Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(388)

Unified Diff: third_party/libjingle/BUILD.gn

Issue 1615433002: Roll WebRTC 11523:11548, Libjingle 11522:11545 (Closed) Base URL: http://chromium.googlesource.com/chromium/src.git@master
Patch Set: Rolling to webrtc@11548 instead to pull in a fix Created 4 years, 10 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « remoting/protocol/webrtc_video_stream.cc ('k') | third_party/libjingle/README.chromium » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: third_party/libjingle/BUILD.gn
diff --git a/third_party/libjingle/BUILD.gn b/third_party/libjingle/BUILD.gn
index bc2c3e7800def608ff237ff3bc8f2b8135e5a5c2..be97212d2fe31e0523e7306513886b45e1c35755 100644
--- a/third_party/libjingle/BUILD.gn
+++ b/third_party/libjingle/BUILD.gn
@@ -296,6 +296,78 @@ if (enable_webrtc) {
# as is supported in the GYP build. It's not clear what this is used for.
source_set("libjingle_webrtc_common") {
sources = [
+ "../webrtc/api/audiotrack.cc",
+ "../webrtc/api/audiotrack.h",
+ "../webrtc/api/datachannel.cc",
+ "../webrtc/api/datachannel.h",
+ "../webrtc/api/dtlsidentitystore.cc",
+ "../webrtc/api/dtlsidentitystore.h",
+ "../webrtc/api/dtmfsender.cc",
+ "../webrtc/api/dtmfsender.h",
+ "../webrtc/api/jsep.h",
+ "../webrtc/api/jsepicecandidate.cc",
+ "../webrtc/api/jsepicecandidate.h",
+ "../webrtc/api/jsepsessiondescription.cc",
+ "../webrtc/api/jsepsessiondescription.h",
+ "../webrtc/api/localaudiosource.cc",
+ "../webrtc/api/localaudiosource.h",
+ "../webrtc/api/mediaconstraintsinterface.cc",
+ "../webrtc/api/mediaconstraintsinterface.h",
+ "../webrtc/api/mediacontroller.cc",
+ "../webrtc/api/mediacontroller.h",
+ "../webrtc/api/mediastream.cc",
+ "../webrtc/api/mediastream.h",
+ "../webrtc/api/mediastreamhandler.cc",
+ "../webrtc/api/mediastreamhandler.h",
+ "../webrtc/api/mediastreaminterface.h",
+ "../webrtc/api/mediastreamobserver.cc",
+ "../webrtc/api/mediastreamobserver.h",
+ "../webrtc/api/mediastreamprovider.h",
+ "../webrtc/api/mediastreamproxy.h",
+ "../webrtc/api/mediastreamtrack.h",
+ "../webrtc/api/mediastreamtrackproxy.h",
+ "../webrtc/api/notifier.h",
+ "../webrtc/api/peerconnection.cc",
+ "../webrtc/api/peerconnection.h",
+ "../webrtc/api/peerconnectionfactory.cc",
+ "../webrtc/api/peerconnectionfactory.h",
+ "../webrtc/api/peerconnectioninterface.h",
+ "../webrtc/api/portallocatorfactory.cc",
+ "../webrtc/api/portallocatorfactory.h",
+ "../webrtc/api/remoteaudiosource.cc",
+ "../webrtc/api/remoteaudiosource.h",
+ "../webrtc/api/remoteaudiotrack.cc",
+ "../webrtc/api/remoteaudiotrack.h",
+ "../webrtc/api/remotevideocapturer.cc",
+ "../webrtc/api/remotevideocapturer.h",
+ "../webrtc/api/rtpreceiver.cc",
+ "../webrtc/api/rtpreceiver.h",
+ "../webrtc/api/rtpreceiverinterface.h",
+ "../webrtc/api/rtpsender.cc",
+ "../webrtc/api/rtpsender.h",
+ "../webrtc/api/rtpsenderinterface.h",
+ "../webrtc/api/sctputils.cc",
+ "../webrtc/api/sctputils.h",
+ "../webrtc/api/statscollector.cc",
+ "../webrtc/api/statscollector.h",
+ "../webrtc/api/statstypes.cc",
+ "../webrtc/api/statstypes.h",
+ "../webrtc/api/streamcollection.h",
+ "../webrtc/api/umametrics.h",
+ "../webrtc/api/videosource.cc",
+ "../webrtc/api/videosource.h",
+ "../webrtc/api/videosourceinterface.h",
+ "../webrtc/api/videosourceproxy.h",
+ "../webrtc/api/videotrack.cc",
+ "../webrtc/api/videotrack.h",
+ "../webrtc/api/videotrackrenderers.cc",
+ "../webrtc/api/videotrackrenderers.h",
+ "../webrtc/api/webrtcsdp.cc",
+ "../webrtc/api/webrtcsdp.h",
+ "../webrtc/api/webrtcsession.cc",
+ "../webrtc/api/webrtcsession.h",
+ "../webrtc/api/webrtcsessiondescriptionfactory.cc",
+ "../webrtc/api/webrtcsessiondescriptionfactory.h",
"../webrtc/media/base/audiorenderer.h",
"../webrtc/media/base/capturemanager.cc",
"../webrtc/media/base/capturemanager.h",
@@ -338,78 +410,6 @@ if (enable_webrtc) {
"../webrtc/media/webrtc/webrtcvideoframefactory.cc",
"../webrtc/media/webrtc/webrtcvideoframefactory.h",
"../webrtc/media/webrtc/webrtcvoe.h",
- "source/talk/app/webrtc/audiotrack.cc",
- "source/talk/app/webrtc/audiotrack.h",
- "source/talk/app/webrtc/datachannel.cc",
- "source/talk/app/webrtc/datachannel.h",
- "source/talk/app/webrtc/dtlsidentitystore.cc",
- "source/talk/app/webrtc/dtlsidentitystore.h",
- "source/talk/app/webrtc/dtmfsender.cc",
- "source/talk/app/webrtc/dtmfsender.h",
- "source/talk/app/webrtc/jsep.h",
- "source/talk/app/webrtc/jsepicecandidate.cc",
- "source/talk/app/webrtc/jsepicecandidate.h",
- "source/talk/app/webrtc/jsepsessiondescription.cc",
- "source/talk/app/webrtc/jsepsessiondescription.h",
- "source/talk/app/webrtc/localaudiosource.cc",
- "source/talk/app/webrtc/localaudiosource.h",
- "source/talk/app/webrtc/mediaconstraintsinterface.cc",
- "source/talk/app/webrtc/mediaconstraintsinterface.h",
- "source/talk/app/webrtc/mediacontroller.cc",
- "source/talk/app/webrtc/mediacontroller.h",
- "source/talk/app/webrtc/mediastream.cc",
- "source/talk/app/webrtc/mediastream.h",
- "source/talk/app/webrtc/mediastreamhandler.cc",
- "source/talk/app/webrtc/mediastreamhandler.h",
- "source/talk/app/webrtc/mediastreaminterface.h",
- "source/talk/app/webrtc/mediastreamobserver.cc",
- "source/talk/app/webrtc/mediastreamobserver.h",
- "source/talk/app/webrtc/mediastreamprovider.h",
- "source/talk/app/webrtc/mediastreamproxy.h",
- "source/talk/app/webrtc/mediastreamtrack.h",
- "source/talk/app/webrtc/mediastreamtrackproxy.h",
- "source/talk/app/webrtc/notifier.h",
- "source/talk/app/webrtc/peerconnection.cc",
- "source/talk/app/webrtc/peerconnection.h",
- "source/talk/app/webrtc/peerconnectionfactory.cc",
- "source/talk/app/webrtc/peerconnectionfactory.h",
- "source/talk/app/webrtc/peerconnectioninterface.h",
- "source/talk/app/webrtc/portallocatorfactory.cc",
- "source/talk/app/webrtc/portallocatorfactory.h",
- "source/talk/app/webrtc/remoteaudiosource.cc",
- "source/talk/app/webrtc/remoteaudiosource.h",
- "source/talk/app/webrtc/remoteaudiotrack.cc",
- "source/talk/app/webrtc/remoteaudiotrack.h",
- "source/talk/app/webrtc/remotevideocapturer.cc",
- "source/talk/app/webrtc/remotevideocapturer.h",
- "source/talk/app/webrtc/rtpreceiver.cc",
- "source/talk/app/webrtc/rtpreceiver.h",
- "source/talk/app/webrtc/rtpreceiverinterface.h",
- "source/talk/app/webrtc/rtpsender.cc",
- "source/talk/app/webrtc/rtpsender.h",
- "source/talk/app/webrtc/rtpsenderinterface.h",
- "source/talk/app/webrtc/sctputils.cc",
- "source/talk/app/webrtc/sctputils.h",
- "source/talk/app/webrtc/statscollector.cc",
- "source/talk/app/webrtc/statscollector.h",
- "source/talk/app/webrtc/statstypes.cc",
- "source/talk/app/webrtc/statstypes.h",
- "source/talk/app/webrtc/streamcollection.h",
- "source/talk/app/webrtc/umametrics.h",
- "source/talk/app/webrtc/videosource.cc",
- "source/talk/app/webrtc/videosource.h",
- "source/talk/app/webrtc/videosourceinterface.h",
- "source/talk/app/webrtc/videosourceproxy.h",
- "source/talk/app/webrtc/videotrack.cc",
- "source/talk/app/webrtc/videotrack.h",
- "source/talk/app/webrtc/videotrackrenderers.cc",
- "source/talk/app/webrtc/videotrackrenderers.h",
- "source/talk/app/webrtc/webrtcsdp.cc",
- "source/talk/app/webrtc/webrtcsdp.h",
- "source/talk/app/webrtc/webrtcsession.cc",
- "source/talk/app/webrtc/webrtcsession.h",
- "source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc",
- "source/talk/app/webrtc/webrtcsessiondescriptionfactory.h",
"source/talk/session/media/audiomonitor.cc",
"source/talk/session/media/audiomonitor.h",
"source/talk/session/media/bundlefilter.cc",
« no previous file with comments | « remoting/protocol/webrtc_video_stream.cc ('k') | third_party/libjingle/README.chromium » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698