Index: third_party/libjingle/BUILD.gn |
diff --git a/third_party/libjingle/BUILD.gn b/third_party/libjingle/BUILD.gn |
index 3843e44c11dad2a2fdcaf460a9a1065c4464af89..33f44ed87afc34fb9ce21cc902a4918abf5ce5ac 100644 |
--- a/third_party/libjingle/BUILD.gn |
+++ b/third_party/libjingle/BUILD.gn |
@@ -296,6 +296,50 @@ 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/media/base/audiorenderer.h", |
+ "../webrtc/media/base/capturemanager.cc", |
+ "../webrtc/media/base/capturemanager.h", |
+ "../webrtc/media/base/capturerenderadapter.cc", |
+ "../webrtc/media/base/capturerenderadapter.h", |
+ "../webrtc/media/base/codec.cc", |
+ "../webrtc/media/base/codec.h", |
+ "../webrtc/media/base/constants.cc", |
+ "../webrtc/media/base/constants.h", |
+ "../webrtc/media/base/cryptoparams.h", |
+ "../webrtc/media/base/hybriddataengine.h", |
+ "../webrtc/media/base/mediachannel.h", |
+ "../webrtc/media/base/mediaengine.cc", |
+ "../webrtc/media/base/mediaengine.h", |
+ "../webrtc/media/base/rtpdataengine.cc", |
+ "../webrtc/media/base/rtpdataengine.h", |
+ "../webrtc/media/base/rtpdump.cc", |
+ "../webrtc/media/base/rtpdump.h", |
+ "../webrtc/media/base/rtputils.cc", |
+ "../webrtc/media/base/rtputils.h", |
+ "../webrtc/media/base/streamparams.cc", |
+ "../webrtc/media/base/streamparams.h", |
+ "../webrtc/media/base/turnutils.cc", |
+ "../webrtc/media/base/turnutils.h", |
+ "../webrtc/media/base/videoadapter.cc", |
+ "../webrtc/media/base/videoadapter.h", |
+ "../webrtc/media/base/videocapturer.cc", |
+ "../webrtc/media/base/videocapturer.h", |
+ "../webrtc/media/base/videocommon.cc", |
+ "../webrtc/media/base/videocommon.h", |
+ "../webrtc/media/base/videoframe.cc", |
+ "../webrtc/media/base/videoframe.h", |
+ "../webrtc/media/base/videoframefactory.cc", |
+ "../webrtc/media/base/videoframefactory.h", |
+ "../webrtc/media/devices/dummydevicemanager.cc", |
+ "../webrtc/media/devices/dummydevicemanager.h", |
+ "../webrtc/media/devices/filevideocapturer.cc", |
+ "../webrtc/media/devices/filevideocapturer.h", |
+ "../webrtc/media/webrtc/webrtccommon.h", |
+ "../webrtc/media/webrtc/webrtcvideoframe.cc", |
+ "../webrtc/media/webrtc/webrtcvideoframe.h", |
+ "../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", |
@@ -368,50 +412,6 @@ if (enable_webrtc) { |
"source/talk/app/webrtc/webrtcsession.h", |
"source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc", |
"source/talk/app/webrtc/webrtcsessiondescriptionfactory.h", |
- "source/talk/media/base/audiorenderer.h", |
- "source/talk/media/base/capturemanager.cc", |
- "source/talk/media/base/capturemanager.h", |
- "source/talk/media/base/capturerenderadapter.cc", |
- "source/talk/media/base/capturerenderadapter.h", |
- "source/talk/media/base/codec.cc", |
- "source/talk/media/base/codec.h", |
- "source/talk/media/base/constants.cc", |
- "source/talk/media/base/constants.h", |
- "source/talk/media/base/cryptoparams.h", |
- "source/talk/media/base/hybriddataengine.h", |
- "source/talk/media/base/mediachannel.h", |
- "source/talk/media/base/mediaengine.cc", |
- "source/talk/media/base/mediaengine.h", |
- "source/talk/media/base/rtpdataengine.cc", |
- "source/talk/media/base/rtpdataengine.h", |
- "source/talk/media/base/rtpdump.cc", |
- "source/talk/media/base/rtpdump.h", |
- "source/talk/media/base/rtputils.cc", |
- "source/talk/media/base/rtputils.h", |
- "source/talk/media/base/streamparams.cc", |
- "source/talk/media/base/streamparams.h", |
- "source/talk/media/base/turnutils.cc", |
- "source/talk/media/base/turnutils.h", |
- "source/talk/media/base/videoadapter.cc", |
- "source/talk/media/base/videoadapter.h", |
- "source/talk/media/base/videocapturer.cc", |
- "source/talk/media/base/videocapturer.h", |
- "source/talk/media/base/videocommon.cc", |
- "source/talk/media/base/videocommon.h", |
- "source/talk/media/base/videoframe.cc", |
- "source/talk/media/base/videoframe.h", |
- "source/talk/media/base/videoframefactory.cc", |
- "source/talk/media/base/videoframefactory.h", |
- "source/talk/media/devices/dummydevicemanager.cc", |
- "source/talk/media/devices/dummydevicemanager.h", |
- "source/talk/media/devices/filevideocapturer.cc", |
- "source/talk/media/devices/filevideocapturer.h", |
- "source/talk/media/webrtc/webrtccommon.h", |
- "source/talk/media/webrtc/webrtcvideoframe.cc", |
- "source/talk/media/webrtc/webrtcvideoframe.h", |
- "source/talk/media/webrtc/webrtcvideoframefactory.cc", |
- "source/talk/media/webrtc/webrtcvideoframefactory.h", |
- "source/talk/media/webrtc/webrtcvoe.h", |
"source/talk/session/media/audiomonitor.cc", |
"source/talk/session/media/audiomonitor.h", |
"source/talk/session/media/bundlefilter.cc", |
@@ -453,8 +453,8 @@ if (enable_webrtc) { |
if (!is_ios) { |
# TODO(mallinath) - Enable SCTP for iOS. |
sources += [ |
- "source/talk/media/sctp/sctpdataengine.cc", |
- "source/talk/media/sctp/sctpdataengine.h", |
+ "../webrtc/media/sctp/sctpdataengine.cc", |
+ "../webrtc/media/sctp/sctpdataengine.h", |
] |
defines = [ "HAVE_SCTP" ] |
deps += [ "//third_party/usrsctp" ] |
@@ -465,14 +465,14 @@ if (enable_webrtc) { |
# as is supported in the GYP build. It's not clear what this is used for. |
source_set("libpeerconnection") { |
sources = [ |
- "source/talk/media/webrtc/simulcast.cc", |
- "source/talk/media/webrtc/simulcast.h", |
- "source/talk/media/webrtc/webrtcmediaengine.cc", |
- "source/talk/media/webrtc/webrtcmediaengine.h", |
- "source/talk/media/webrtc/webrtcvideoengine2.cc", |
- "source/talk/media/webrtc/webrtcvideoengine2.h", |
- "source/talk/media/webrtc/webrtcvoiceengine.cc", |
- "source/talk/media/webrtc/webrtcvoiceengine.h", |
+ "../webrtc/media/webrtc/simulcast.cc", |
+ "../webrtc/media/webrtc/simulcast.h", |
+ "../webrtc/media/webrtc/webrtcmediaengine.cc", |
+ "../webrtc/media/webrtc/webrtcmediaengine.h", |
+ "../webrtc/media/webrtc/webrtcvideoengine2.cc", |
+ "../webrtc/media/webrtc/webrtcvideoengine2.h", |
+ "../webrtc/media/webrtc/webrtcvoiceengine.cc", |
+ "../webrtc/media/webrtc/webrtcvoiceengine.h", |
] |
configs += [ ":jingle_unexported_configs" ] |