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Unified Diff: third_party/libjingle/BUILD.gn

Issue 1669023004: Roll WebRTC 11486:11495, Libjingle 11485:11495 (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Update includes in content and remoting Created 4 years, 10 months ago
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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" ]
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