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author | ronghuawu@google.com <ronghuawu@google.com@0039d316-1c4b-4281-b951-d872f2087c98> | 2011-09-27 23:27:45 +0000 |
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committer | ronghuawu@google.com <ronghuawu@google.com@0039d316-1c4b-4281-b951-d872f2087c98> | 2011-09-27 23:27:45 +0000 |
commit | 780f1c5109a2a4485723de2f79e9ad4a87f8a594 (patch) | |
tree | 634b28046102de52b17d7b7be82688b5a1c9bd55 /third_party | |
parent | eb5302298d3ce64994f6921537c4ef951f3b4d4c (diff) | |
download | chromium_src-780f1c5109a2a4485723de2f79e9ad4a87f8a594.zip chromium_src-780f1c5109a2a4485723de2f79e9ad4a87f8a594.tar.gz chromium_src-780f1c5109a2a4485723de2f79e9ad4a87f8a594.tar.bz2 |
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.gyp | 73 |
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 ], } |