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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.
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
@ -47,7 +47,7 @@ gst-launch-1.0 rtmpsrc location=$RTMP_SRC ! \
Incidentally, all of these work with a direct *flv* file:
```
gst-launch-1.0 filesrc location="/Users/clarkm22/workspace/silver/assets/test.flv" ! \
gst-launch-1.0 filesrc location="/path/to/test.flv" ! \
flvdemux name=t t.audio ! decodebin ! autoaudiosink
```
@ -114,16 +114,40 @@ gst-launch-1.0 \
## Sending to an RTMP server
The examples below use the `RTMP_DEST` environment variable. You can set it to reference your RTMP server, e.g.
```
export RTMP_DEST="rtmp://example.com/live/test"
```
If you're using [Nginx RTMP](https://github.com/arut/nginx-rtmp-module), the name you give your application needs to be the first part of the URL path. For example, if your NGINX configuration is:
```
rtmp {
server {
listen 1935;
hunk_size 4096;
notify_method get;
application livestream {
live on;
}
}
}
```
then your URL will be `rtmp://your-domain.com/livestream/whatever-you-want`.
### Sending a test stream to an RTMP server
This will send a video test source:
To send a video test source:
```
gst-launch-1.0 videotestsrc is-live=true ! \
queue ! x264enc ! flvmux name=muxer ! rtmpsink location="$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):
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 ! \
@ -171,27 +195,6 @@ gst-launch-1.0 filesrc location=$SRC ! \
rtmpsink location=$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