Difference between revisions of "GstWebRTC - AppRTC Audio Examples - x86"
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==== Example ==== | ==== Example ==== | ||
| − | In this example we use | + | In this example, we use rrwebrtcbin to send an audio stream and rrwebrtcbin to receive the audio stream. |
==== Send Pipeline ==== | ==== Send Pipeline ==== | ||
<syntaxhighlight lang=bash> | <syntaxhighlight lang=bash> | ||
| − | gst-launch-1.0 | + | gst-launch-1.0 rrwebrtcbin start-call=true signaler=GstApprtcSignaler signaler::server_url=http://localhost:8080 \ |
| − | signaler::session_id=1234ridgerun name=web audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web. | + | signaler::session_id=1234ridgerun name=web audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink |
</syntaxhighlight> | </syntaxhighlight> | ||
| Line 47: | Line 47: | ||
<syntaxhighlight lang=bash> | <syntaxhighlight lang=bash> | ||
| − | gst-launch-1.0 | + | gst-launch-1.0 rrwebrtcbin start-call=false signaler=GstApprtcSignaler signaler::server_url=http://localhost:8080 \ |
| − | signaler::session_id=1234ridgerun name=web web. | + | signaler::session_id=1234ridgerun name=web web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink |
</syntaxhighlight> | </syntaxhighlight> | ||
Revision as of 12:04, 14 March 2019
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This page presents some GstRrWebRTC audio examples on x86 platform using AppRTC.
Server Setup
To run the examples first enable the Websocket server:
$GOPATH/bin/collidermain -port=8089 -tls=false
Then, enable the AppRTC Node server in a different terminal window:
cd <PATH>/apprtc-node-server
node ./bin/www
Note: Make sure you previously install dependencies needed for enable the servers, if you didn't follow this link [AppRTC Node Server with our websocket server]
Opus
Unidirectional elements
Example
In this example, we use rrwebrtcbin to send an audio stream and rrwebrtcbin to receive the audio stream.
Send Pipeline
gst-launch-1.0 rrwebrtcbin start-call=true signaler=GstApprtcSignaler signaler::server_url=http://localhost:8080 \
signaler::session_id=1234ridgerun name=web audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink
Receive Pipeline
gst-launch-1.0 rrwebrtcbin start-call=false signaler=GstApprtcSignaler signaler::server_url=http://localhost:8080 \
signaler::session_id=1234ridgerun name=web web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink
Bidirectional elements
Example
In this example we use two rrwebrtcbins, each send an audio stream and receives each other audio stream.
Send-Receive Pipeline
gst-launch-1.0 rrwebrtcbin start-call=true signaler=GstApprtcSignaler signaler::server_url=http://localhost:8080 \
signaler::session_id=1234ridgerun name=web web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink \
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink
Send-Receive Pipeline
gst-launch-1.0 rrwebrtcbin start-call=false signaler=GstApprtcSignaler signaler::server_url=http://localhost:8080 \
signaler::session_id=1234ridgerun name=web web.audio_src ! rtpopusdepay ! opusdec ! autoaudiosink \
audiotestsrc is-live=true ! opusenc ! rtpopuspay ! web.audio_sink
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