ActionScript 2.0 :: Audio Delay Without Netstream?
Mar 17, 2011
I have been working on a voice-chat system in AS 2.0 using FMS 3.5 for a while now. I have been experiencing a lot of latency. Usually there's no latency to begin with but it grows over time to several seconds. Up until this point, I have assumed that this latency is due to a slow server/low bandwidth since I have tinkered around with all of the buffer settings (that I know about) already.However, today I decided to see what would happen if I just attached the audio of the microphone to _root. In other words, the script does not connect to FMS and there's no streaming going on at all. All I have is essentially:
However, even in this case I get a delay! In fact the same sort of thing happens: there's no delay to begin with, but it grows to approximately 1.5 seconds over time (very quickly in fact). In order to track down where it could be coming from, I decided to add a microphone activity clip that showed the activity of the mic. Perhaps not so surprisingly this activity clip was perfectly in sync with the actual activity of the mic (i.e. no delay). So it seems like there's a delay due to the attachAudio command.
I am working on a video conferance projectthere will be a lecturer and an interpreter.interpreter will write the translated text simultaneously.what I am trying to is add vocal translation but I also want to keep the original audio is it possible to attach multiple audio to netStream or do I have to create another netStream for 2nd audio??
instead of the audio coming from an mp3 file via aURLRequest, I want the audio to come from a NetStream originatingfrom a Flash Media Server.The ultimate goal is to have multiple netstreams coming ineach with their own audio amplitude bar being displayed.This is being written in Flex 3 using Actionscript 3."To control the audio associated with a video file, you canuse the DisplayObjectContainer.addChild() method to route the audioto an object on the display list; you can then create a Soundobject to control some aspects of the audio. For more information,see the DisplayObjectContainer.addChild() method."but theDisplayObjectContainer.addChild() method doesn'tprovide any further information and I cannot figure out how to usean AS3 Sound or SoundChannel object with a DisplayObjectContaineror a NetStream.
In Flex you can stream microphone audio to an FMS/Red5 server using NetStream.attachAudio, which requires a Microphone object. Is it possible to stream audio through the NetStream from somewhere other than a Microphone? For example, from a file/embedded resource?
The reason I'm asking is that I'd like to be able to run automated tests that don't require using an actual microphone.
just wondering if it is possible to play an FLV and attach the video from that netstream to another and publish it to FM2.
is there a a way to attach video and audio from one netstream to another? i have tried everything so far that i know of. when i think its attaching and publish it, i see nothing. publshing my cam and mic work flawlessly.
here's my code:
Quote:
nec = new NetConnection(); nec.connect(null); nes = new NetStream(nec); nes.play("http://testbox/transformers.flv");
I have an mp3 file that is mainly narration and I have a very simple soundSpectrum routine placing a soundwave. the issue is that the spectrum starts before the actual audio, so it is completely out of sync. it appears like it's buffering, so I am curious if there is a way to prevent the buffering and play right away or buffer the soundSpectrum.
I have an application that uses FMS 3.5 to stream live audio between 2 computers.Currently, the delay between when the person speaks and the time the other person hears it is about 2 - 3 seconds.This makes it difficult to have a conversation because one person regularly steps on the other.I am using a NetStream object that connects to the fms server.I then attach audio.What configurations can be set so that the least amount of delay is experienced between the two computers? Can fms ever relay audio as quick as something like Google Talk?
how to get flv's to load up dynamically using xml - it all works well apart from this one problem that i have.
when the thumbnail for the flv is clicked the preloader appears and the flv starts to load - but for some reason the audio begins to play almost immediately - then when the flv actually loads then the audio plays again whilst the initial (unwanted) audio carries on playing!
The task seems easy: I am loading a video via a netsream object (could be FLV or an h264 mp4) and I would need to check if this video has audio available (otherwhise sound control objects should be disabled or become hidden).
I have a project I'm working where I need a single media player to be able to play either audio or video depending on what gets passed (flv or m4a).
PHP Code: ExternalInterface.addCallback("playMedia", playMedia); // The way a/v gets changed is from a dropdown menu in the HTMLvar av:String; / The param that gets passed to my playMedia function which will determine if its audio or video var nc:NetConnection = new NetConnection();nc.connect(null);var ns:NetStream = new NetStream(nc);ns.client(this); [Code] .....
How do I add the audio to the container in such a way that it 'replaces' the video that's in the container? I may want to add a static image as a visual representation.*/ ns.play(av); }}
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 have a few animations that run on different scenes each time a scene is loaded the external audio should start with it. Well here is the thing the animation always starts before my audio files and each scene gets bugged. I need to create an action that only allows my scene to start with the audio and also starts loading the next audio so it wont be dellayed.
On the first scene I have the following code: scene 1 frame 1 var autoPlay:Boolean; if(!autoPlay){ stop();} scene 1 frame 2 animation starts and audio 1 loads import flash.media.Sound; [Code] .....
I would like use NetStream class to stream AAC audio file stored locally on server (because i can't use Sound class, AAC not compatible) How can i do this ? (without flash media server)
I have tried using http protocol, no errors occured, but i receive no audio data.
I'm streaming audio using NetConnection and NetStream. I know that you can modify sample data in real-time with the Sound object, however I cannot find the SampleDataEvent for audio playing with the NetStream object. Is there a way to pass the audio from the NetStream object to a Sound object and modify the sound at that object instead?
Basically I want to recreate a sound installation that I made a couple of months ago which used Logic Express and a Tape Delay effect to take ambient sound captured by a mic, delay its output by 3 seconds, and then play it into the headphones. All without recording anything and in realtime.I've basically got this far, which lets me hear the mic input through my headphones loud and clear. Now all I want to do is apply a 3 second delay to the input before it outputs. Do I have to send the audio to a buffer where it can be constantly stored and rewritten?
Code: //Create a new Microphone var mic:Microphone = Microphone.getMicrophone();
I'm making a small video players in AS3, and I've found that after calling NetStream.pause() or NetStream.togglePause(), no status messages are being fired any more.If I click the "pause" button while the video is buffering, I never get the Buffer.Full message.Here is some code:
_connection = new NetConnection(); _connection.addEventListener(NetStatusEvent.NET_STATUS, netStatusHandler); _connection.connect(null);[code].....
I am working on an app that will take a snapshot of an rtmp stream of an archived video, save it to a file, and ExternalInterface the file name to the hosting page to javascript it up for display.
I actually have all pieces working great. The only time an error is thrown is when I pause the video, seek to somewhere, and then try to take a snapshot. Then there is a runtime error of: SecurityError: Error #2123: Security sandbox violation: BitmapData.draw: mySwf.swf cannot access rtmp://myFlashServer/myApp. No policy files granted access.I've seen post after post of suggestions and have tried them, but without success.
I have the crossdomain.xml and my FlashMediaServer security setup fine (it is actually working, except for this one glitch). It just seems to be the pause-then-seek-then-BitMapData.draw() combination which is creating issues.
Another weird thing: after the error is thrown, if I resume the stream and take a snapshot, it works flawlessly again.
I want to create a function based (not frame) delay of about 10 seconds to the function at the start of a flash movie, the code only needs to run once.I'm very new to actionscript and have been given links to SetInterval and SetTimer examples but they all seem to be very complex.The function is below. I believe it's possible to add the commandTimer(delay:Number, repeatCount:int = 0) how me to a simple example.
I am using NetStream, NetConnection and Video object to play an mp4 file which is hosted over a web server using http.This is an AIR application and the relevant code is pasted below:
var url:String = <some http url>; connect_nc = new NetConnection(); connect_nc.connect(null);
In my application I have a video playing from a NetStream. Every second on timer I update a text label with statistics like stream.info.currentBytesPerSecond. The problem occurs when the NetConnection associated with this NetStream closes: the getter for stream.info throws
A NetStatusEvent with info.code of "NetStream.Connect.Closed" is dispatched by NetConnection when a NetStream is closed. This seems kind of weird, shouldn't the NetStream dispatch the event? how do I know which NetStream has been closed if I have more than one running (which I do)? I need to know so I can cleanup the now dead stream.
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.
Anyone know of a skinnable audio player (or something similar to Windows Vista audio control in task bar) that i can embed in my website?
A player that can be customized to use a 'speaker' icon instead of a 'play' icon. When the speaker is clicked, the volume can be adjusted or turned off. The option of being able to autoplay and set the default volume level.
I am converting from .FLV's to .F4V. Video outputs fine but the audio sound stops after a few seconds from the beginning.I have tried many different settings for the audio with no luck. Also, tried on a colleague's machine who is running CS3, which works fine, so something is buggy in CS4...
Because the player optimizes framerate, antialiasing etc dependent on the performance capabilities of the platform upon which it is running, it was nearly impossible to ensure animation syncronization w/ audio. If the audio were on the timeline you could set sync params to stream, but if the audio is loaded programically forget it .. well until AS3 ..The following code throttles the player framerate to lock step the animation to the audio .. not perfect but the principal works ..
Let me first establish what I want to do:My user is able to record voicenotes on my website, add tags to said notes for indexing as well as a title. When the note is saved I save the path of the note along with the other info in my DB.Now, I have 2 choices to do the recording, both involve a .swf embedded in my site:
1) I could use Red5 server to stream the audio to my server and save the file and return the path to said file to my app to do the DB saving, seems rather complicated since I would have to convert the audio and move it to the appropriate folder that belongs to the user in a server side Red5 app, which I'm not very aware of how to build.
2) I could simply record the audio and grab its byte array, do a Base64 encoding on it and send it to PHP along with the rest of the data that is necessary (be it by a simple POST or an AJAX call), decode it on the server and make the file with the appropriate extension, audio conversion would also occur here using ffmpeg, this option seems simpler but I do not know how viable it is.
I have an audio player that has a custom timeline and playhead. I am using a start and stop drag to move in the time line but the only way i can make it work is to have it play when ever I release the play head. I would like it to stay stopped if the player is paused and play if the player was playing.
