I am streaming live video using rtp and ffmpeg using this command:
ffmpeg -re -f v4l2 -framerate 30 -video_size 640x480 -i /dev/video0 -c:v libx265 -tune zerolatency -s 320x240 -preset ultrafast -pix_fmt yuv420p -r 10 -strict experimental -f rtp rtp://127.0.0.1:49170 > ffmpeg.sdp
The generated sdp file is:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 56.36.100
m=video 49170 RTP/AVP 96
a=rtpmap:96 H265/9000
Vlc gives the following error:
The format of 'file:///home/username/ffmpeg.sdp' cannot be detected. Have a look at the log for details.
Terminal gives the following error:
[0xaf801010] ps demux error: cannot peek
[0xaf801010] mjpeg demux error: cannot peek
[0xaf801010] mpgv demux error: cannot peek
[0xaf801010] ps demux error: cannot peek
[0xb6d00618] main input error: no suitable demux module for `file/:///home/username/ffmpeg.sdp'
If I simply change libx265 -> libx264 in the command and H265 -> H264 the stream runs perfectly fine.
However I need to stream on H265. Any Suggestions?
I guess the problem is because VLC (or ffplay) doesn't get the VPS,SPS,PPS frames.
In order to start to decode H265 stream you need a VPS, a SPS, a PPS and an IDR frame.
In order to ask to libx265 to repeat these configuration frames before each IDR frame you could add to your streaming command :
-x265-params keyint=30:repeat-headers=1
Then the command becomes :
ffmpeg -re -f v4l2 -framerate 30 -video_size 640x480 -i /dev/video0 -c:v libx265 -tune zerolatency -x265-params keyint=30:repeat-headers=1 -s 320x240 -preset ultrafast -pix_fmt yuv420p -r 10 -strict experimental -f rtp rtp://127.0.0.1:49170 > ffmpeg.sdp
It generate the following ffmpeg.sdp file:
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 56.36.100
m=video 49170 RTP/AVP 96
a=rtpmap:96 H265/90000
I was able to display the stream with ffplay ffmpeg.sdp and VLC ffmpeg.sdp (removing the first line SDP: of ffmpeg.sdp)
Don't shoot me down in flames, as I do not use VLC for this type of thing but I recall that to get Gstreamer working with H265, I had to install:
libde265 and gstreamer1.0-libde265
There is also a vlc-plugin-libde265 listed in the ubuntu repositories.
See: https://github.com/strukturag/libde265
Related
I use ffmpeg to make an rtp stream to kurento media server and then send it to browsers via webrtc.
ffmpeg (h264 RTP) -> Kurento -> (h264 WebRTC) Browser
I'm capturing virtual xorg display.
This is my ffmpeg command:
ffmpeg -y -v info -fflags +genpts -f x11grab -draw_mouse 0 -r 25 -s 1280x720 -thread_queue_size 4096 -i :0.0+0,0 -an -c:v libx264 -preset veryfast -crf 25 -g 50 -pix_fmt yuv420p -maxrate 2976k -bufsize 5952k -ssrc 112233 -payload_type 103 -tune zerolatency -f rtp rtp://172.16.1.115:40258
This is my fake sdp offer used in negotiation with kurento RtpEndpoint
v=0
o=- 0 0 IN IP4 127.0.0.1
s=Stream
c=IN IP4 127.0.0.1
t=0 0
m=video 9 RTP/AVP 103
a=rtpmap:103 H264/90000
a=fmtp:103 packetization-mode=1
a=sendonly
a=direction:active
a=ssrc:112233 cname:user#example.com
Here is the problem:
Some I-frames are choppy on bottom half slice of the frame while others have no problem.
It is sometimes corrected when another I-frame has arrived, but mostly choppy.
When choppiness happened, Kms log says:
kmsutils kmsutils.c:483:gap_detection_probe:kmsagnosticbin2-108:sink Stream gap detected, timestamp: 0:51:22.574766908, duration: 0:00:00.000008237
Normal stream (No choppy at all)
Choppy stream
When it is corrected with an I-Frame (Sometimes it happens)
I have no clue about what could lead the issue.
Things that I tried to solve the problem.
Adding ?pkt_size=1000 (1100,900,1200. Since the mtu in kurento is 1200 default)
Changing -crf to different values in between 18-35
Changing -preset between medium and ultrafast
Changing framerate
Changing gop length (When i lower the gop length -More I-Frames- choppiness become shorter in duration but more frequent)
When I disable sliced-threads, there is no issue with bottom side but whole screen freezes in same scenario
Any help would be appreciated. Thanks.
I need some help getting ffplay to receive and decode a Real Time stream encoded in h264.
I'm trying to make a point-to-point stream between computer A receiving video frames from a Kinect and computer B running ffplay to show the livestream.
These are the commands I'm running on both computers.
Computer A (RPI 3)
ffmpeg -f rawvideo -vcodec rawvideo -pix_fmt rgb24 -s 640x480 -i - -threads 4 -preset ultrafast -codec:v libx2 64 -an -f rtp rtp://192.168.0.100:2000
This is what ffmpeg outputs:
fps= 14 q=12.0 size=856kB time=00:00:05.56 bitrate=1261.4kbits/s speed=0.54x
The out stream runs in between 10-20 frames. It's not good, but I can work with that.
Computer B
ffplay -protocol_whitelist "file,udp,rtp" -probesize 32 -sync ext -i streaming.sdp
streaming.sdp
v=0
0=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 192.168.0.100
t=0 0
a=tool:libavformat 57.56.100
m=video 2000 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1
I'm getting the stream, but at about 0.0001fps which is clearly bad. My guess is I'm missing something on the ffplay command, since ffmpeg shows a more stable and fast stream, but I can't seem to find what I'm missing.
The problem wasn't on ffmpeg, but on the code I wrote that was grabbing the data from the device. I was receiving the same frame multiple times, and blocking the thread capturing data, making most of the frames a duplicate of the first one.
I am streaming a webcam/audio with the command:
ffmpeg.exe -f dshow -framerate 30 -i video="xxx" -c:v libx264 -an -f rtp rtp://localhost:50041 -f dshow -i audio="xxx" -c:a aac -vn -f rtp rtp://localhost:50043
This outputs the following sdp info:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
t=0 0
a=tool:libavformat 57.65.100
m=video 50041 RTP/AVP 96
c=IN IP6 ::1
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1
m=audio 50043 RTP/AVP 97
c=IN IP6 ::1
b=AS:128
a=rtpmap:97 MPEG4-GENERIC/44100/2
a=fmtp:97 profile-level-id=1;mode=AAC-
hbr;sizelength=13;indexlength=3;indexdeltalength=3; config=121056E500
And I read the stream with the command:
ffmpeg.exe -protocol_whitelist file,udp,rtp -i D:\test.sdp -c:v libx264 -c:a aac d:\out.mp4
In the resulting file, the audio is slightly ahead of the video. I have read that RTCP runs on the RTP port + 1, and contains synchronization information. I don't see any RTCP information in the SDP file though.
Do I need to specify something to include RTCP?
If that's not the issue, what else can I do to sync the audio and video?
Not sure if RTCP is your issue, but I would start by trying to use one directshow input and splitting it to two outputs like this:
ffmpeg.exe -f dshow -framerate 30 -i video="XX":audio="YY" -an -vcodec libx264 -f rtp rtp://localhost:50041 -acodec aac -vn -f rtp rtp://localhost:50043
The ffmpeg DirectShow documentation mentions synchronization issues when multiple inputs are used. It also mentions trying the "-copy_ts" flag to resolve sync issues if you want to keep the inputs separate.
For stream my screen, i used :
ffmpeg -s 1920x1080 -f X11grab -i :0.0+0,0 -codec:v libvpx -b:v 4M -b:a libvorbis -crf 20 capture.webm
This command save the stream in a file : capture.webm.
But now, I want send stream on a udp server. So i make this command :
ffmpeg -s 1920x1080 -f X11grab -i :0.0+0,0 -codec:v libvpx -b:v 4M -b:a libvorbis -crf 20 -f webm udp://192.168.232.2:8080
But it doesn't run.
To send a stream to a server instead of
-f webm udp://192.168.232.2:8080
use
-f rtp rtp://192.168.232.2:32200
where 32200 is some not used port
To play it from there you can use:
1. ffplay with sdp
2. Set up WebRTC with Janus
3. Publish video in flv format to rtmp server and play it using Flash:
ffmpeg -protocol_whitelist file,udp,rtp -loglevel repeat+info -i source.sdp -flags +global_header -f flv rtmp://127.0.0.1/mystream/mystream1
where source.sdp(comes from output from ffmpeg -s 1920x1080 -f X11grab -i :0.0+0,0 -codec:v libvpx -b:v 4M -b:a libvorbis -crf 20 -f webm udp://192.168.232.2:32200)
v=0
o=- 0 0 IN IP4 127.0.1.1
s=No Name
c=IN IP4 192.168.232.2
t=0 0
a=tool:libavformat 57.71.100
m=video 32200 ....
If you don't want to build media server and viewing logic in the browser you can either send stream from your screen to any of Media Servers for example Wowza or Facebook Live(according to instructions)
What steps are needed to stream RTSP from FFmpeg?
Streaming UDP is not a problem, but as I want to stream to mobile devices which can natively read RTSP streams, I couldn't find any setup which tells what exactly is needed. Do I need an RTSP streaming server like LIVE555 or can I use FFmpeg only?
My Command:
ffmpeg -i space.mp4 -vcodec libx264 -tune zerolatency -crf 18 -f rtsp -muxdelay 0.1 rtsp://192.168.1.200:1234
I get an Input/Output error.
Do I need a SDP description to use RTSP?
And if yes where do I have to put it?
You can use FFserver to stream a video using RTSP.
Just change console syntax to something like this:
ffmpeg -i space.mp4 -vcodec libx264 -tune zerolatency -crf 18 http://localhost:1234/feed1.ffm
Create a ffserver.config file (sample) where you declare HTTPPort, RTSPPort and SDP stream. Your config file could look like this (some important stuff might be missing):
HTTPPort 1234
RTSPPort 1235
<Feed feed1.ffm>
File /tmp/feed1.ffm
FileMaxSize 2M
ACL allow 127.0.0.1
</Feed>
<Stream test1.sdp>
Feed feed1.ffm
Format rtp
Noaudio
VideoCodec libx264
AVOptionVideo flags +global_header
AVOptionVideo me_range 16
AVOptionVideo qdiff 4
AVOptionVideo qmin 10
AVOptionVideo qmax 51
ACL allow 192.168.0.0 192.168.255.255
</Stream>
With such setup you can watch the stream with i.e. VLC by typing:
rtsp://192.168.0.xxx:1235/test1.sdp
Here is the FFserver documentation.
FWIW, I was able to setup a local RTSP server for testing purposes using simple-rtsp-server and ffmpeg following these steps:
Create a configuration file for the RTSP server called rtsp-simple-server.yml with this single line:
protocols: [tcp]
Start the RTSP server as a Docker container:
$ docker run --rm -it -v $PWD/rtsp-simple-server.yml:/rtsp-simple-server.yml -p 8554:8554 aler9/rtsp-simple-server
Use ffmpeg to stream a video file (looping forever) to the server:
$ ffmpeg -re -stream_loop -1 -i test.mp4 -f rtsp -rtsp_transport tcp rtsp://localhost:8554/live.stream
Once you have that running you can use ffplay to view the stream:
$ ffplay -rtsp_transport tcp rtsp://localhost:8554/live.stream
Note that simple-rtsp-server can also handle UDP streams (i.s.o. TCP) but that's tricky running the server as a Docker container.
Another streaming command I've had good results with is piping the ffmpeg output to vlc to create a stream. If you don't have these installed, you can add them:
sudo apt install vlc ffmpeg
In the example I use an mpeg transport stream (ts) over http, instead of rtsp. I've tried both, but the http ts stream seems to work glitch-free on my playback devices.
I'm using a video capture HDMI>USB device that sets itself up on the video4linux2 driver as input. Piping through vlc must be CPU-friendly, because my old dual-core Pentium CPU is able to do the real-time encoding with no dropped frames. I've also had audio-sync issues with some of the other methods, where this method always has perfect audio-sync.
You will have to adjust the command for your device or file. If you're using a file as input, you won't need all that v4l2 and alsa stuff. Here's the ffmpeg|vlc command:
ffmpeg -thread_queue_size 1024 -f video4linux2 -input_format mjpeg -i /dev/video0 -r 30 -f alsa -ac 1 -thread_queue_size 1024 -i hw:1,0 -acodec aac -vcodec libx264 -preset ultrafast -crf 18 -s hd720 -vf format=yuv420p -profile:v main -threads 0 -f mpegts -|vlc -I dummy - --sout='#std{access=http,mux=ts,dst=:8554}'
For example, lets say your server PC IP is 192.168.0.10, then the stream can be played by this command:
ffplay http://192.168.0.10:8554
#or
vlc http://192.168.0.10:8554
UPDATE:
Here is a command to use VLC for rtsp, instead of using the rtsp-simple-server:
ffmpeg -thread_queue_size 1024 -f video4linux2 -input_format mjpeg -video_size 1280x720 -r 30 -i /dev/video0 -f alsa -thread_queue_size 1024 -i plughw:CARD=MS2109,DEV=0 -acodec mp2 -vcodec libx264 -preset ultrafast -crf 20 -s hd720 -vf format=yuv420p -profile:v main -f mpegts -|vlc -I dummy - --sout='#rtp{sdp=rtsp://:8554/} --sout-all --sout-keep'
If your PC ip is 192.168.0.10, then the rtsp stream is played by this command:
vlc rtsp://192.168.0.10:8554/
An alternative that I used instead of FFServer was Red5 Pro. On Ubuntu, I used this line:
ffmpeg -f pulse -i default -f video4linux2 -thread_queue_size 64 -framerate 25 -video_size 640x480 -i /dev/video0 -pix_fmt yuv420p -bsf:v h264_mp4toannexb -profile:v baseline -level:v 3.2 -c:v libx264 -x264-params keyint=120:scenecut=0 -c:a aac -b:a 128k -ar 44100 -f rtsp -muxdelay 0.1 rtsp://localhost:8554/live/paul