2 ; generating dot file representing pipeline state
6 ; enable verbose logging
9 ; enable nice verbose logging
12 ; | separated list of arguments that will pass to gst_init
13 gstreamer arguments = --gst-debug=webrtcbin:7,3
15 ; comma separated list of elements that will not use in the gstreamer pipeline
16 gstreamer excluded elements =
18 ; latency of RTP jitterbuffer
19 rtp jitterbuffer latency = 100
21 ; default STUN server URL
22 stun server = stun://stun.l.google.com:19302
26 ; default values for video source pipeline (e.g, videotest, camera, screen)
27 video raw format = I420
32 video hw encoder element =
33 video encoded format support = no
34 video drc support = no
35 ; default values for audio source pipeline (e.g, audiotest, mic)
36 audio raw format = S16LE
37 audio samplerate = 8000
40 audio hw encoder element =
41 ; default FEC setting of RTP packets
46 source element = videotestsrc
47 ; values below will override the default one of [media source] above
53 ;video hw encoder element =
54 ;video encoded format support =
55 video drc support = yes
59 source element = v4l2src
60 ; values below will override the default one of [media source] above
66 ;video hw encoder element =
67 ;video encoded format support =
72 source element = waylandsrc
73 ; values below will override the default one of [media source] above
79 ;video hw encoder element =
80 ;video encoded format support =
85 source element = audiotestsrc
86 ; values below will override the default one of [media source] above
91 ;audio hw encoder element =
95 source element = pulsesrc
96 ; values below will override the default one of [media source] above
101 ;audio hw encoder element =
105 audio sink element = pulsesink
106 video sink element = tizenwlsink
108 ; comma separated list of elements, it should be one by one per codec type
109 audio hw decoder elements =
110 video hw decoder elements =
113 [resource acquisition]