I am really close I can feel it. I am doing something stupid I know. I have a file where one button simply controls stop and play of a looping audio stream. (by overlapping the buttons and making them visible or invisible depending on buttonClick) I have it where it does work for stop/start but I really need to to Play/Resume.
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The pause button in my MP3 player resets the track to the beginning, there is something wrong with my sound channel because when i trace the position it is always zero.[code]...
View 1 Repliesi want to stream like a tv channel a list of files.
in this case i have 6 files: s1.flv - s2.flv - s3.flv - s4.flv - s5.flv - s6.flv
this videos are 1366 * 768 pixeles. this videos are about 15 seconds every one, they are short videos.every video are about 700 kb
in the server inside the application file i have an app name: test nd inside test i have streams/_definst_
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I've modified and created a display for a stream that starts automatically. I used "ns.play(sample)" to start the stream, but creating a button and calling the ns.pause() doesn't work.
var video:Video = new Video(550,310);
addChild(video);
var nc:NetConnection = new NetConnection();
nc.connect("rtmp://myserver/vod");
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I can stream video RTMP just fine in JW Player by LongTail Media. The Pause seems to work, however using Windows 7's Resource Monitor I can see that the player continues to download data when the video is paused, it downloads for about 60 seconds (and does not seem to be longer for longer videos). I have found the same using the examples on longtailmedia's website, so I know it is not something I have done.
I want a way to pause the video and have it stop downloading, this will save us lots of bandwidth. But also have it continue where it left off if play is pressed again.
I checked the logs on our wowza media server, which indicates that the player actually waits 60s before it sends the pause command to the server.
LongTail's support say that the pause functionality uses the built in NetStream class and its behaviour is out of their control.
I have tested flowplayer online examples, and a couple of Adobe / OSMF examples and they all continue to download for 60 seconds after the video is paused.
A way to get this working in JW player would be best, but Are there any flash players that will stop downloading while paused?
If this is not possible with RTMP are there any technologies that will do this with a flash player?
So I followed a tutorial and got my video player working perfectly, then hit a wall when trying to replace the volume slider with volume up and down buttons.
attached is the current working video player, I didn't bother uploading the .XML and the .FLV files, but I will if you're interested.
Here is the code which handles the SoundTransform object, but I can't figure what to apply to my EventListener on the Volume Up and Down buttons.
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I am trying to create a button to switch the sound in an embedded movieclip (mc_1) on and off. I believe should be able to use something like
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I have this code, and it works perfectly fine for the most part:
soundTrans.volume = 0.1
What this does, it plays the Sound at about 10% of its total volume.But about 1 out of 10 times, there seems to be a bug, that causes it to play 100% of its volume for like a split second then it goes down to the 10%.Like if the soundTransform is applied after the sound has already started playing, and the Player just sometimes cant synchronize it perfectly? I tried to play around with trying to set the soundTrans BEFORE starting the channel, like:
soundChannel.SoundTransform = soundTrans;
but to no effect, it seems the player Applies the soundTransform always just when the Sound starts playing.(or actually just a bit after)This is especially annoying if you are trying to Fade in a music slowly maybe over the course of 5 seconds, and then it starts with a big BANG, and then it goes down, then it fades in, ruining the whole effect.
Trying to reduce frame overhead in my game, im trying to figure out how to speed it up.These are some 'simple' questions if you know your way around sounds and channels so i'm hoping you'll read on and perhaps give me some insights.
I'm noticing that playing soundclips sometimes make it stutter so im looking for the best way to use sounds and channels.The main question is below: how costly is a soundtranform.volume and is it persistent on a channel?secondairy; does it matter howmany soundclips i drop into the same channel?What i noticed: The channel.soundtransform.volume only works after playing the sound. Q: How costly is a channel.soundtransform.volume?&; Will it stay this volume on the next sound that plays? or do i have to reset it each time?
Also: playing a lot of sounds in 1 channel isnt a good idea.So i created a channel for each gameobject that could produce sound.On each sound play i calculate distance from main player and set volume accordingly.
1: I could just pre create like 10 channels set the volume from 0.1 to 1.0 accordingly then play each sound in the specific range channel This way i wouldnt have to do soundtransforms on the fly.If however every sound producing object would be at the same distance it would clutter 1 channel with a lot of clips to play at the same time.
2: I could pre create 10 channels as a queue. On every play i would pick the next free channel, looping & wrapping through them as i go.
Here however i will have to set the soundtransform volume again on each play according to distance.
Somehow my class is giving me the following error, after I added a SoundTransform object.What am I missing The issue is somewhere with the
private var soundTrans:SoundTransform;
soundTrans = new SoundTransform();
Why would that be happening?
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I am having trouble with my flash program. im trying to make a media player but the tutorial that i am using told me to enter SoundTransfor into the action scripts but when i run the program all i get is an error... 1046: Type was not found or was not a compile-time constant: SoundTransform.
View 1 RepliesI have a video chat application where there can be 6 participants. What I would like to do is give the option to each publisher to control their own stream's volume. The code I have looks like this.
[global]
private var volumeTransform:SoundTransform;
[In the init method where ]
outgoingStream = new NetStream( nc );
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A possible solution can be this : Calibrate the slider and set the microphone gain, a gain of zero effectively is mute and a gain of 100 is full volume.
i'm trying to play a sound from left to right using actionscript3. i have putted in the first keyframe the code. what i have wrote is:
Code:
var soundClip:Sound;
var channel:SoundChannel = new SoundChannel();
var trans:SoundTransform = new SoundTransform();
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I am trying to create a pause button that will pause everything on the screen including movieclips/audio. Right now I can't figure out how to pause the movieclips.
View 2 RepliesI have some banners I am doing right now and have a pause timer question. I am fairly green at coding. In my first frame I have this:
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I want it to pause each time in the last frame for 8 seconds and loop only 3x. Is there a better way to write this? I know all my code should be in the first frame but I still suck.
I am using Macromedia Flash Pro 8. I have a flash intro that has words (phrases) which slide in. I would like to add a 7 second pause between each phrase to give people time to read them (no buttons). Could someone tell me the script(s) to use with all functions, etc. included - as I am so new to all this. I have been looking for weeks & tried many codes but none seem to work - or I don't know exactly where to place them - or both
View 3 RepliesI have a timeline of 30 frames, each it's own mc (page01_mc, etc.). In each mc I show a picture or two and hear narration.I have a first, prev and next nav on the main timeline. I need to add a play/pause btn that will pause both pictures and sound. I assume I do that in each mc, but do not know where to find the code.Here is what is in each mc now:
var mySound:Sound = new Sound();
mySound.load(new URLRequest("english/Intro01.mp3"));
mySound.play()
I am using the following code, in the last line when I play the sound I can still hear it even when I have transformed it to 0.
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close another live stream B if opening when stream A is closed? Does FMS support this kind of logic?
View 8 RepliesOnly 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);
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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
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I build a client side application where is only a FLVPlayback2.5 component and a short AS3 script.
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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.
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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]...
View 4 RepliesI'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:
vid.attachCamera(null);
vid.clear();
ns.attachCamera(null);
ns.attachAudio(null);
ns.pause();
What could be causing this? Is the stream cut off too soon? Is the client unable to send buffered data to the server?
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.
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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.
View 3 RepliesI'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.
View 0 RepliesCan I re-stream a stream from wowza to fms 4.5?I have a wowza stream with the url similiar to URL... Is it possible to restream this to fms 4.5?
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