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| author | grunell@chromium.org <grunell@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2012-04-12 07:37:05 +0000 |
|---|---|---|
| committer | grunell@chromium.org <grunell@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2012-04-12 07:37:05 +0000 |
| commit | e6eb171a5edca28341f77dd40c63c1792ddbb5cb (patch) | |
| tree | 42ce9e5ad68c8975620d7aa82f092b3a5b0712b2 /content/content_renderer.gypi | |
| parent | 2d4729587b4377874e12be4c578333f02f0cba4b (diff) | |
| download | chromium_src-e6eb171a5edca28341f77dd40c63c1792ddbb5cb.zip chromium_src-e6eb171a5edca28341f77dd40c63c1792ddbb5cb.tar.gz chromium_src-e6eb171a5edca28341f77dd40c63c1792ddbb5cb.tar.bz2 | |
Adding JSEP PeerConnection glue.
This adds glue for JSEP PeerConnection. PeerConnectionHandler is split in two classes and a base class. The class name is kept for the old ROAP PeerConnection to be aligned with WebKit naming. ROAP is planned to be removed pretty soon, then the classes can be refactored.
See also main WebKit bug https://bugs.webkit.org/show_bug.cgi?id=80589
(In particular https://bugs.webkit.org/show_bug.cgi?id=82450)
TEST=content_unittests and manual webrtc test.
Review URL: http://codereview.chromium.org/9699069
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@131949 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'content/content_renderer.gypi')
| -rw-r--r-- | content/content_renderer.gypi | 4 |
1 files changed, 4 insertions, 0 deletions
diff --git a/content/content_renderer.gypi b/content/content_renderer.gypi index e3ca2ea..d64fe40 100644 --- a/content/content_renderer.gypi +++ b/content/content_renderer.gypi @@ -269,6 +269,10 @@ 'renderer/media/media_stream_impl.cc', 'renderer/media/peer_connection_handler.cc', 'renderer/media/peer_connection_handler.h', + 'renderer/media/peer_connection_handler_base.cc', + 'renderer/media/peer_connection_handler_base.h', + 'renderer/media/peer_connection_handler_jsep.cc', + 'renderer/media/peer_connection_handler_jsep.h', 'renderer/media/video_capture_module_impl.cc', 'renderer/media/video_capture_module_impl.h', 'renderer/media/webrtc_audio_device_impl.cc', |