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authorronghuawu@google.com <ronghuawu@google.com@0039d316-1c4b-4281-b951-d872f2087c98>2011-09-27 23:27:45 +0000
committerronghuawu@google.com <ronghuawu@google.com@0039d316-1c4b-4281-b951-d872f2087c98>2011-09-27 23:27:45 +0000
commit780f1c5109a2a4485723de2f79e9ad4a87f8a594 (patch)
tree634b28046102de52b17d7b7be82688b5a1c9bd55 /third_party
parenteb5302298d3ce64994f6921537c4ef951f3b4d4c (diff)
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Add some more files (needed by webrtc/peerconnection) to libjingle targets.
TEST=Unittests Review URL: http://codereview.chromium.org/8051017 git-svn-id: svn://svn.chromium.org/chrome/trunk/src@103045 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'third_party')
-rw-r--r--third_party/libjingle/libjingle.gyp73
1 files changed, 70 insertions, 3 deletions
diff --git a/third_party/libjingle/libjingle.gyp b/third_party/libjingle/libjingle.gyp
index 0e3ca9c..7d6fb61 100644
--- a/third_party/libjingle/libjingle.gyp
+++ b/third_party/libjingle/libjingle.gyp
@@ -16,6 +16,9 @@
'_USE_32BIT_TIME_T',
'SAFE_TO_DEFINE_TALK_BASE_LOGGING_MACROS',
'EXPAT_RELATIVE_PATH',
+ 'WEBRTC_RELATIVE_PATH',
+ 'HAVE_WEBRTC_VIDEO',
+ 'HAVE_WEBRTC_VOICE',
],
'configurations': {
'Debug': {
@@ -47,6 +50,7 @@
'FEATURE_ENABLE_SSL',
'FEATURE_ENABLE_VOICEMAIL',
'EXPAT_RELATIVE_PATH',
+ 'WEBRTC_RELATIVE_PATH',
],
'conditions': [
['OS=="win"', {
@@ -351,7 +355,7 @@
],
}],
],
- },
+ }, # target libjingle
# This has to be is a separate project due to a bug in MSVS:
# https://connect.microsoft.com/VisualStudio/feedback/details/368272/duplicate-cpp-filename-in-c-project-visual-studio-2008
# We have two files named "constants.cc" and MSVS doesn't handle this
@@ -422,8 +426,71 @@
'source/talk/session/tunnel/tunnelsessionclient.h',
],
'dependencies': [
- 'libjingle',
+ 'libjingle',
],
- },
+ }, # target libjingle_p2p
+ {
+ 'target_name': 'libjingle_peerconnection',
+ 'type': 'static_library',
+ 'sources': [
+ 'source/talk/app/webrtc/peerconnection.h',
+ 'source/talk/app/webrtc/peerconnectionfactory.h',
+ 'source/talk/app/webrtc/peerconnectionfactory.cc',
+ 'source/talk/app/webrtc/peerconnectionproxy.cc',
+ 'source/talk/app/webrtc/peerconnectionproxy.h',
+ 'source/talk/session/phone/audiomonitor.cc',
+ 'source/talk/session/phone/audiomonitor.h',
+ 'source/talk/session/phone/call.cc',
+ 'source/talk/session/phone/call.h',
+ 'source/talk/session/phone/channel.cc',
+ 'source/talk/session/phone/channel.h',
+ 'source/talk/session/phone/channelmanager.cc',
+ 'source/talk/session/phone/channelmanager.h',
+ 'source/talk/session/phone/codec.cc',
+ 'source/talk/session/phone/codec.h',
+ 'source/talk/session/phone/cryptoparams.h',
+ 'source/talk/session/phone/currentspeakermonitor.cc',
+ 'source/talk/session/phone/currentspeakermonitor.h',
+ 'source/talk/session/phone/filemediaengine.cc',
+ 'source/talk/session/phone/filemediaengine.h',
+ 'source/talk/session/phone/mediachannel.h',
+ 'source/talk/session/phone/mediaengine.cc',
+ 'source/talk/session/phone/mediaengine.h',
+ 'source/talk/session/phone/mediamessages.cc',
+ 'source/talk/session/phone/mediamessages.h',
+ 'source/talk/session/phone/mediamonitor.cc',
+ 'source/talk/session/phone/mediamonitor.h',
+ 'source/talk/session/phone/mediasession.cc',
+ 'source/talk/session/phone/mediasessionclient.cc',
+ 'source/talk/session/phone/mediasessionclient.h',
+ 'source/talk/session/phone/mediasink.h',
+ 'source/talk/session/phone/rtcpmuxfilter.cc',
+ 'source/talk/session/phone/rtcpmuxfilter.h',
+ 'source/talk/session/phone/rtpdump.cc',
+ 'source/talk/session/phone/rtpdump.h',
+ 'source/talk/session/phone/rtputils.cc',
+ 'source/talk/session/phone/rtputils.h',
+ 'source/talk/session/phone/soundclip.cc',
+ 'source/talk/session/phone/soundclip.h',
+ 'source/talk/session/phone/srtpfilter.cc',
+ 'source/talk/session/phone/srtpfilter.h',
+ 'source/talk/session/phone/videocommon.h',
+ 'source/talk/session/phone/voicechannel.h',
+ 'source/talk/session/phone/webrtccommon.h',
+ 'source/talk/session/phone/webrtcpassthroughrender.cc',
+ 'source/talk/session/phone/webrtcvideoframe.cc',
+ 'source/talk/session/phone/webrtcvideoframe.h',
+ 'source/talk/session/phone/webrtcvie.h',
+ 'source/talk/session/phone/webrtcvoe.h',
+ ],
+ 'dependencies': [
+ '../../third_party/webrtc/modules/modules.gyp:video_capture_module',
+ '../../third_party/webrtc/modules/modules.gyp:video_render_module',
+ '../../third_party/webrtc/video_engine/video_engine.gyp:video_engine_core',
+ '../../third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine_core',
+ '../../third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ 'libjingle_p2p',
+ ],
+ }, # target libjingle_peerconnection
],
}