GstWebRTC - Audio Examples - x86

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GstWebRTC Basics


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This page presents some audio examples of GstWebRTC using OpenWebRTC's signaler.

Opus

Unidirectional

Example: In this example we use webrtcsink to send an audio stream and webrtcsrc to receive the audio stream.

The following pipeline will send periodic ticks:

gst-launch-1.0  webrtcsink start-call=true signaler::server_url=http://webrtc.ridgerun.com:8080 \
signaler::session_id=1234ridgerun name=web audiotestsrc is-live=true wave=8 ! audioconvert ! audioresample \
! queue ! opusenc ! rtpopuspay ! web.audio


The following pipeline will receive the periodic ticks:

gst-launch-1.0 webrtcsrc start-call=false signaler::server_url=http://webrtc.ridgerun.com:8080 \
signaler::session_id=1234ridgerun name=web web.audio ! rtpopusdepay ! opusdec ! audioconvert ! \
alsasink async=false

When executing the two previous pipelines, you should be able to listen the ticks in the receiving computer.



Bidirectional

Example: In this example we use two webrtcbin elements, each sends an audio stream and receives each other audio stream.

The following pipeline will send a white noise audio stream and receive the ticks audio stream sent by the next pipeline. This pipeline starts the call.

gst-launch-1.0 webrtcbin start-call=true signaler::server_url=http://webrtc.ridgerun.com:8080 \
signaler::session_id=1234ridgerun name=web audiotestsrc is-live=true wave=5 ! audioconvert ! \
audioresample ! queue ! opusenc ! rtpopuspay ! web.audio_sink web.audio_src ! rtpopusdepay ! \
opusdec ! audioconvert ! alsasink sync=false async=false

The following pipeline will send ticks audio stream and receive the white noise audio stream sent by the previous pipeline. This pipeline joins the call.

gst-launch-1.0 webrtcbin start-call=false signaler::server_url=http://webrtc.ridgerun.com:8080 \
signaler::session_id=1234ridgerun name=web audiotestsrc is-live=true wave=8 ! audioconvert ! \
audioresample ! queue ! opusenc ! rtpopuspay ! web.audio_sink web.audio_src ! rtpopusdepay ! \
opusdec ! audioconvert ! alsasink sync=false async=false

When executing the two previous pipelines, you should be able to listen the ticks and the white noise.




GstWebRTC Basics


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