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| author | deadbeef <deadbeef@webrtc.org> | 2015-10-13 13:23:41 -0700 |
|---|---|---|
| committer | Commit bot <commit-bot@chromium.org> | 2015-10-13 20:23:48 +0000 |
| commit | 97c392935411398b506861601c82e31d95c591f0 (patch) | |
| tree | f1a691c85e691612f58063de02da4c3fc9908c13 /talk/libjingle_tests.gyp | |
| parent | a0751c5c068ee76aaeeac56173ca043da1d568ff (diff) | |
| download | webrtc-97c392935411398b506861601c82e31d95c591f0.tar.gz | |
Moving MediaStreamSignaling logic into PeerConnection.
This needs to happen because in the future, m-lines will be offered
based on the set of RtpSenders/RtpReceivers, rather than the set of
tracks that MediaStreamSignaling knows about.
Besides that, MediaStreamSignaling was a "glue class" without
a clearly defined role, so it going away is good for other
reasons as well.
Review URL: https://codereview.webrtc.org/1393563002
Cr-Commit-Position: refs/heads/master@{#10268}
Diffstat (limited to 'talk/libjingle_tests.gyp')
| -rwxr-xr-x | talk/libjingle_tests.gyp | 1 |
1 files changed, 0 insertions, 1 deletions
diff --git a/talk/libjingle_tests.gyp b/talk/libjingle_tests.gyp index 366267cc7e..2e4204784a 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/mediastreamsignaling_unittest.cc', 'app/webrtc/peerconnection_unittest.cc', 'app/webrtc/peerconnectionendtoend_unittest.cc', 'app/webrtc/peerconnectionfactory_unittest.cc', |