6.4 KiB
RTMP (GStreamer command-line cheat sheet)
GStreamer can receive an RTMP stream from an RTMP server. It can also send an RTMP stream to an RTMP server.
If you need your own RTMP server, the Nginx RTMP extension works quite well though is no longer supported.
Play an RTMP stream
To play from RTMP server, playbin can be used (as with files, HLS streams, DASH streams, etc):
gst-launch-1.0 playbin uri=$RTMP_SRC
A test RTMP stream is available at rtmp://184.72.239.149/vod/BigBuckBunny_115k.mov which serves as a useful example:
gst-launch-1.0 playbin uri='rtmp://184.72.239.149/vod/BigBuckBunny_115k.mov'
Instead of using playbin, it's possible to get video only with uridecodebin then shown with autovideosink:
gst-launch-1.0 uridecodebin uri=$RTMP_SRC ! autovideosink
Or as a step further we can split out into the source and decode too. This does video:
gst-launch-1.0 rtmpsrc location=$RTMP_SRC ! decodebin ! autovideosink
and this does the audio:
gst-launch-1.0 rtmpsrc name=rtmpsrc location=$RTMP_SRC ! decodebin ! \
queue ! audioconvert ! autoaudiosink
We can vget flvdemux to pull out the audio:
gst-launch-1.0 rtmpsrc location=$RTMP_SRC ! \
flvdemux name=t t.audio ! decodebin ! autoaudiosink
Incidentally, all of these work with a direct flv file:
gst-launch-1.0 filesrc location="/Users/clarkm22/workspace/silver/assets/test.flv" ! \
flvdemux name=t t.audio ! decodebin ! autoaudiosink
If you want to use flvdemux to do the video, you need to capture the audio too or else it will fail. This example puts it in fakesink which is basically discarding it:
gst-launch-1.0 rtmpsrc location="$RTMP_SRC" ! \
flvdemux name=demux \
demux.audio ! queue ! decodebin ! fakesink \
demux.video ! queue ! decodebin ! autovideosink
You could then use this to capture the RTMP as an MP4, e.g.
gst-launch-1.0 -e rtmpsrc location="$RTMP_SRC" ! \
flvdemux name=demux \
demux.audio ! queue ! decodebin ! audioconvert ! faac bitrate=32000 ! mux. \
demux.video ! queue ! decodebin ! videoconvert ! video/x-raw,format=I420 ! x264enc speed-preset=superfast tune=zerolatency psy-tune=grain sync-lookahead=5 bitrate=480 key-int-max=50 ref=2 ! mux. \
mp4mux name=mux ! filesink location="out.mp4"
According to this conversation, the following also works, although personally I find it intermittent:
export QUEUE="queue max-size-time=0 max-size-bytes=0 max-size-buffers=0"
gst-launch-1.0 \
rtmpsrc location="$RTMP_SRC live=1" ! \
$QUEUE ! \
flvdemux name=demux ! \
$QUEUE ! \
aacparse ! \
avdec_aac ! \
autoaudiosink sync=0 demux.video ! \
$QUEUE ! \
h264parse ! \
avdec_h264 ! \
$QUEUE ! \
videoconvert ! \
autovideosink sync=0
Adding an RTMP as picture-in-Picture
This overlays an RTMP source as a picture-in-picture on top of a local filesource (set as $SRC)
export QUEUE="queue max-size-time=0 max-size-bytes=0 max-size-buffers=0"
gst-launch-1.0 \
filesrc location="$SRC" ! \
decodebin ! videoconvert ! \
videoscale ! video/x-raw,width=640,height=360 ! \
compositor name=mix sink_0::alpha=1 sink_1::alpha=1 sink_1::xpos=50 sink_1::ypos=50 ! \
videoconvert ! autovideosink \
rtmpsrc location="$RTMP_SRC" ! \
flvdemux name=demux \
demux.audio ! $QUEUE ! decodebin ! autoaudiosink \
demux.video ! $QUEUE ! decodebin ! \
videoconvert ! \
videoscale ! video/x-raw,width=320,height=180! \
mix.
Sending to an RTMP server
Sending a test stream to an RTMP server
This will send a video test source:
gst-launch-1.0 videotestsrc is-live=true ! \
queue ! x264enc ! flvmux name=muxer ! rtmpsink location="$RTMP_DEST live=1"
This will send a audio test source (note: flvmux is still required even though there is no muxing of audio & video):
gst-launch-1.0 audiotestsrc is-live=true ! \
audioconvert ! audioresample ! audio/x-raw,rate=48000 ! \
voaacenc bitrate=96000 ! audio/mpeg ! aacparse ! audio/mpeg, mpegversion=4 ! \
flvmux name=mux ! \
rtmpsink location=$RTMP_DEST
This sends both video and audio as a test source:
gst-launch-1.0 videotestsrc is-live=true ! \
videoconvert ! x264enc bitrate=1000 tune=zerolatency ! video/x-h264 ! h264parse ! \
video/x-h264 ! queue ! flvmux name=mux ! \
rtmpsink location=$RTMP_DEST audiotestsrc is-live=true ! \
audioconvert ! audioresample ! audio/x-raw,rate=48000 ! \
voaacenc bitrate=96000 ! audio/mpeg ! aacparse ! audio/mpeg, mpegversion=4 ! mux.
Send a file over RTMP
Audio & video:
gst-launch-1.0 filesrc location=$SRC ! \
qtdemux name=demux \
demux.video_0 ! queue ! \
decodebin ! videoconvert ! x264enc bitrate=1000 tune=zerolatency ! video/x-h264 ! h264parse ! \
video/x-h264 ! queue ! flvmux name=mux ! \
rtmpsink location=$RTMP_DEST \
demux.audio_0 ! queue ! decodebin ! audioconvert ! audioresample ! \
audio/x-raw,rate=48000 ! \
voaacenc bitrate=96000 ! audio/mpeg ! aacparse ! audio/mpeg, mpegversion=4 ! mux.
Just video:
gst-launch-1.0 filesrc location=$SRC ! \
qtdemux name=demux \
demux.video_0 ! queue ! \
decodebin ! videoconvert ! x264enc bitrate=1000 tune=zerolatency ! video/x-h264 ! h264parse ! \
video/x-h264 ! queue ! flvmux name=mux ! \
rtmpsink location=$RTMP_DEST
TODO - Can we work out why a bad RTMP brings down the other mix?
export QUEUE="queue max-size-time=0 max-size-bytes=0 max-size-buffers=0"
gst-launch-1.0 \
filesrc location="$SRC2" ! \
decodebin ! videoconvert ! \
videoscale ! video/x-raw,width=640,height=360 ! \
compositor name=mix sink_0::alpha=1 sink_1::alpha=1 sink_1::xpos=50 sink_1::ypos=50 ! \
videoconvert ! autovideosink \
rtmpsrc location="$RTMP_DEST" ! \
flvdemux name=demux \
demux.audio ! $QUEUE ! decodebin ! fakesink \
demux.video ! $QUEUE ! decodebin ! \
videoconvert ! \
videoscale ! video/x-raw,width=320,height=180! \
mix.
Misc: latency
There's a comment about reducing latency at https://lists.freedesktop.org/archives/gstreamer-devel/2018-June/068076.html