Is it possible to play an MP4/H.264 movie in flash from half way thru?
Currently cant figure out how to seek to a time that hasn't buffered yet. Is there anyway of getting flash to start buffering half way into a movie?
edit:
The moov atom has already been set at the beginning of the MP4 file.
use pseudo-streaming
Normally you need a to use an streaming server which cost $$$ but you can use Pseudo-streaming.
Pseudo-streaming uses server-side scripts/modules where you have the server slice up the movie starting at the point you want to start from and wrap it as an mp4.
Stream your videos with standard HTTP servers
Pseudo-streaming (another article)
Oh yeah, I forgot about red5 as mentioned below in the comments. That is a free media server option for real streaming.
red5.org
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'm streaming a video from an Amazon CloudFront RTMP source with
video.attachNetStream(myNetStream);
myNetStream.play(myVideoFileName,0,-1);
and it's working quite well. Now, what I want to do is something like this:
video1.attachNetStream(myNetStream);
video2.attachNetStream(myNetStream);
myNetStream.play(myVideoFileName,0,-1);
This doesn't work as written because only one of the two videos will play at a time for some unknown reason. I want video1 and video2 to play the same video from a single NetStream (to save bandwidth) and remain completely in-sync with each other. How can I accomplish this?
If you are playing an FLV file directly (not streaming it from FMS), you should be able to :
Load the file with URLStream
Wait for sufficient data to start copying the data to a ByteArray object
Create as many NetStream objects as you need and use the appendBytes(bytes) method
I haven't actually tested it, and the logic of appendBytes() needs to be looked at, but theoretically it should work.
Also, it deserves a benchmark. But it's probably better than re-drawing a bitmap copy at the same rate as the video, and keep the two videos in sync.
bitmapData can't be accessed at all because of security restrictions. I'm going with plan B which is playing 2 netstreams, but reducing the file size of the one the videos by removing its audio. I'll have to wait until Amazon allow security policy access to use the bitmapData solution.
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
I'm recording a mono audio stream using a PIC at 8-bit 8Khz and streaming it raw to another microprocessor that houses a web server. I'm currently buffering the data and turning it into a wav file that gets played in the browser. What I'd like to be able to do is continuously stream the audio as it's being recorded without putting a lot of encoding overhead on the second processor. I've been searching but most searches turn up just streaming from a stored file, but since the file size isn't known ahead of time I'm not sure how to do this without the overhead of mp3 encoding.
You may find that simply creating a WAV file (or other raw format) that keeps growing will, in most players/browser plugins, cause the file to act as a live stream. This is, I believe, basically how like Ogg streaming and similar works. Because the player begins playing before it is done downloading anyway, it keeps playing and downloading until the end of the file, but the file has no end so it just keeps going.
Vlc media player can stream flv and many other formats.