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Unified Diff: talk/libjingle.gyp

Issue 1351803002: Exposing RtpSenders and RtpReceivers from PeerConnection. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Adding some stubs so that Chromium build won't break. Created 5 years, 3 months ago
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Index: talk/libjingle.gyp
diff --git a/talk/libjingle.gyp b/talk/libjingle.gyp
index 804170a3fc16a1efc0acd58cebaebace667313c0..9f41b2705518f4fb5d3466402ef2899bc5031bf3 100755
--- a/talk/libjingle.gyp
+++ b/talk/libjingle.gyp
@@ -733,8 +733,6 @@
'app/webrtc/mediacontroller.h',
'app/webrtc/mediastream.cc',
'app/webrtc/mediastream.h',
- 'app/webrtc/mediastreamhandler.cc',
- 'app/webrtc/mediastreamhandler.h',
'app/webrtc/mediastreaminterface.h',
'app/webrtc/mediastreamprovider.h',
'app/webrtc/mediastreamproxy.h',
@@ -757,6 +755,12 @@
'app/webrtc/remoteaudiosource.h',
'app/webrtc/remotevideocapturer.cc',
'app/webrtc/remotevideocapturer.h',
+ 'app/webrtc/rtpreceiver.cc',
+ 'app/webrtc/rtpreceiver.h',
+ 'app/webrtc/rtpreceiverinterface.h',
+ 'app/webrtc/rtpsender.cc',
+ 'app/webrtc/rtpsender.h',
+ 'app/webrtc/rtpsenderinterface.h',
'app/webrtc/sctputils.cc',
'app/webrtc/sctputils.h',
'app/webrtc/statscollector.cc',
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