ActionScript 3.0 :: URLStream Object Does Not Have A Stream Opened
Dec 14, 2009
I successfully built a XML gallery; everything seems to work fine. But everytime I click on a thumbnail, I will get the following error:Error: Error #2029: This URLStream object does not have a stream opened.I researched on this error, and it seems this error involves sounds. Something about trying to load something that's already fully loaded. Therefore, I'm confused since I don't have any sound in my gallery. On top of that, the gallery works just fine but this error keeps popping up in Flash.[code]
I successfully built a XML gallery; everything seems to work fine. But everytime I click on a thumbnail, I will get the following error:
Error: Error #2029: This URLStream object does not have a stream opened.
I researched on this error, and it seems this error involves sounds. Something about trying to load something that's already fully loaded. Therefore, I'm confused since I don't have any sound in my gallery. On top of that, the gallery works just fine but this error keeps popping up in Flash. Does anyone have information about this error and how to resolve this?
I'm using URLStream to download 30-80mb files over HTTP. It downloads fine on Windows XP, but is extremely laggy in OSX. Anyone had similar issues? Here's some snippets of the code.
I'm trying to open a connection to a URL that will periodically send over a JSON object. I know that I need to use URLStream and some event callbacks, but I'm in the dark when it comes to knowing how streams of data "work," or the general flow of operations. Assume I know how to open the connection and send the correct request headers. First, some relevant code:
I am trying to upload and download a file from the server. After uploading if I Try to download the file it is not downloading. URLStream.bytes Available returns incorrect value. If I close the AIR application and try it ,it will work perfectly.. Why URLStream.bytesAvailable returns incorrect value just after uploading..
In my AIR app,I am trying to implement a file downloader using URLStream.
public class FileDownloader { // Class to download files from the internet Function called every time data arrives[code].....
I simply create an object of the above class and passing the url and the file and call the load function. For some files I get the following error.
remotePath: http:[url].....error while writing bytes from...Error:Error #2029: This URLStream object does not have a stream opened.Which means the error is from the file stream(fileAccess) that I am using. I am unable to figure out why this could be happening. If I try to open the url http:[url]... in the browser, it opens properly. This happens randomly for some files.
I'm attempting to connect to an existing Comet-style long-poll service using an AIR app. The service expects a client to make a GET request with a Connection: Keep-Alive header. This request will remain open for long periods of time so that the server can push data through. In my app, the connection is terminated with an IOError after 30 seconds.
Is this an undocumented limitation of URLStream? A restriction on adl (I've only been running my app through adl)?
The server does not send any "keep-alive" messages to the client but, unfortunately this is not something i have control over.
Update
To test this, I've set up a stripped-down version using a little php script (linked by leggetter below) and am hitting it from a simple AIR app. I'm finding that my connections are closed after 30 seconds whether I use URLStream or URLLoader. the PHP:
<?php set_time_limit(0); sleep(40);
[Code]....
If i adjust the initial sleep time in the php script to anything over 30 seconds, the IOError event is triggered. If I lower the sleep time, but the request continues adding data past 30 seconds, the onComplete event is called, but _urlLoader.data is empty.
The only way this process completely successfully is if the entire thing is over before 30 seconds elapses.
In my case I need to be able to cancel loading as the user could possibly be scrolling through alot of images and the if not canceled it can take a few minutes to catch up to the scroll position.So after research the URLStream class seems to cure this problem and I found a good streaming class hereThis class works very nice except when there is heavy load on the server causing lag.When this lag happens the class will dispatch the complete event yet the image will not show.Watching the download with HTTPFox shows the image in fact does get loaded.
I assume there is some kind of delayed issue with the URLStream class and the author of this class tried to correct it using the setTimeout method. I also tried delaying the dispatching of the complete event using a timer and not dispatching until if( this.width != 0 ) was detected and this seemed to work much better.But it is still haunting me just not as often.
I have to be able to cancel and kill a Loader process properly and many times. If I dont do this, it seems that the flash player (at least on mac) eventually crashes as some loader processes are flying about , strangling the CPU.
There doesnt seem to be a straightforward way to know this and just calling a Loader.close causes an error, if the stream is not open.
Is anyone else aware of the fact that Internet Explorer (at least version 8) keeps downloading a file even if the close() method is called on an URLStream instance? Heck, it even keeps downloading if you close the tab displaying the site which called the load()/close() methods on an URLStream instance. Does anyone know of a solution to overcome this problem? More details: My preloader is loading a dozen of files and immediately closing the download streams with the purpose of checking each resource's file size. In Mozilla Firefox all goes well, exactly as expected, but Internet Explorer keeps downloading the resource even if the stream has been interrupted by the action script close() method.
I'm writing a Comet-like app using Flex on the client and my own hand-written server.
I need to be able to send short bursts of data from the client at quite a high frequency (e.g. of the order of 10ms between sends).
I also need the server to push short bursts of data at a similarly high frequency.
I'm using NetConnection.call() to send the data to the server, and URLStream (with chunked encoding) to push the data from the server to the client.
What I've found is that the data isn't being sent/received as soon as it's available. For example, in IE, it seems the data is sent every 200ms rather than as soon as NetConnection.call() is called. Similarly, URLStream isn't making the data available as soon as the server is sending it.
Judging by the difference in behaviour between the browsers, it seems as though the Flash Player (version 10) is relying on the host browser to do all the comms. Can anyone confirm this? Update: This is very likely as only the host browser would know about the proxy settings that might be set.
I've tried using the Socket class and there's no problem with speed there: it works perfectly. However, I'd like to be able to use HTTP-based (port 80) connections so that my app can run in heavily fire-walled environments (I tried using a Socket over port 80, but that has its problems).
Incidentally, all development/testing has been done on an internal LAN, so bandwidth/latency is not an issue.
Update: The data being sent/received is in small packets and doesn't need to be in any particular format. For example, I might need to send a short array of Numbers, and this could either be encoded in AMF (e.g. via NetConnection.call()) or could be put into GET parameters (e.g. using sendToURL()). The main point of my question is really to see whether anyone else has experienced the same problem in calling NetConnection/URLStream frequently, and whether there is a workaround (it's also possible that the fault lies with my server code of course, rather than Flash).
This is absolutely driving me crazy. While I'm a fan of the availability of asynchronous calls in AIR, I'm finding that being forced to use them for something that should be SUPER simple is a severe limitation. So severe that I may end up abandoning AIR and writing native Android and iOS apps instead of using the shared AIR platform.
OK, now that I have that off my chest, here's what I'm trying to accomplish. I have an app that, when deployed, is relatively small. But once deployed to a user's device that user will log in using a login name/password. Once they log in, content specific to that user needs to be downloaded and saved to the local device. Since the content varies by user I can't include it in the package for deployment.
But I cannot figure out how to accomplish this: say I want to download 10 files and each file is 2-3mb and I want to show a "Downloading, please wait..." view during the download. The application cannot proceed until all 10 files are downloaded. But since from what I've seen URLStream and URLLoader are both async I cannot figure out how to block the app from opening the "View available content" and say on the "Downloading, please wait..." view.
My question is regarding the use of RTMP files (delivered via a Flash Media Server) and Papervision; specifically using an RTMP stream as a texture on a Papervision object. I've already developed a number of successful methods for applying progressive flv/f4v files to Papervision objects. The RTMP streams, however, have presented a number of obstacles.
I will bypass the myriad of research I've done looking into this issue and say that others have inquired about this as well - but from what I can tell, no solid answer or solution has been found - especially under Flash Player 10. So, here's what we have so far:
1. A Flash Media Server running on a University server.
I'm trying to append a live stream to an existing file, but I cant figure out how to open the existing file and then append the live stream. This is server side as well.I know I need to open the existing file using stream = stream.get("file"). but I don't understand how to associate stream with the live stream coming in.
We need to call different instances of youTube Video into our own swf and play them.
Now as youTube video is from a different domain, the Flash sandbox restrictions naturally come into play. We am loading the youTube swf into a clip using the MovieClipLoader Class.
If we remove the instance using the removeMovieClip() command the video is removed but the sound continues. We have even tried the MovieClipLoader unloadClip() method but that too did not yield results.
We wish to be able to kill the sound object or atleast mute it so that it does not play even after the movie clip containing the video is removed.
As the swf belongs to a different domain we are unable to control the sound object directly through script.
We have even tried loading another youTube instance into the same movie clip after first removing it using the removeMovieClip() command, then creating it again, and finally loading the youTube instance using the MovieClipLoader class, but the earlier sound still continues to play even while the second clip sound also plays.
Only just getting started on this whole domain of learning, so go easy!If I set up a P2P video/audio chat (similar to the sample VideoPhone thing on the Cirrus site), can I get the stream from both parties to send to a server at the same time so that I can record it? If so, would I have to use a FMS to stream it to and perform the recording (and if so which version could I get away with)? Are there any (preferably free, or just tutorialised) solutions for the recording side of things?
Currently it seems like the only option for doing the P2P thing is to use Stratus/Cirrus unless I use FMS4 Enterprise.
how effective this kind of situation can be, in terms of quality of the stream and recording? Does any of this make sense?
I've had FMS running on my local machine for a while and have had a little experience writing FMS apps, but I've just tried recording audio for the first time using the standard vod application and I keep getting a "Write access denied for stream" error. My AS3 code is copied and pasted for various examples and am confident that it works.
I'm running Windows XP service pack 3 & FMIS 3.5.
I've had a look at the vod/media directory and under windows->properties the read-only attribute is ticked. Every time I un-tick this it reverts back to being ticked. I've googled this and MS say that most programs ignore the read-only attribute and that it only really applied to files. I've also tried the MS fix for setting the read-only attribute via cmd and still no joy (doesn't fix read-only attribute or FMS recording the audio after setting via cmd).
I've also tried our dev server install of FMS (running under linux) and am getting the same results.
Here's my AS3 code...
private function initApp(event:Event):void { removeEventListener(Event.ADDED_TO_STAGE,initApp);
i test the fms 4 update 1 rtmfp streams multicast after 10 minutes i get this message RTMFP Multicast stream has exceeded max duration allowed; closing stream. but i do not use IP multicast
I build a client side application where is only a FLVPlayback2.5 component and a short AS3 script.
[Code]....
My Encoder is setup with three streams: Vid: 500 kbps - Audio: 48 kbpsVid: 800 kbps - Audio: 48 kbpsVid: 1500 kbps - Audio: 48 kbps I start the encoder and everything looks fine in the log. In my browser (Safari or Firefox) I go to my html site and the stream starts after 6-8 sec. But anytime with the lowest bitrate 548 kbps and nothing look like the stream is switching to another bitrate. I tried it with the smil playlist and the result is the same. Only the lowest bitrate is plublished.
I have recently installed FMIS 3.5.3. In checking the access logs I find data in both logs that display the same stream stop and stream play time .I'm not sure why the time is the same (00:19:27 example below). Videos play fine when testing from work (T3 connection). However, occasionally a very slight hesitation when playing video from home (I have cable connection). [code]...
I'm having a problem with recording a live webcam stream. The last few seconds of the stream is getting cut off. The recording is stopped with the following piece of code:
I am having trouble getting audio stream meta data from an Akamai FMS stream. Everything is undefined and I'm not sure why. I am hoping maybe someone will notice something that I am overlooking. The stream is connecting and playing without a problem I just can't seem to figure out why all the meta data is undefined.
I have a layout with narration and a nav bar. When I click a nav button for section 2, the audio from section 1 (set to stream) continues to play over the audio for section 2. This cumulates so if I click buttons for sections 3, 4 and 5, I get five audio files playing on top of each other. Sections are individual movie clips with embedded audio streaming on a Sounds layer in each movie clip.
I'm trying to stream a HDS live multi-bit stream, it seems to push to the FMS but my player doesn't display the stream.Are these settings and files correct? The documenation is confusing on what and which files need to be edited and/or created.
Encoder settings: Bit Rate: 150,500,700 FMS URL: rtmp://myserver/livepkgr Stream: liveevent%i?adbe-live-event?liveevent
FMS 4.5
I see the following directories being created when I start encoding and each directory has a single file with a .stream extension in them. Are these correct? fC:FMS-HOMEapplicationslivepkgrevents\_definst_liveevent1[code].....
I am very new to action script. I want to stop the stream from publishing then later on i want to publish the same stream again. My problem is that I have 3 levels of stream processing (Ingress-Rename ProxyLimelight-AuthProxy-Live) before it goes live. User wants to stop stream from publishing and then publish the same stream when he wants in each level. User does not want to start the stream from the begining, he wants to restart the stream on the same where he stopped.
I have some problems with flash display in the browser. If i open with the browser a file .swf it appears smaller than the original version. I can see it in the correct dimensions only if I use the Full Sceen option (F11). How can I correct this problem to let the user get the original version?
