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背景

大家一定碰到過(guò)這種問(wèn)題, 在本地測(cè)試播放成功的 flv 視頻在服務(wù)器上無(wú)法播放, 并且直接訪問(wèn) flv 文件被告知找不到文件, 這是因?yàn)?IIS 6 的默認(rèn)配置中并沒(méi)有 flv 后綴的 mime 映射, 所以會(huì)產(chǎn)生 404 找不到文件的錯(cuò)誤. 通常的做法是修改 IIS 配置, 添加 flv 的 mime 映射, 問(wèn)題是解決了, 但如果因?yàn)槟承�條件限制無(wú)法修改配置怎么辦, 第一個(gè)想到的是修改成讓他可以被識(shí)別的后綴名, 可惜的是自帶的組件默認(rèn)情況下并不允許你那么做, 否則會(huì)報(bào)告錯(cuò)誤

player.contentPath = "water.swf"; // player 是舞臺(tái)中 FLVPlayer 組件實(shí)例名, 這里我把后綴名改成 swf, 以便可以讓 IIS 識(shí)別

引用：
報(bào)錯(cuò)
1005: Invalid xml: URL: "water.swf" No root node found; if file is an flv it must have .flv extension

解決： 使用自定義的 NCManager 替代默認(rèn)的 NCManager
判斷 flv 后綴的代碼在 NCManager 的 connectToURL 方法中, 我修改了他的實(shí)現(xiàn)從而跳過(guò)了這個(gè)步驟
NCManager.as 262 行處

原：

代碼拷貝框
if (parseResults.streamName.slice(-4).toLowerCase() == ".flv") { var canReuse:Boolean = canReuseOldConnection(parseResults); _isRTMP = false; _streamName = parseResults.streamName; return (canReuse || connectHTTP()); } else { _smilMgr = new SMILManager(this); return _smilMgr.connectXML(parseResults.streamName); }
[Ctrl+A 全部選擇 然后拷貝]

現(xiàn)：

代碼拷貝框
var canReuse:Boolean = canReuseOldConnection(parseResults); _isRTMP = false; _streamName = parseResults.streamName; return (canReuse || connectHTTP());
[Ctrl+A 全部選擇 然后拷貝]

我修改代碼跳過(guò)了這個(gè)判斷, 這樣即使不是 flv 后綴同樣能夠播放, 當(dāng)然帶來(lái)的副作用就是無(wú)法使用 SMIL 了.

最后我把修改了的代碼另存成 HttpConnectionManager.as, 同時(shí)類(lèi)名也改成 HttpConnectionManager, 放到 fla 所在目錄中

代碼拷貝框
//**************************************************************************** //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
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這樣場(chǎng)景中所有的 FLVPlayer 都使用 HttpConnectionManager 作為 NCManager 了. 不過(guò)等等, 這樣做是不夠的, 因?yàn)槲覀兊?HttpConnectionManager 還沒(méi)有被編譯進(jìn) SWF 中, 只要簡(jiǎn)單的加一句HttpConnectionManager 就可以了。

完整的代碼：

import mx.video.FLVPlayback;
import mx.video.VideoPlayer;
HttpConnectionManager; // 為了讓類(lèi)能夠被編譯進(jìn) swf
VideoPlayer.prototype.ncMgrClassName = "HttpConnectionManager";
var player:FLVPlayback;
player["_vp"][0]._video.smoothing = true; // 讓視頻能夠平滑播放的小 hack
player.contentPath = "water.swf"; // water.flv 后綴名改成了 swf

到這里, 我們的播放器就完成了。

教程及源碼下載：Player.zip

經(jīng)典論壇交流：
http://bbs.blueidea.com/thread-2726163-1-1.html 

本文鏈接：http://www.95time.cn/tech/multimedia/2008/6246.asp　

出處：藍(lán)色理想
責(zé)任編輯：moby

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