Example of a streaming player in a native Flash / Flex application
The following example shows how to play a video stream while simultaneously publishing another stream by using a client Flash application that can be run by a simple swf file. Streaming in this example can work via two protocols: rtmp:// and rtmfp://
The screenshot displays an example Flash application that sends a video stream to the server and plays the video from the server via the RTMFP protocol.
The 'Server' field contains the RTMFP address of the server to establish connection.
The 'Publish' field contains the name of the video stream to send to the server from the web camera.
The 'Play' field contains the name of the video stream to play from the server.
Below you can specify additional parameters of capturing and sending a video stream:
- width and height of a frame
- fps
- quality
- key frames per second
- is audio or video present in the video stream being sent
Example files
This example is a compiled SWF file embedded to an HTML page using Flex / ActionScript3 and MXML and accessible at:
/usr/local/FlashphonerWebCallServer/client/examples/demo/streaming/flash_client/streaming.html
streaming.html - example page
streaming/bin/streaming.swf - application file
Working with the source code of the example
To examine the code, let’s take this version of the streaming.mxml file with the hash 90eb5073687bbe63bbb7467de3f3be4f3fe33802 located here. The result of compiling the streaming.mxml file is the example application streaming.swf. The compiled swf and the source code are available for download in the corresponding build 0.5.3.1894.
1. Right after the application is loaded, we get access to the web camera and the microphone.line 42, line 44
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cam = Camera.getCamera();
videoMy.attachCamera(cam);
mic = Microphone.getEnhancedMicrophone(); |
2. Applying camera and microphone settings line 76, line 84
Recommended setting for the web camera:
- A motion sensibility threshold to send video:cam.setMotionLevel(0,2000);
Recommended setting for the microphone:
- Speex audio codec:mic.codec = SoundCodec.SPEEX;
- The number of frames per packet:mic.framesPerPacket=1;
- A sound threshold to send audio:mic.setSilenceLevel(0,2000);
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privatefunctioninitCam():void{
cam.setMode(int(camWidth.text),int(camHeight.text),int(camFPS.text),true);
cam.setQuality(0,int(camQuality.text));
cam.setKeyFrameInterval(int(camKeyFrame.text));
cam.setMotionLevel(0,2000);
Logger.info("Cam initizlized "+cam.width+"x"+cam.height);
}
privatefunctioninitMic():void{
varoptions:MicrophoneEnhancedOptions =newMicrophoneEnhancedOptions();
options.mode = MicrophoneEnhancedMode.FULL_DUPLEX;
options.echoPath = 128;
options.nonLinearProcessing =true;
mic.codec = SoundCodec.SPEEX;
mic.encodeQuality = 5;
mic.framesPerPacket=1;
mic.gain=50;
mic.setSilenceLevel(0,2000);
mic.enhancedOptions = options;
Logger.info("Mic initialized");
} |
3. Connecting to the server.line 103
Here we establish connection to the server and send obj.appKey = "flashStreamingApp";
This appKey tells the server, that it should deal with a Flash application, not with a Websocket/WebRTC client
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privatefunctionconnect():void{
trace("connect");
varurl:String = connectUrl.text;
nc =newNetConnection();
//if (url.indexOf("rtmp") == 0){
nc.objectEncoding = ObjectEncoding.AMF0;
//}
nc.client =this;
nc.addEventListener(NetStatusEvent.NET_STATUS, handleConnectionStatus);
varobj:Object =newObject();
obj.login = generateRandomString(20);
obj.appKey = "flashStreamingApp";
nc.connect(url,obj);
} |
4. Sending the stream to the server is performed in the publish() method of the example. line 165
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if (publishAudio.selected){
initMic();
publishStream.attachAudio(mic);
Logger.info("Init audio stream")
}
if (publishVideo.selected){
initCam();
publishStream.attachCamera(cam);
addH264();
Logger.info("Init video stream");
}
addListenerAndPublish(); |
5.Directly before sending the stream, we set additional buffering parameters and parameters of the H.264 codec in addH264() and addListenerAndPublish() methods.line 199, line 208
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privatefunctionaddListenerAndPublish():void{
publishStream.videoReliable=true;
publishStream.audioReliable=false;
publishStream.useHardwareDecoder=true;
publishStream.addEventListener(NetStatusEvent.NET_STATUS, handleStreamStatus);
publishStream.bufferTime=0;
publishStream.publish(publishStreamName.text);
}
publicfunctionaddH264():void{
varvideoStreamSettings:H264VideoStreamSettings =newH264VideoStreamSettings();
videoStreamSettings.setProfileLevel(H264Profile.MAIN,H264Level.LEVEL_3_1);
publishStream.videoStreamSettings = videoStreamSettings;
} |
6. Playing the stream starts after the play() method is invoked. line 223
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privatefunctionplay():void{
if(playStreamName.text == "") {
playStatus.text = "Empty stream name";
playStatus.setStyle("color","#ff0000");
return;
}
playStatus.setStyle("color","#000000");
Logger.info("play");
subscribeStream =newNetStream(nc);
addListenerAndPlay();
} |
7. All settings and buffer sizes are set directly before playing in the addListenerAndPlay() method. line 244
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privatefunctionaddListenerAndPlay():void{
subscribeStream.videoReliable=true;
subscribeStream.audioReliable=false;
subscribeStream.useHardwareDecoder=true;
subscribeStream.addEventListener(NetStatusEvent.NET_STATUS, handleSubscribeStreamStatus);
subscribeStream.bufferTime=0;
varsoundTransform:SoundTransform =newSoundTransform();
soundTransform.volume=0.7;
subscribeStream.soundTransform = soundTransform;
subscribeStreamObject = createStreamObject();
subscribeStream.play(playStreamName.text);
videoFarEnd.attachNetStream(subscribeStream);
videoFarEnd.width = 320;
videoFarEnd.height = 240;
videoFarEnd.visible =true;
} |