I had set up a point-to-point stream using ffmpeg via UDP protocol and the stream worked, but there was screen tearing etc. I already tried raising the buffer size, but it did not help. This is a work network, so the UDP protocol won't work.
here is the full command:
ffmpeg -f dshow -i video="UScreenCapture" -r 30 -vcodec mpeg4 -q 12 -f mpegts udp://192.168.1.220:1234?pkt_size=188?buffer_size=65535
I've tried to make this work with TCP with no success
Here's what i've got now:
ffmpeg -f dshow -i video="UScreenCapture" -f mpegts tcp://192.168.1.194:5555
this returns an error:
real-time buffer [UScreenCapture] [Video input] too full or near too
full <323% of size: 3041280 [rtbufsize parameter]>! frame dropped!
This last message repeated xxxx times (it went up to around 1400 and I just turned it off).
I've tried to implement the -rtbufsize paremeter and raising the buffsize up to 800000000, didn't help.
I would appreciate any suggestions on how to solve this.
Related
I'm using ffmpeg to do some work on a network like this:
RtmpServer1 -- FfmpegServer -- RtmpServer2
I put a 6mins.mp4 on RtmpServer1(10.10.1.1) and play it on RtmpServer2(10.10.2.2) by this instruction:
ffmpeg -i rtmp://10.10.1.1:1935/play/6mins.mp4 -vcodec copy -c:v libx264 -f flv rtmp://10.10.2.2:1935/live
I would like to know is there a way to test the length of the time ffmpeg use to decode, compress and encode?(the duration from ffmpeg server get the data to the ffmpeg server send it out)
I tried tcpdump to listen on the both two eth ports(one for get and one for send) of ffmpeg server. But I can't match RTMP packets by pairs("pairs" means the packets containing the same data, one "got" packet matches one "sent" packet). I'd also like to know if there's a way to match the RTMP packets by their content(data).
I tried to use tcpdump on both of the input and output eths and get when the rtmp streamings starts and ends. It is helpful in some way.
So, I've read all the articles here and unfortunately I can't seem to find the answers I'm looking for. I've gotten close, but the certain magic strings allude me.
I'm running hls live streaming (nginx) on ubuntu 17.10 server. In short, I can get the server running one video at a time fine with ffmpeg (with subtitles) using the following:
ffmpeg -re -i "1.mkv" -vcodec libx264 -vprofile baseline -g 30 -b:v 1000k -s 852x480 -acodec aac -strict -2 -b:a 192k -ac 2 -vf subtitles=1.srt -f flv rtmp://localhost:1935/show/stream
Though, I cannot find a solution to run a playlist using this method. It seems impossible, and when I try vlc via sout (internally, or externally) I reveive either buffer problems, or the aac experimental codec error:
[aac # 0xb162e900] The encoder 'aac' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it.
Example string that spits that error:
vlc "1.mkv" --sout '#transcode{soverlay,vb=1000,vcodec=h264,width=853,height=480,acodec=mp4a,ab=128,channels=2,samplerate=44100}:std{access=rtmp,mux=ffmpeg{mux=flv},dst=rtmp://localhost:1935/show/stream}'
Every other audio codec doesn't work with flv. I'm at a loss, I've tried almost every combination I could think of and digout just to get to this point. The best functioning out of them has been ffmpeg: it doesn't buffer video at all, plays smoothly, but just can't play a playlist. Whereas vlc can play a playlist but buffers, and has no sound (internally). I've tried aenc=ffmpeg{strict=-2}, batch pipes, etc, etc. I need help. Nothing works. Is there any solution? All I want is to run a playlist of 25 videos, all different variations, on a loop to the m3u8 for embedding.
A friend of mine mentioned he used bash scripts to have a seamless playlist like viewing feature. Hopefully that points you in the direction you need. I can try digging them up if you want to work together on this, coz I too am interested in finding out more about it.
I'm trying to create a fairly simple streaming server/site. Here's the current flow:
OBS streams to an RTMP URL
Nginx accepts the RTMP stream and uses exec-push to have FFmpeg pick up the stream and transcode it
FFmpeg transcodes the stream and outputs it to a JSMpeg application, which displays the stream on a webpage.
When I have my exec_push statement as follows, everything seems to work perfectly, except the browser says Possible garbage data. Skipping. on every frame it receives:
exec_push /usr/bin/ffmpeg -re -i rtmp://127.0.0.1:1935/$app/$name -f mpeg1video http://localhost:8080/supersecret;
This behavior is understandable, because JSMpeg must receive MPEG-TS data, not MPEG1 data. It sees the MPEG1 frames and thinks they're garbage.
So through some online research, I found this:
exec_push /usr/bin/ffmpeg -re -i rtmp://127.0.0.1:1935/$app/$name -c:v copy -c:a copy -f mpegts http://localhost:8080/supersecret;
Supposedly, this is supposed to transcode my RTMP stream into an MPEG-TS format, which should be compatible with JSMpeg.
However, with the second version of the command, my FFmpeg -> JSMpeg stream keeps connecting and disconnecting, connecting and disconnecting, and so on. This behavior is observed in terminal:
Stream Connected: ::1:40208
close
Stream Connected: ::1:40212
close
Stream Connected: ::1:40216
close
Stream Connected: ::1:40220
close
Stream Connected: ::1:40224
close
...
What would cause this? I am pretty certain the issue is in my exec_push command. OBS is perfectly content, which tells me that the stream is making it to the server, and if I do a push, I can do a test push to Ustream just fine, which tells me that Nginx is at least processing the stream with some reasonable degree of success.
Disclaimer: I have no idea what I'm talking about. Everything I know about FFmpeg and JSMpeg/Node is from snippets of code that I found online.
Answer credit goes to #Mulvya.
In the second exec_push command, the -c:v copy -c:a copy should not be there. By using that, there isn't any transcoding going on-- it's just a stream passthrough.
Removing the -c:v copy -c:a copy from the command and restarting Nginx yields a successful stream.
I'm trying to deploy a live stream delivery system with nginx and nginx-rtmp-module.
For every node in my system, I wish it could 'forward' the live stream received to downstream node. I try to implement it by following config in my nginx.conf:
exec_push ffmpeg -i rtmp://localhost/src/test -vcodec copy -strict -2 -ar 44100 -ac 1 -f flv rtmp://<downstreaming A>/src/test -f flv rtmp://<downstreaming B>/src/test
it works when everything runs well, but if the downstream node is down, this command will exit and none of the downstream nodes could receive the live stream.
How could I force ffmpeg to ignore the connetion refused, or is there any better alternative to my implementation?
You cannot ignore connection refused since RTMP uses TCP which needs a connection.
If I understood correctly you're trying to transcode a RTMP source and send it to a number of servers.
You could duplicate your command to send to each downstream node
individually but you'll be doing the transcoding twice.
An alternative is to transcode and publish the transcoded stream using
ffserver on the same machine and then push to / pull on each downstream server
I'm currently doing a stream that is supposed to display correctly within Flowplayer.
First I send it to another PC via RTP. Here, I also checked with VLC that the codec etc. arrive correctly, which they do.
Now I want to expose this stream to Flowplayer as a file, so it can be displayed, via something I used in VLC:
http://localhost:8080/test.mp4
for example.
The full line I got is: ffmpeg -i input -f mp4 http://localhost:8080/test.mp4
However, no matter how I try to do this, I only get an input/output error. Is this only possible with something like ffserver or another?
What I think is this doesn't work because ffmpeg can't act as a server; on VLC it works since it can. (Though VLC ruins the codecs I set and it can't be read afterwards for some reason)
A (sort of) workaround I can use is saving the RTP stream to a file, and then letting flowplayer load it. This, however, only works once the file is not accessed anymore; I get a codec error otherwise.
To have FFmpeg act as an HTTP server, you need to pass the -listen 1 option. Additionally, -f mp4 will result in a non-fragmented MP4, which is not suitable for streaming. You can get a fragmented MP4 with -movflags frag_keyframe+empty_moov. A full working command line is:
ffmpeg -i input -listen 1 -f mp4 -movflags frag_keyframe+empty_moov http://localhost:8080
Other options you may find helpful are -re to limit the streaming speed to the input framerate, -stream_loop -1 to loop the input, and -c copy to avoid reencoding.
you need this command line
ffmpeg -f v4l2 -s 320x240 -r 25 -i /dev/video0 -f alsa -ac 1 -i hw:0 http://localhost:8090/feed1.ffm
make sure that your feed name ends with ".ffm" and if it's not the case, then add "-f ffm" before your feed URL, to manually specify the output format (because ffmpeg won't be able to figure it out automatically any more), like this "-f ffm http://localhost:8090/blah.bleh".