commit 2c2c62ccaf8bd43ef03a4dbac2896bc04fe30ec2
author Chung-Cheng Lou <roylou@google.com> Wed Mar 07 16:12:15 2012 -0800
committer Chung-Cheng Lou <roylou@google.com> Sun Mar 18 16:31:31 2012 -0700
tree 25001c3405a53ae30f2376a722de7d710ed6ee12
parent 5710e2d1d86eb8c9e47ad7197ee137f2f45bda47

Turn on webrtc related file build and hook webrtc library

Change-Id: I294ab34f32c60be039be97bc3ceaa09bdab59142

talk/libjingle.scons

diff --git a/talk/libjingle.scons b/talk/libjingle.scons
index ae8497a..9deef23 100644
--- a/talk/libjingle.scons
+++ b/talk/libjingle.scons
@@ -123,7 +123,8 @@
                "EXPAT_RELATIVE_PATH",
                "SRTP_RELATIVE_PATH",
                "XML_STATIC",
-               "HAVE_WEBRTC_AUDIO",
+               "HAVE_WEBRTC_VOICE",
+               "HAVE_WEBRTC_VIDEO",
              ],
              srcs = [
                "base/asyncfile.cc",
@@ -258,6 +259,11 @@
                "session/phone/videocapturer.cc",
                "session/phone/videocommon.cc",
                "session/phone/videoframe.cc",
+               "session/phone/webrtcpassthroughrender.cc",
+               "session/phone/webrtcvideocapturer.cc",
+               "session/phone/webrtcvideoengine.cc",
+               "session/phone/webrtcvideoframe.cc",
+               "session/phone/webrtcvoiceengine.cc",
                "sound/nullsoundsystem.cc",
                "sound/nullsoundsystemfactory.cc",
                "sound/platformsoundsystem.cc",
@@ -421,6 +427,7 @@
            "srtp",
            "xmpphelp",
            "VoiceEngine_Linux_gcc",
+           "WebRtcMediaEngine",
            "vcme_audio",
            "nexus",
          ],
@@ -601,6 +608,11 @@
                 "session/phone/testutils.cc",
                 "session/phone/videocapturer_unittest.cc",
                 "session/phone/videocommon_unittest.cc",
+                "session/phone/webrtcpassthroughrender_unittest.cc",
+                "session/phone/webrtcvideocapturer_unittest.cc",
+                "session/phone/webrtcvideoengine_unittest.cc",
+                "session/phone/webrtcvideoframe_unittest.cc",
+                "session/phone/webrtcvoiceengine_unittest.cc",
               ],
               includedirs = [
                 "third_party/gtest/include",

talk/main.scons

diff --git a/talk/main.scons b/talk/main.scons
index 7692cd7..588e695 100644
--- a/talk/main.scons
+++ b/talk/main.scons
@@ -67,7 +67,8 @@
     'FEATURE_ENABLE_VOICEMAIL',
     'FEATURE_ENABLE_PSTN',
     'HAVE_SRTP',
-    'HAVE_WEBRTC_AUDIO',
+    'HAVE_WEBRTC_VOICE',
+    'HAVE_WEBRTC_VIDEO',
   ],
   # Ensure the os environment is captured for any scripts we call out to
   ENV = os.environ,