Flash :: Professional - RTMP / FMS & Video Audio Echoing?
Jan 27, 2010
Our client has encountered a bizarre problem where they hear an audio feedback or some sort of "duality" when playing an F4V file through FMS. When I tested this myself, I did not encounter this issue. I'm wondering if anyone here can provide some insight as to some of the potential causes of this issue and how we can fix it. We are using
1) FLVPlayback 2.5.0.15
2) Application targets the Flash 10.0 runtime
3) We are using a standard FMS 3.5 install, videos located in the vod folder
4) Trying to playback F4V files. The samples provided by FMS also exhibit the same issue.
On another note, I built a bare-bones Flex application (SDK version 3.5.0.12683 and 3.4.0.9271) using the VideoDisplay component and it always has the duality problem. It even gets worse as I refresh the page, the echoing / multiple audio playbacks seem to compound.
I have an mp4 file with the audio sounding perfect that I'm running through Adobe Flash CS3 Video Encoder to convert to an flv which I'm trying to then embed in a fla.My problem comes in the fact that after its embedding when previewing it, the sound quality has -significantly- deteriorated to the point where it sounds crackly and echoing. I've tried changing the sound data rate to maximum, but it makes no difference at all.
Writing RTMP Streaming Server for streaming AVC+AAC video. And it works fine with rtmpdump. But I can't force it to work in flowplayer and other flash video players.The message sequence after handshake is similar to FMS / RED5 / erlyvideo / haxevideo servers: I've tried a lot of variations.
From Chrome debug console I can see, what all negotiating messages passed to the flowplayer. The last one is onMetaData. And after this the working sample (rtmp://flash.tvwmedia.net/LiveVideo//Live300) gets NetStream.Buffer.Full. And streaming from my server don't get it.
I'm starting with AVC Header message, containing sps/pps. After it first AVC picture passed. After - AAC header and AAC sample. And then AVC/AAC samples. This dumped OK by rtmpdump - I have working flv on exit. But flowplayer and others does not work.
I'm trying to make a software which sends video and audio data to a flash media server by using RTMP protocol. Currently, my program can communicate with a flash media server correctly. RTMP specifications does not describe about the raw data in video/audio messages, so I muxed raw H.264 and AAC data into video/audio messages and sent to the server. The server seems to accept them, but a video player cannot playback the stream sending from the server. The player just says "Loading..." For a test purpose, I sniffed the network packets between Wirecast and the flash media server and ripped off only video and audio data. Then, I muxed those data into video/audio message and sent to the flash media server. In this case, the video player connected to the server can playback the stream correctly.
I checked the stream sent from Wirecast, the stream seems not to be H.264 raw data because those data are not started from 0x17 instead of H.264 start code. With those situation, I am wondering what kind of container format I should use for H.264/AAC data to the flash media server.
I am trying to publish a video to an RTMP Server but it doesn't publish. It might be a pre-release bug. I am able to play a NetStream but not able to publish one. Is there a sandbox issue? I'm not sure. Because, the FMS RTMP Server will not let a client connect unless it has been downloaded from the same host as the server (something like a sandbox condition)
I have a Flash animation that starts with an audio clip imported. When that's done it goes to a video. After that another audio clip plays, but I can't get it to start when it's supposed to. It keeps coming in too early even though its keyframe is after the movie ends. In fact I had to move the audio's keyframe about 1500 frames past where it should be to get it to come in at the right time. My Flash movie is 24 FPS. The .mov file being referenced in the FLVPlayback is 23.98 FPS. What's going on here?
I immediately fired firebug in firefox but surprisingly the video source is not in the requests.tp://hwcdn.net/m7n9i8d5/fms/videos/5_Standard_Zipper/B_Overview_of_Zipper_Types.flv.smil is the last request that is being made. response being:
I'm trying to stream an audio clip via rtmp in as2. I'm not particularly attached to the method, but this is the starting point I am at below:
Code: var s:Sound = new Sound(); var req:String = "rtmp://path/filename"; //doesn't work var req:String = "[URL]"; //This works s.loadSound(req, true);
I need to get an RTMP audio stream working, does anyone have any experience with this? (Also not sure if this goes in the actionscript folder or the sounds folder.
Is it possible to return php data without echo'ing print'ing the results in the PHP script? I need to find a secure way, to get data from a PHP script, so that the user can't sniff it or just read the script.
Usually, flash streaming is done by capturing webcam video/audio and streaming using NetConnection and other objects to servers like FMS,Red5,Wowza etc.I haven't found any example on how to create your own stream of images and stream as a video to the server.I know it would be possible to convert the image to bytes and send via SharedObjects. Then decode on server and create a video file on the server (e.g. using ffmpeg), but I would rather do it in realtime on the client side if possible.
(1) An ec2 instance with an SWF on it - this SWF plays streaming video - i.e. is a video player like JWPlayer (2) A streaming video distribution set up via Cloudfront
If I stream the the video via RTMP from Cloudfront to the SWF (which is on ec2) - would I incur charges for data transfer into the server (i.e. for data being read by the SWF) and out of the server (i.e. for data being displayed by the SWF to the user) on account of streaming the video to users (assuming that data transfer into and out for the server is being charged for)?
When I go to test my video the audio and video are playing at different playback speeds. I'm trying to create a video using Flash that requires events based on the song itself. how to make the audio and video play at the same speed during Timeline playback and testing?
Am developing an application with Flex 4.5 & Flash Media Server. I need a visualization curresponding to the plaing track . It is possible with SoundMixer.computeSpectrum(bytes, true, 0) in case of progressive downloading. But not working with rtmp streaming. And also need audio wave curresponding to the track which is also working with progressive ownloading usinf Sound object.
I have this streaming video player with an asyncErrorHandler function which isn't working correctly.In Flash player 10 I get no errors, but in Flash player 9 I'll get a popup window with the error message that shows up in my output window when I test on my local machine.
Error #2044: Unhandled AsyncErrorEvent:. text=Error #2095: flash.net.NetConnection was unable to invoke callback onBWDone. error=ReferenceError: Error #1069: Property onBWDone not found on src.display.VideoClass and there is no default value.
I feel that I have my AsyncErrorEvent setup correctly however, but am unsure as to why I still get a massive error in my output window, here is the popup window and code below:My statusEventHandler function
// ☼ --- Handle Status and Errors function statusEventHandler(event:NetStatusEvent):void { trace("connected is: "+ncConnection.connected); trace("event.info.level: "+event.info.level);
there is a way to record the stream of webcam without use media server or similars servers? with flash or air can do that? record both audio and video?i found a air class that record video capturing bitmap data of each frame and saves to a byte array but no record audio
when a video is playing..and i click a button to play to a different label in the main time line..the video component disappears but the audio keeps playing.. all so the volume component, the slider indicator stays stuck to the stage.. what AS3 is needed..to stop the audio and fix the indicator..
I am making my second video, so I'm not too knowledgeable. Anyway, I imported audio into my video and it plays back fine. So I play the video, and pause it to try and synchronize the audio. The video stops, but the audio keeps playing, and it will not stop until the whole thing has played. If I try to start from the beginning again and play the audio to sort of override it, it just overlaps with the original audio, and the resulting noise gives me a headache. I don't know if somehow I can stop the audio a portion of the way through
I am using Flex and FMS to develop a video conferencing application. I am using DynamicStream to automatically switch incoming streams to the appropriate bitrate depending on the available bandwidth. I was wondering if there is anyway in which I can instruct the DynamicStream to allow more bandwidth for audio (obviously at the cost of losing some frames from the video).
making a simple game with video stills matching a running audio track.my audio is an MP3 i built in GarageBand. all works well in flash, but when rendered to QT, the audio and video go out of sync, and some video drops out too.have put all audios on their own layer and streamed them all too, set settings in Publish to 64, but still have same problem
I have a SWF movie (only animation, no Actionscript) and I would like to programatically extract the resulting video and audio and whatever resolution.
Let me preface this by saying: This is definitely a bug either with latest Adobe Flash Player or the streaming Media Server (in this case Real Helix):In the latest version of the Flash player (10.1.82.76) Code that worked previously in version 10.1.53.64 does not work now. This is basic code to connect to an RTMP server (Real Helix) using Netstream class. NOW the video does not even display and there is no error message . THIS ONLY OCCURS in a BROWSER . Also I am using swfobject to display the flash and I tried with latest version and the old 1.4.1 version.
private function init():void video = new Video(); NetConnection.defaultObjectEncoding = ObjectEncoding.AMF0; nc = new NetConnection(); nc.client = this;
I ran a simple live video streaming application for the first time with actual users and ran into a couple of serious performance issues that had not turned up during testing. In this instance there was one video stream from a live web cam and used FMLE at 150 kbps using VP6 and MP3 @22k. There were 16 clients and everything worked pretty good for about 30 minutes. (although some clients said their audio and video were out of sync by up to 3 seconds)
Then individual clients would have either the video freeze or the video would continue and the audio would stop. These clints had to "disconnect" and then "connect" again to the application. This happened to all of the clients at one time or another for several minutes. I stopped and restarted the FMLE with progessively lower bandwidth settings down to 75 kbps but still clients were having the same issue.
I eventually stopped the FMLE and used the applications built in publisher at 45 kbps and that seemed to eliminate the freeze/dropping issue. But of course the video quality was very poor and some clients still reported that the audio was out of sync with the video. The server hosting the FMS application is a quad processor dell with lots of memory and network connectivity. The Flash Media Admin Console performance graph showed the total Bandwidth as 3 Mbps at maximum.
I am using Lee's xml video playlist and it works great with http video links. I now need this flv player to stream rtmp link. Streaming one video from the rtmp server works fine but when using the xml list the video doesn't load.
Can someone point me to the right direction please? I have a video asset that is streamed from rtmp. Say, url is: [URL]..I need to invoke BitmapData.draw() on this asset.
There is crossdomain.xml file on http://blah.domain.com/crossdomain.xml that allows all the domains.
When is instantiate stream I write: stream.checkPolicyFile = true;
