talk/libjingle_tests.gyp - Issue 1351803002: Exposing RtpSenders and RtpReceivers from PeerConnection.

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Unified Diff: talk/libjingle_tests.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_tests.gyp

diff --git a/talk/libjingle_tests.gyp b/talk/libjingle_tests.gyp

index 2719626a9ba1db3f8d861fa1bd003e6f08b7fde6..366267cc7e59605c9f335825ff4a8fbbfe6eea39 100755

--- a/talk/libjingle_tests.gyp

+++ b/talk/libjingle_tests.gyp

@@ -200,7 +200,6 @@

'app/webrtc/jsepsessiondescription_unittest.cc',

'app/webrtc/localaudiosource_unittest.cc',

'app/webrtc/mediastream_unittest.cc',

- 'app/webrtc/mediastreamhandler_unittest.cc',

'app/webrtc/mediastreamsignaling_unittest.cc',

'app/webrtc/peerconnection_unittest.cc',

'app/webrtc/peerconnectionendtoend_unittest.cc',

@@ -208,6 +207,7 @@

'app/webrtc/peerconnectioninterface_unittest.cc',

# 'app/webrtc/peerconnectionproxy_unittest.cc',

'app/webrtc/remotevideocapturer_unittest.cc',

+ 'app/webrtc/rtpsenderreceiver_unittest.cc',

'app/webrtc/sctputils.cc',

'app/webrtc/statscollector_unittest.cc',

'app/webrtc/test/fakeaudiocapturemodule.cc',

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