While showing the rstp-stream, we are trying to transcode, and record it to file for later use.
VLC 2.1.1 seems to work when using
vlc rtsp-stream :sout=#transcode{vcodec=mp1v,vb=700,scale=1}:duplicate{dst=display,dst=std{access=file,mux=asf,dst="C:\file.mpg"}}"
This seems to be broken on VLC 2.2.4
Documentation on VLC's wiki seems outdated.
Discrepancies with the vcodec/mux pairing as well as using samplerate instead of vb seem to fix it.
vlc rtsp-stream :sout=#transcode{vcodec=mp4v,scale=0.5,samplerate=44100}:duplicate{dst=display,dst=std{access=file,mux=mp4,dst='C:\file.mpg'}}"
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I need to replay guacamole replay file by html5.
But the only thing I can do now is to use guacenc to convert the playback file into a video and play it locally.
I tried to build a demo based on the secondary development of guacamole-example to realize the playback of the screen file.
Although I still don't know the principle of its implementation, it solves my problem very well.
I also tried the official demo, but there is a problem with maven dependencies.
Is there any gem out there that can interact with users webcam to capture video and audio and upload it to the server?
Have you heard about navigator.getUserMedia()? That could the trick. More info: http://www.html5rocks.com/en/tutorials/getusermedia/intro/
They talk about it in the Mobile Web Development course in Udacity (lesson 10).
https://www.youtube.com/watch?v=j6mzYt5fJpg
Have you tried headshot ?
I just tried it yesterday, but ran into some trouble with the flash player, not beeing able to set permissions for camera use
try it out, its easy to add to your application, maybe you'll make it work.
and better backup your files before.
I'm developing some Video Editing Apps on Android.
the objective of the app is "Editing Videos on Android".
and...
I'm just completed making video file using some images.
but.. I can't attach audio into the video.
my method is same as follows.
1.VideoStream, audio stream creation using AVFormatContext
2.Movie encoding in video stream was successful
3.Encode codec open in audio stream was successful
4.Set sample format to AV_SAMPLE_FMT_FLTP
5.Sample rate and channel was set same as source audio
6.Choose appropriate Decoder and read packet
7.Convert packets using swr_converter, setting same as sample format
8.Encode converted data
9.memory deallocation
10.END!
Problem is here:
Video of finally created video file was normally played. but the Audio wasn't.
It heared like weird. It have many noises and plays slowly.
I've googled with many keywords but they only say about "FFmpeg command line usage".
I wanna make with FFMpeg API. not a Command line tool.
Please help.
Your question is vague without some kind of code to go along with it, as trust me there are a lot of things that can go wrong when using ffmpeg's libraries directly (and on Windows there is no debuging). Unfortunately ffmpeg's libraries are not well documented so it is generally best to read the source code for ffmpeg in order to use its libraries. Find the equivalent command line options to perform what you want and track that through ffmpeg's source to see the library calls.
I've recorded streams using streamPublishStart callback with Red5 streaming server. It works. But a few times, the internet connection fall down in the publisher side. Then, in the streams directory, I have got a .flv.ser file. It's not playable. I've tried to repair/fix it with all software that propose it. No success. I've use flvcheck.exe and the report is : Error -18 truncated box. I've seen discussion on Adobe forums but no interesting things. Could you propose me a technic or a software to solve my problem.
thanks in advance,
Pascal.
Did you get your question answered? The .flv.ser is a temp file, created until Red5 is done processing the stream. When done, there is a new file without the .ser extension. What I had to do was create a ajax script that looks at the directory for a .flv.ser file and prevents closing the page until the conversion is completed. Red5 version one is slow at doing the conversion. I'm testing 1.0.2 RC1 right now, but initial results look like it is even worse. I hear version .8 is the best for recording so I may have to downgrade to that.
I'm late to the party but you simply have to concatenate the files. On Linux this works like this:
cat foo.flv foo.flv.ser > playable_foo.flv
I read that somewhere else but I forgot where it was.
Using carrierwave or another file upload plugin, there should be an easy way to convert uploaded mp4 (H.264) video files to Apple HLS. You need this for streaming videos on mobile devices.
This way, while uploading 1 streaming video file you can support most of the browsers (every browser except Opera and FF for Linux/OSX).
It can be done with ffmpeg, is anyone working on a gem?
How about streamio-ffmpeg? It seems to support custom command line parameters. You could add then use parameters like from this answer: https://stackoverflow.com/a/10047372/759140
You might also give carrierwave-video a try. It's backed by streamio-ffmpeg and allows passing custom settings through.