I am trying to run my Ubuntu machine as vlc server. where i run below command to stream my local video over http.vlc 1.avi
:sout=#transcode{vcodec=theo,vb=800,acodec=vorb,ab=128,channels=2,samplerate=44100}:duplicate{dst=http{dst=:8080/test.ogg}} :sout-all :sout-keep
Below is vlc client commad to display the http streaming output which is stopping always after 10 sec. For subsequent attempt this is not working.("failed to find url")
vlc http://localhost:8080/test.ogg .
Please suggest any workaround. Also please let me knwo if i should switched to ffmpeg if this is legacy problem. please suggest the command as well.
Note : using the latest vlc
Thanks in advance!
this was vlc version mismatch,once i make same vlc version in both client and server then it works perfectly
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Currently I am trying to play the Hls stream that is being sent to my nginx server via rtmp, in Unity with this plugin: AVPro. It is working on my PC but I want to play it on Android and it seems like the codec is not supported for Android regardless of which Video API I use. Currently im sending the stream with Streamlabs if that is important.
So I wondered if it is possible to somehow change the codec that nginx is using for the Hls stream.
When I watch the stream on my PC with VLC Media Player and look at the codec information it tells me this:
Videocodec: H264 - MPEG-4 AVC(part 10) (h264)
Audiocodec: ADTS
Im using nginx for Windows and this version: nginx 1.7.11.3 Gryphon
And if it is not possible with nginx I would like to know if there is an alternative that you can use as a streaming server. I haven't really found anything yet.
I tried using ffmpeg -i inputsource http://IP/live and even tried modifying win.conf file in my nginx folder by pushing rtmp://IP but when i use either http or rtmp URL in media playback device, I keep getting errors.
This Answer might help you in using ffmpeg as RTMP server.
I am using VLC to transcode the rtsp stream from an IP camera to a http mjpg stream via the following command:
cvlc -vvv -Idummy -q rtsp://user:password#hostname:554 --sout '#transcode{vcodec=MJPG,venc=ffmpeg{strict=1}}:standard{access=http{mime=multipart/x-mixed-replace;boundary=--7b3cc56e5f51db803f790dad720ed50a},mux=mpjpeg,dst=:8081/}'
This works fine.
I do not need to transcode the stream all the time but only a fraction of it.
VLC is transcoding even if no client is connected. That utilizes a whole CPU core on my server all the time.
Is there any possibility to start transcoding only if at least one client is connected and stop transcoding if the last client is disconnected?
Thank you very much!
I think you are asking if you can run the command line command above from your server - if so, then it does depend on the server and language, but in general yes, you can.
Your server logic would be something like:
When a client connects:
if this is the first client, run the command to start the transcoding
provide the link to the stream to the client
When a client disconnects:
if this is the last client, stop the transcoding
There will be a delay for the first client as the stream is buffered, but I am guessing you know that already.
The way to run the command will depend on the sever, but you can usually find examples - e.g. for Node: Execute a command line binary with Node.js
If you are using Java, there is a very well know and useful article on running cmd line from Java also - even if you are not using Java it is good reading: https://www.javaworld.com/article/2071275/core-java/when-runtime-exec---won-t.html
Using VLC 1.1.4 on Ubuntu 10.04 I can get the GUI to stream from my capture card to an HTTP client (e.g. IE8) but when I try using a command line version of properties the stream is not coming across. I am not sure I am coding the command line correctly. I grabbed the properties from the GUI and wrote a command line that seems reasonable, and gives no apparent errors in the VLC log, but no stream. When I point my browser at the camera.asf file, Windows Media Player pops up but almost immediately shows error "reached the end of file".
Here's what I pull from the GUI:
:sout=#transcode{vcodec=WMV2,vb=800,scale=1,acodec=wma2,ab=128,channels=2,samplerate=44100}:http{dst=:8080/camera.asf} :no-sout-rtp-sap :no-sout-standard-sap :sout-keep
And here's what I am trying as a command line:
cvlc v4l2:///dev/video0 --v4l2-input 1 --sout "#transcode{vcodec=WMV2,vb=800,scale=1,acodec=wma2,ab=128,channels=2,samplerate=44100}:http{dst=:8080/camera.asf}" --no-sout-rtp-sap --no-sout-standard-sap --sout-keep
I tried to get help from the VLC forum, but no replies.
Any ideas?
Thanks.
Can someone help me to buffer a mp3 file on a server using the Blackberry buffered playback demo app provided with the jde?
I have loaded it in the simulator and my mds is started but I'm unable to play the audio.
There is no error but it doesn't play/load.
The code looks all fine.
This may help:
Blackberry Enterprise Server Limitations
By default, the BlackBerry Enterprise Server (BES) limits the response size of a single HTTP response to 128K. If you try to fetch anything bigger your application will receive a 413 (Request Entity Too Large) response code. To get around this you must use the BES management console to change the value of the Maximum number of kilobytes per connection field to a higher value, up to 1024K.
Note that this limit also applies to the MDS simulator, so you'll need to change the simulator's settings as well. Edit the mds\config\rimpublic.property file in your JDE installation directory and change the value of the IPPP.connection.MaxNumberOfKBytesToSend property to match the BES setting and then restart the simulator.
Also check if you using correct ip instead of localhost
In the end you might want to check file in Blackberry browser before open it in app, don't forget to enable streaming in browser settings.