2 ; generating dot file representing pipeline state
6 ; | separated list of arguments that will pass to gst_init
7 gstreamer arguments = --gst-debug=webrtcbin:7,3
9 ; comma separated list of elements that will not use in the gstreamer pipeline
10 gstreamer excluded elements =
12 ; latency of RTP jitterbuffer
13 rtp jitterbuffer latency = 100
15 ; FEC setting of RTP packets
18 ; default STUN server URL
19 stun server = stun://stun.l.google.com:19302
23 ; default values for video source pipeline (e.g, videotest, camera, screen)
24 video raw format = I420
29 video hw encoder element =
30 video encoded format support = no
31 video drc support = no
32 ; default values for audio source pipeline (e.g, audiotest, mic)
33 audio raw format = S16LE
34 audio samplerate = 8000
37 audio hw encoder element =
41 source element = videotestsrc
42 ; values below will override the default one of [media source] above
48 ;video hw encoder element =
49 ;video encoded format support =
50 video drc support = yes
54 source element = v4l2src
55 ; values below will override the default one of [media source] above
61 ;video hw encoder element =
62 ;video encoded format support =
67 source element = waylandsrc
68 ; values below will override the default one of [media source] above
74 ;video hw encoder element =
75 ;video encoded format support =
80 source element = audiotestsrc
81 ; values below will override the default one of [media source] above
86 ;audio hw encoder element =
90 source element = pulsesrc
91 ; values below will override the default one of [media source] above
96 ;audio hw encoder element =
100 audio sink element = pulsesink
101 video sink element = tizenwlsink
103 ; comma separated list of elements, it should be one by one per codec type
104 audio hw decoder elements =
105 video hw decoder elements =
108 [resource acquisition]