ActionScript 3.0 :: Make A New Request To The Server If The Seconds Parameter Specifies A Time Outside Of The Currently Buffered Video Data
Jul 14, 2011
when uing netsream.seek(), how can i make a new request to the server if the seconds parameter specifies a time outside of the currently buffered video data. I want to seek unloaded time
I can get which part of file in bytes is loaded using netstream.bytesLoaded,netstream.bytesTotal, I can get the current playing position using netstream.time. But I want to know how many seconds of video are already loaded (not the length of buffer, which remains constant).
I'm testing Live streaming with FMSS. The stream is pushed to FMSS buy Adobe FMLE software. Streaming works fine until I reach 1300-1400 simultaneous connections.No matter what the encoding ratebit is (150kbps or 2000kbps) the stream is no longer play from time to time (5-8 seconds).
The CPU (2xIntel Quad) is loaded less than 20% and memory used is about 2 GB (there is plenty of memory installed 32G). The OS is RedHat 5.3 64bit platform. Network uplink maximum rate is 4Gbps.I disabled the Queue but the problem still persist.
I am sending a request to a server. If i didn't get a response i need to send a request once again after 3sec. Like this i have to check for 3 times. If 3 rd time also it fails i need to terminate the request by prompting a msg to customer.
Is it possible to request some data in a Flash movie from PHP at run-time? Maybe my real-world implementation can clarify some things:
I use a Flash movie to store a Local Shared Object (because for some reason I need LSO's instead or regular PHP cookies). Now, when I load up a PHP file I want to somehow retrieve the data from the LSO at runtime, assign it to some variables, and use the variables through the rest of the script.
i want to develop video chat and there will be a admin and users. Admin can see their name on the screen and choose an user can start his/her cam play ?
am having this problem of sending XML request to an addresswhich starts with "https". The problem is I always get the errormessage "Error opening URL "https:......."", instead of the actualresponse which should be in XML format as well.The function I am using is "sendAndLoad()"requestXML and responseXML are objects of XML class . Whatfunction "sendAndLoad" above does is to post variables in the"requestXML" object to the specified URL "servelet_address". Theserver response is downloaded, parsed as variable data, and theresulting variables are placed in the responseXML object.However, as I stated above, I always get the error message.
i have a video in which I can seek to unload parts (Pseudo-Streaming), the server return the video and it loads it from the requested second,the problem, i don't want to make a new request to the server if a part has already been buffered, here's what I have in mind, but can't achieve
Code: if(netStream_has_been_download){ //Just run the seek() command
1) I want a 60 seconds video to start playing after 40 seconds have been downloaded - to do that I set the NetStream.bufferTime to 40 seconds and retrieve "NetStream.Buffer.Full" event causing the video to really start playing. This step is OK.
2) However, the "NetStream.Buffer.Full" causes data to stop downloading. So the remainder of the video begins to download no sooner than after the 40 seconds have been played. This step is my issue. Can anyone tell me how to avoid this unintended effect? (i.e. playing a video and downloading data at the same time?)
I've a site where consumer can take a live meeting with the beauty consultant. The problem I'm facing is during the video chat. Basically there is a delay of 5 seconds between Video and Audio when I access the site from out of my network but when I access it within my network (VPN) then it's work fine.
I've build a simple video chat application. The problem is that there is a delay of seconds between video display i.e. user1 video motion change displayed bit late at the user2 window. I'm using FMS 3.5 and FLEX.. IS THIS PROBLEM RELATED TO THE BANDWIDTH. My FMS bandwidth is 256kbps.
When recording a video to FMIS if the stream is buffered, does this get buffered in memory or does it use up local storage?
I am developing an application that will use a web cam to record videos which will then be submitted to the server. Ideally I would have liked to write the video to disk first and then upload later however, I understand that I can't do this with flash (my solution has to be in browser AIR apps aren't an option). I am ultimately concerned about video quality so in order to ensure that I loose as few frames as possible I'm planning on setting NetStream.bufferTime to a large value. I am concerned however that someone with a poor internet connection and a good webcam will quickly use up local storage assuming that it is set to the default of 100K.
If this is the case then I potentially need to think again about my approach - maybe even look into Silverlight 4, or at the very least try and explain to my users how to go into flash settings and set local storage to be unlimited.
I did a live stream last week using 282,482,832,1500Kbps streams. What would cause the audio to get out of sync with the live video stream? I'm trying to determine if it was bandwidth related, cpu/memory issue on the FMIS 4.5 server, or an issue with encoding PC exceeding it's limits?
Normally we use NetStream.Seek method but it will make the seek after the buffer area crossed the seek time length.(HTTP). Can we seek the video that is beyond the buffered area like youtube in red5. Will it start the buffer from the seek point.
The upgrade version of FMS 3.5.3 append the new function about buffered stream, in order to resolve problem when the bandwidth no good caused the video getting more and more slow, but I can't find the configuration in which file.
I'd like to start playing a web-based video at a specific time, say 2 minutes in, even if the video hasn't been downloaded that far. I thought I could just "seek" to that time but that apparently only works when the video is buffered first.
So for example, this: Code: nc = new NetConnection(); nc.connect(null); ns = new NetStream(nc); ns.play("[URL]"); ns.seek(123); video.attachVideo(ns);
Seems to work if you have the video cached or buffered or whatever, but if you don't it gives the "NetStream.Seek.InvalidTime" error, which makes sense. I want to know if it's possible to start playing a video at a specific time, even if the video hasn't loaded up to that point.
So I have FMLE and FMS. to record and stream video at the same time. On server side I have "main" application.onPublish = function(client, potok){ potok.onStatus = function(info){ for(var i in info){ trace(i + "=" + info[i]); [Code] .....
start a video playing from a precise point in time?Using rtmp protocol to stream a video:rtmp:\my-servervodmy-video.mp4is it possible to use a similar form, as the following:rtmp:\my-servervodmy-video.mp4 =00:20:21to start video file from that point?
I want to write application for facebook and vkontakte, which will consist of such main blocks as: SocialNetworkAPI (which include all work dedicated to social networks: posting to wall, get all user's info) and ServerAPI (which will send HTTP POST requests to my Java based server and receive data from there in JSON).
I am looking now to Adobe flash URLLoader
is there any good ServerAPI libraries, which I can use or rework to prevent rewriting standart code lines.
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.
why dynamic streaming taking too much time to switch video from lower bit rate to higher bit rate and vice versa. I am doing dynamic streaming in following ways -
var param:NetStreamPlayOptions = new NetStreamPlayOptions(); param.oldStreamName=oldStream(); param.streamName=newStream()
[code]....
I am using duel buffering and that is 3 seconds when video starts and 10 seconds when "NetStream.Buffer.Full". Video taking approximately 30-50 seconds to switch video and when I am calling the above code.