件学习
	As程序		动画教程
	Flash cs3		AS 3.0
	FCS/FMS教程		Loading教程
	Flash与Web		Flash教程连载

·FMS在线客服系统（客户端篇）
·FMS在线客服系统（fms脚本）
·FMS在线客服系统（分析篇）
·分布式的Flash Media Server: Gr
·基于SharedObject 3*3数组做的M
·用flex做的多服务器 im 及源码
·基于SharedObject做的房间列表和
·fms 2.0A星寻路
·FMS音频延迟分析
·fms中介函数