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Updated RTMP and SRT
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Matthew Clark
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4 changed files with 135 additions and 21 deletions
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@ -4,6 +4,8 @@ GStreamer can send and receive audio and video via a network socket, using eithe
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*UDP* is faster but lossy - there is no attempt to resend lost network packets to it will fail if the network is not perfect. *TCP* acknowledges every network packet so is slower, but more reliable.
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See also [SRT](srt.md) as an alternative format.
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## UDP
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### Audio via UDP
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52
rtmp.md
52
rtmp.md
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@ -1,22 +1,23 @@
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# RTMP (GStreamer command-line cheat sheet)
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GStreamer can receive an RTMP stream from an RTMP server. It can also send an RTMP stream to an RTMP server.
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GStreamer can:
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* retrieve an RTMP stream from an RTMP server, and
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* also send an RTMP stream to an RTMP server (including YouTube).
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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/)
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### Play an RTMP stream
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To play from RTMP server, playbin can be used (as with files, HLS streams, DASH streams, etc):
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RTMP can be live streams, or on-demand streams - playback is the same in both cases.
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To play from RTMP server, [playbin](https://gstreamer.freedesktop.org/documentation/playback/playbin.html) can be used. (Playbin is magical - it can also play files, HLS streams, DASH streams, and many other sources!) Example:
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```
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export RTMP_SRC="rtmp://matthewc.co.uk/vod/scooter.flv"
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gst-launch-1.0 playbin uri=$RTMP_SRC
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```
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A test RTMP stream is available at `rtmp://184.72.239.149/vod/BigBuckBunny_115k.mov` which serves as a useful example:
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```
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gst-launch-1.0 playbin uri='rtmp://184.72.239.149/vod/BigBuckBunny_115k.mov'
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```
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A test RTMP VOD stream is available at `rtmp://matthewc.co.uk/vod/scooter.flv` which serves as a useful example:
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Instead of using `playbin`, it's possible to get video only with `uridecodebin` then shown with `autovideosink`:
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@ -112,7 +113,7 @@ gst-launch-1.0 \
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mix.
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```
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## Sending to an RTMP server
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## Sending a live stream to an RTMP server
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The examples below use the `RTMP_DEST` environment variable. You can set it to reference your RTMP server, e.g.
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@ -136,9 +137,9 @@ rtmp {
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}
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```
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then the application name is `livestream`, and so your URL will be `rtmp://<your-domain>/livestream/<stream-name>`.
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then the application name is `livestream`, and so your URL will be `rtmp://<your-domain>/livestream/<stream-name>` (where `<stream-name> can be anything`).
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### Sending a test stream to an RTMP server
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### Sending a test live stream to an RTMP server
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To send a video test source:
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@ -168,6 +169,37 @@ gst-launch-1.0 videotestsrc is-live=true ! \
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voaacenc bitrate=96000 ! audio/mpeg ! aacparse ! audio/mpeg, mpegversion=4 ! mux.
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```
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### Live streaming to YouTube via RTMP
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YouTube accepts live RTMP streams. They must have both audio and video.
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Set up a stream by visiting [YouTube.com](https://www.youtube.com/) on desktop, and selecting 'Create' from the top-right.
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YouTube will provide a 'Stream URL' and a 'Stream key'. Combine these to create the full URL.
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For example, if the URL is `rtmp://a.rtmp.youtube.com/live2` and the key is `abcd-1234-5678`, then:
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```
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export RTMP_DEST="rtmp://a.rtmp.youtube.com/live2/abcd-1234-5678"
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```
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Given the [YouTube stream suggestions](https://support.google.com/youtube/answer/2853702)) here's a good test stream:
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```
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gst-launch-1.0 \
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videotestsrc is-live=1 \
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! videoconvert \
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! "video/x-raw, width=1280, height=720, framerate=30/1" \
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! queue \
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! x264enc cabac=1 bframes=2 ref=1 \
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! "video/x-h264,profile=main" \
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! flvmux streamable=true name=mux \
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! rtmpsink location="${RTMP_DEST} live=1" \
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audiotestsrc is-live=1 wave=ticks \
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! voaacenc bitrate=128000 \
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! mux.
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```
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### Send a file over RTMP
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Audio & video:
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54
srt.md
54
srt.md
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@ -1,26 +1,57 @@
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# SRT (GStreamer command-line cheat sheet)
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SRT is a means of sending AV between to servers.
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SRT has a client-server relationship. One side must be the server, the other the client.
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The server can either be the side that is sending the audio/video (pushing) or the side that is
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receiving (pulling). The server must have started exist before the client.
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NOTE: THIS IS FOR GSTREAMER 1.14; IT HAS CHANGED IN 1.16.
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## Pre-requisites
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## Server sending the AV
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Create a sender server like this:
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If on MacOS, ensure the SRT library is installed:
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```
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gst-launch-1.0 videotestsrc ! video/x-raw, height=360, width=640 ! videoconvert ! x264enc tune=zerolatency ! video/x-h264, profile=high ! mpegtsmux ! srtserversink uri=srt://127.0.0.1:8888/
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brew install srt
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```
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And create a receiver client like this:
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Then at the time of writing, the default Homebrew build does not include SRT, so build from scratch:
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```
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gst-launch-1.0 -v srtclientsrc uri="srt://127.0.0.1:8888" ! decodebin ! autovideosink
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brew install --build-from-source gst-plugins-bad
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```
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## Server receiving the AV
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## Client receiving from server
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The most common usage is that the server has the video, and a client reads from it.
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To create a sender server, use `srtsink` (or alternatively, `srtserversink`):
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```
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gst-launch-1.0 -v videotestsrc ! video/x-raw, height=360, width=640 ! videoconvert ! x264enc tune=zerolatency ! video/x-h264, profile=high ! mpegtsmux ! srtsink uri=srt://:8888
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```
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And then recieve with `srtsrc` (or alternatively, `srtclientsrc`):
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```
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gst-launch-1.0 -v srtsrc uri="srt://127.0.0.1:8888" ! decodebin ! autovideosink
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```
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Don't forget, the amazing `playbin` can show anything - including SRT streams. This is useful for debugging:
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```
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gst-launch-1.0 playbin uri=srt://127.0.0.1:8888
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```
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By default, `srtsink` will wait for a clilent connection before allowing the stream to start. If you'd prefer this not to happen, set `wait-for-connection=false`:
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```
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gst-launch-1.0 -v videotestsrc ! video/x-raw, height=360, width=640 ! \
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videoconvert ! clockoverlay font-desc="Sans, 48" ! x264enc tune=zerolatency ! \
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video/x-h264, profile=high ! mpegtsmux ! srtsink uri=srt://:8888 wait-for-connection=false
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```
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## Client sending to server
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The alternative way round is to have the producer sending the AV to the receiving server.
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To have the server receiving, rather than sending, swap 'srtclientsrc' for 'srcserversrc'.
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Likewise, to have the client sending rather than receiving, swap 'srtserversink' for 'srtclientsink'.
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@ -28,11 +59,14 @@ Likewise, to have the client sending rather than receiving, swap 'srtserversink'
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Create a receiver server like this:
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```
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gst-launch-1.0 -v srtserversrc uri="srt://127.0.0.1:8888" ! decodebin ! autovideosink
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gst-launch-1.0 -v srtserversrc uri="srt://127.0.0.1:8889" ! decodebin ! autovideosink
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```
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And a sender client like this:
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```
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gst-launch-1.0 videotestsrc ! video/x-raw, height=360, width=640 ! videoconvert ! x264enc tune=zerolatency ! video/x-h264, profile=high ! mpegtsmux ! srtclientsink uri="srt://127.0.0.1:8888/"
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gst-launch-1.0 -v videotestsrc ! video/x-raw, height=360, width=640 ! \
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videoconvert ! clockoverlay font-desc="Sans, 48" ! x264enc tune=zerolatency ! \
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video/x-h264, profile=high ! mpegtsmux ! srtclientsink uri=srt://:8889
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```
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46
web_page_capture.md
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46
web_page_capture.md
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# Web page capture (WPE)
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The [wpesrc](https://gstreamer.freedesktop.org/documentation/wpe/wpesrc.html?gi-language=python) plugin can take a web page, and offer it as a GStreamer source. This allows you to:
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* Show web pages on screen, and
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* Use web pages as a means of doing graphics.
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The `wpesrc` plugin isn't frequently used. To see if you have it installed:
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```
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gst-inspect-1.0 | grep wpe
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```
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## Installing
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MacOS: The `wpesrc` plugin isn't part of the Homebrew build, unortunately.
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Ubuntu: `wpesrc` has a separate package called `gstreamer1.0-wpe`. So to install:
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```
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sudo apt-get install gstreamer1.0-wpe
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```
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## Using
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TODO
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### Streaming to SRT
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```
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LIBGL_ALWAYS_SOFTWARE=true gst-launch-1.0 -v wpevideosrc location="https://www.bbc.co.uk" ! videoconvert ! x264enc tune=zerolatency ! \
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video/x-h264, profile=high ! mpegtsmux ! srtsink uri=srt://:8889 wait-for-connection=false
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```
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### Without GPU
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Setting `LIBGL_ALWAYS_SOFTWARE=true` allows the `wpserc` element to work without GPU. This can lead to issues if the format is not set to `BGRA`. An example, sending a web page as a live stream to an RTMP server:
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```
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LIBGL_ALWAYS_SOFTWARE=true gst-launch-1.0 \
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wpevideosrc location="https://en.wikipedia.org/wiki/Main_Page" \
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! videoconvert ! videoscale ! videorate \
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! "video/x-raw, format=BGRA, width=854, height=480, framerate=30/1" \
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! videoconvert ! queue ! x264enc speed-preset=1 ! flvmux name=muxer \
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! rtmpsink location="$RTMP_DEST live=1"
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```
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