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lines changed Original file line number Diff line number Diff line change @@ -262,13 +262,6 @@ bool GStreamerDataChannelHandler::checkState()
262262
263263 RTCDataChannelState state;
264264 switch (channelState) {
265- #if !GST_CHECK_VERSION(1, 22, 0)
266- // Removed in https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2099. In
267- // GStreamer < 1.22 GST_WEBRTC_DATA_CHANNEL_STATE_NEW had the 0 value. We keep this case only to
268- // avoid adding a default case.
269- case GST_WEBRTC_DATA_CHANNEL_STATE_NEW:
270- break ;
271- #endif
272265 case GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING:
273266 state = RTCDataChannelState::Connecting;
274267 break ;
Original file line number Diff line number Diff line change @@ -51,8 +51,9 @@ if (ENABLE_VIDEO OR ENABLE_WEB_AUDIO)
5151 message (FATAL_ERROR "GStreamerTranscoder >= 1.20 is needed for USE_GSTREAMER_TRANSCODER." )
5252 endif ()
5353
54- if (USE_GSTREAMER_WEBRTC AND (PC_GSTREAMER_VERSION VERSION_LESS "1.20" OR NOT PC_GSTREAMER_WEBRTC_FOUND))
55- message (FATAL_ERROR "GStreamerWebRTC >= 1.20 is needed for USE_GSTREAMER_WEBRTC." )
54+ # We use GStreamer 1.18 with backports for GstWebRTC support. In upstream WebKit, >= 1.20 is required.
55+ if (USE_GSTREAMER_WEBRTC AND (PC_GSTREAMER_VERSION VERSION_LESS "1.18" OR NOT PC_GSTREAMER_WEBRTC_FOUND))
56+ message (FATAL_ERROR "GStreamerWebRTC >= 1.18 is needed for USE_GSTREAMER_WEBRTC." )
5657 endif ()
5758 endif ()
5859endif ()
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