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Update RTMP to clarify use of Nginx

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Matthew Clark 2022-08-16 22:06:54 +01:00
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rtmp.md
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@ -2,7 +2,7 @@
GStreamer can receive an RTMP stream from an RTMP server. It can also send an RTMP stream to an RTMP server. 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](https://github.com/arut/nginx-rtmp-module) works quite well though is no longer supported. If you need your own RTMP server, [the Nginx RTMP extension](https://github.com/arut/nginx-rtmp-module) works quite well. [Linode has a good NGINX RTMP installation guide.](https://www.linode.com/docs/guides/set-up-a-streaming-rtmp-server/)
### Play an RTMP stream ### Play an RTMP stream
@ -47,7 +47,7 @@ gst-launch-1.0 rtmpsrc location=$RTMP_SRC ! \
Incidentally, all of these work with a direct *flv* file: Incidentally, all of these work with a direct *flv* file:
``` ```
gst-launch-1.0 filesrc location="/Users/clarkm22/workspace/silver/assets/test.flv" ! \ gst-launch-1.0 filesrc location="/path/to/test.flv" ! \
flvdemux name=t t.audio ! decodebin ! autoaudiosink flvdemux name=t t.audio ! decodebin ! autoaudiosink
``` ```
@ -114,16 +114,40 @@ gst-launch-1.0 \
## Sending to an RTMP server ## Sending to an RTMP server
The examples below use the `RTMP_DEST` environment variable. You can set it to reference your RTMP server, e.g.
```
export RTMP_DEST="rtmp://example.com/live/test"
```
If you're using [Nginx RTMP](https://github.com/arut/nginx-rtmp-module), the name you give your application needs to be the first part of the URL path. For example, if your NGINX configuration is:
```
rtmp {
server {
listen 1935;
hunk_size 4096;
notify_method get;
application livestream {
live on;
}
}
}
```
then your URL will be `rtmp://your-domain.com/livestream/whatever-you-want`.
### Sending a test stream to an RTMP server ### Sending a test stream to an RTMP server
This will send a video test source: To send a video test source:
``` ```
gst-launch-1.0 videotestsrc is-live=true ! \ gst-launch-1.0 videotestsrc is-live=true ! \
queue ! x264enc ! flvmux name=muxer ! rtmpsink location="$RTMP_DEST live=1" queue ! x264enc ! flvmux name=muxer ! rtmpsink location="$RTMP_DEST live=1"
``` ```
This will send a audio test source (note: `flvmux` is still required even though there is no muxing of audio & video): To send an audio test source (note: `flvmux` is still required even though there is no muxing of audio & video):
``` ```
gst-launch-1.0 audiotestsrc is-live=true ! \ gst-launch-1.0 audiotestsrc is-live=true ! \
@ -171,27 +195,6 @@ gst-launch-1.0 filesrc location=$SRC ! \
rtmpsink location=$RTMP_DEST rtmpsink location=$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 ## Misc: latency
There's a comment about reducing latency at https://lists.freedesktop.org/archives/gstreamer-devel/2018-June/068076.html There's a comment about reducing latency at https://lists.freedesktop.org/archives/gstreamer-devel/2018-June/068076.html