I want to publish a mp3 file from one of the peers and play it from other peers, very similiar to a RTMFP chat app.
From what I understand till now:
netstream.publish is used to publish a stream to RTMFP netconnection and netstream.play for playing a stream from other peers.
The steps for streaming mic and camera capture are:
netstream.attachCamera(cam);
netstream.attachAudio(mic);
netstream.publish('video');
However I don't see a way to publish (stream) mp3 files using Netstream. Note that using Netstream is essential since I want to "publish" audio to listening peers.
Please correct me if I was wrong above. Ideally what I want to achieve should be easily possible, yet I can't find any pointers for it. Is is possible to use ByteArray for the same. Any alternative streaming strategy would be welcome as long as it works with RTMFP. Links to code examples would be great too.
You stumbled upon one of the strange quirks of NetStream. It can publish sound from the microphone, but not from a arbitrary sound source. There is some workarounds, some more complex than the others.
Streaming through a virtual microphone. The easiest workaround, and the best (IMO) if your project allows you to use it. You just have to install a virtual microphone/camera software (ex : ManyCam) and use it to stream your mp3 file(s) through a virtual microphone. With that done, you just have to bind this microphone to your AS3 app. Sadly, it doesn't work for your project, since you can't reasonably ask of the publishing peer to install a virtual microphone.
Streaming usingSound.extract(), NetStream.send() and SampleDataEvent.SAMPLE_DATA. As you might know, NetStream.send() can be used to send messages to peers. The thing is, those messages are serialized and can be ByteArray. Thus, you can send audio data samples with NetStream.send(). The publishing pee apps can obtain the data samples with Sound.extract(), and the receiving apps can play them thanks to the SAMPLE_DATA event. One of the problems will be to know when you should send new samples. To manage that, you would recommend also using the SAMPLE_DATA in the publishing app and send new data each time the SAMPLE_DATA event occurs. The main issue with this method is that since you don't use the standard way of RTMP to stream audio, it needs a custom app for the receiver to play it. Given what you said of your project, it shouldn't be a problem.
Reproducing the RTMFP protocol using Socket. It would be long, very complex, and error-prone. I would never recommend to do this except perhaps as a learning experience. You would need to read, understand and implement most of the RTMFP Specification.
Related
I've spent plenty of time solving this problem, but it looks like I need some help. I have a web conference application which provides ability to stream live video, chat, share documents, draw on a whiteboard, share desktop, etc. And now I want to record everything that happens in taken separately so called webinar, including video and sound. So I'm looking for tools that can help with this goal.
Here's input data:
This is Adobe Flash based application
Using wowza server
Everything should be recorded on server
Many webinars can be in recording mode at the same time
Record should be represented in video (flv, mp4 or whatever)
What I've done so far and what I problems I have:
I have implemented recording on server side. But this is not a video, this is just a list of commands to recreate passed webinar. It works, but has lot's of limitations and problems with rewinding.
And now I'm testing this FLV Encoding library. I created AIR application that starts on server when record is needed, connects to taken webinar and takes screenshots from itself with BitmapData.draw() method. Works pretty neat, but has some limitation that I'm looking help with:
First of all, this is sound problem. I have no idea how to catch all
sounds from all sources in flash. So far from my tests and googling I conclude that SoundMixer.computeSpectrum() won't help me to do this. Maybe this can be done on server side by mixing all streams on the right time but I think this can lead to synchronization problems and I prefer to capture sound on client. Maybe there is way to capture audio byte array from rtmp stream somehow?
Security problems. We have 2 kinds of them. First ones are with streaming videos. BitmapData.draw() method throws exeptions even after adding <StreamAudioSampleAccess>true</StreamAudioSampleAccess>
<StreamVideoSampleAccess>true</StreamVideoSampleAccess> on server. There are lots of posts about this problem and no good solution.
But more complex problem is that YouTube videos can be opened in webinar using api player. And in this situation I have no idea how to resolve security problem. Maybe someone knows a way or workaround to use BitmapData.draw() on YouTube AS3 player?
Or maybe there is another good way to solve my recording issue?
Free Apache Openmeetings conferencing [1] has a java recording application inside which should work in 3.0 release. Just use it.
[1] http://openmeetings.apache.org/
I was wondering if someone has ever done something like this. I have a HD movie (or even 720p one) and I want to send it to a Flash client. I was thinking of using OpenCV in C++ for the decoding and sending part. I had even implemented some of this, but have problems with wrong packet size.
But my question is different, has anyone did anything similar to this? Can this give a chance for performance improvement? I have strong doubts about this, because I think the sending and decoding will be still difficult for the Flash machine. Looking forward to hearing some opinions from more experienced guys.
not a real answer, more like thoughts about your problem:
yes, you must encode HD images, sending 25 fps x 1.5mb over the net is a no-go.
gstreamer was build for exactly that purpose. complicated, maybe, but look at it anyway !
why write a program, when vlc can do all of this already ? (even headless/scripted!)
if there's audio to stream, too - forget opencv. it's a computer-vision lib, not build for your problem there
There are essentially two network protocols that are commonly used to send video from a server to a flash client, HTTP, and RTMP.
HTTP is a well-known standard, easily implemented because it is a plain-text protocol, that allows Flash Player to play on-demand video files, or do what is called pseudo-streaming.
RTMP is a proprietary protocol created by Adobe, that allows real-time streaming as well as video on demand, and can also transport structured binary data (the AMF format) to act as a remote procedure call protocol.
Although now documented, it is much more complicated to implement than HTTP, but there is an open-source library that implements this protocol, librtmp, found at http://rtmpdump.mplayerhq.hu/.
Please note that I have used librtmp with success, on the client side, to have a C program act as a Flash client to publish video on a FMS server. I have no experience of using it on the server side, I don't even know if it's possible at all.
In your case I certainly recommend using HTTP.
Now there is another problem to overcome, it is the fact that for video frames to be properly recognized, they must be embedded in a container that the Flash player can read.
Flash currently supports two container formats, FLV and F4V, the latter being a subset of the MPEG-4 container format.
Also, the video stream must be readable by Flash, and so it must be properly encoded into a format supported on the client-side, for example H.264, Sorensen, or VP6.
It is possible to directly send GIF, JPEG or PNG images as frames, as seen on page 8 of the official Flash Video Specification, but you must realize that in a HD resolution, this will be extremely inefficient, just imagine that at 25 FPS, a single image at 1920x1080 pixels in JPEG is much bigger than the equivalent H.264 frame.
So, in the end, my advice is: do not decode the video on the server, make sure it is in a format compatible with Flash, and use a well-documented protocol to send it as-is.
I successfully created an application where i can record microphone sound using flash and then save that stream to a server called "Red5" .
But lately i came across a strange requirement of capturing the output volume from machine and then saving that stream to red5 server.Like if i listen to a song or a skype call or listening to any other sound.I just want to capture those sounds.
I searched for this sort of situation just to get an headstart but i havent found any solution so that i can proceed with it.
Can anyone here provide a start up solution for this.
Can this be done through flash?Or any other way ?
Any help will be appreciated.Please provide suggestions
Thanks
Certainly there are ways to get this done, however using Flash as is I don't see it being possible (most likely a security concern, don't want ads arbitrarily recording peoples conversations).
Programs like Audacity are capable of recording off of the StereoMix output of a computer which is essentially what you're asking about. You could potentially build an AIR app that includes an ANE that packages the functionality from Audacity, but would require quite a bit of porting and... well time.
So, I'm working with a video source that I'm feeding into my Adobe AIR application via some native extension work, with the goal of ultimately getting it to a Flash Media Server. The video is H.264 encoded and muxed into a FLV container, which aligns me with supported Flash Media Server codecs and NetStream (appendBytes) requirements. I can get the data into AIR just fine.
The mine I stepped onto today, however, is that documentation for NetStream.appendBytes states I must call NetStream.play(null):
Call this method on a NetStream in "Data Generation Mode". To put a NetStream into Data Generation Mode, call NetStream.play(null) on a NetStream created on a NetConnection connected to null. Calling appendBytes() on a NetStream that isn't in Data Generation Mode is an error and raises an exception.
NetStream.play() called with a null parameter yields local FLV playback. I can't publish the stream to FMS in this mode. But my research into Flash seems to indicate NetStream's byte access is my only real hope here when dealing with non-camera or non-web video data.
Q: Can I latch onto the video playback buffer for publish to a FMS? Can I create a sort of pipeline of NetStreams or NetConnections to achieve this? Or is there an alternate approach here for transmitting H.264/FLV data to FMS? (The source of my video cannot communicate with FMS directly.)
The answer to your question is quite simply no. This is apparently implemented as a security feature, which is probably less of a security based issue and more of a sales issue. Adobe likes to block certain capabilities intentionally in order to create the possibility of, or need of another product aka more revenue.
I tried looking into this for you to see if there was some dirty hack where you could attach a camera or something and override the binary data being sent to the stream like you can with Audio but unfortunately, to my knowledge, no such hack is possible. More nfo here: NetStream.appendBytes
Update
You might be able to do something hackish by using ManyCam which is a virtual webcam driver (from what I understand). This will provide a valid camera you can select from flash and you can also select a video file as the source file for ManyCam. See http://manycam.com/user_guide/#HowtoSelectaVideofileasthePictureSourceforManyCam
Update #2
If you're looking for something open source that will do the same thing as manycam, check out the following:
http://code.google.com/p/webcamstudio/wiki/VideoSourceMovie (GPL Licensed)
I have an interesting project wherein I need to allow users to capture video of themselves with a webcam at a kiosk, after which I email them a link to their video. The trick is the resulting video needs to be a 'slow motion' version of the captured video. So for example, if someone creates a 2 minute movie, the resulting movie will be 4 minutes.
I'd like to build this in Flex / AS3 if possible. I don't have issues capturing the video and storing it / generating and emailing a link, but slowing down the video is the real mind bender. I'm unsure how to approach 'batch post-processing' a set of videos using Adobe tools.
Has anyone had a project similar to this or have suggestions on routes to take in order to do this?
Thanks!
-Josh
This is absolutely feasible from the client side, contrary to what some may believe. :)
http://code.google.com/p/flvrecorder/
Just adjust the capture rate, which shouldn't be too difficult all the source is there.
Alternatively, you could write an AIR app that launches Adobe Media Encoder after writing a file and launch it with a preset that has FTP info etc. Or you can just use the socket class to connect and upload over FTP.
http://code.google.com/p/fl-ftp/
It is not feasible to do this client-side.
Capture the video and send it to the server.
Use a library like FFMpeg to do your coneversions