Update webrtc to 2565.

Review URL: https://chromiumcodereview.appspot.com/10837131

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@151214 0039d316-1c4b-4281-b951-d872f2087c98

DEPS

@@ -34,7 +34,7 @@ vars = {
   # the commit queue can handle CLs rolling Skia
   # and V8 without interference from each other.
   "v8_revision": "12295",
-  "webrtc_revision": "2549",
+  "webrtc_revision": "2565",
   "jsoncpp_revision": "248",
   "nss_revision": "145873",
 }

content/renderer/media/webrtc_audio_device_unittest.cc

@@ -10,11 +10,11 @@
 #include "media/audio/audio_manager.h"
 #include "media/audio/audio_util.h"
 #include "testing/gmock/include/gmock/gmock.h"
-#include "third_party/webrtc/voice_engine/main/interface/voe_audio_processing.h"
-#include "third_party/webrtc/voice_engine/main/interface/voe_base.h"
-#include "third_party/webrtc/voice_engine/main/interface/voe_external_media.h"
-#include "third_party/webrtc/voice_engine/main/interface/voe_file.h"
-#include "third_party/webrtc/voice_engine/main/interface/voe_network.h"
+#include "third_party/webrtc/voice_engine/include/voe_audio_processing.h"
+#include "third_party/webrtc/voice_engine/include/voe_base.h"
+#include "third_party/webrtc/voice_engine/include/voe_external_media.h"
+#include "third_party/webrtc/voice_engine/include/voe_file.h"
+#include "third_party/webrtc/voice_engine/include/voe_network.h"
 
 using testing::_;
 using testing::AnyNumber;

content/test/webrtc_audio_device_test.cc

@@ -34,10 +34,10 @@
 #include "net/url_request/url_request_test_util.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "third_party/webrtc/voice_engine/main/interface/voe_audio_processing.h"
-#include "third_party/webrtc/voice_engine/main/interface/voe_base.h"
-#include "third_party/webrtc/voice_engine/main/interface/voe_file.h"
-#include "third_party/webrtc/voice_engine/main/interface/voe_network.h"
+#include "third_party/webrtc/voice_engine/include/voe_audio_processing.h"
+#include "third_party/webrtc/voice_engine/include/voe_base.h"
+#include "third_party/webrtc/voice_engine/include/voe_file.h"
+#include "third_party/webrtc/voice_engine/include/voe_network.h"
 
 using base::win::ScopedCOMInitializer;
 using testing::_;

third_party/libjingle/libjingle.gyp

@@ -19,7 +19,6 @@
       'NO_MAIN_THREAD_WRAPPING',
       'NO_SOUND_SYSTEM',
       'SRTP_RELATIVE_PATH',
-      'WEBRTC_RELATIVE_PATH',
       '_USE_32BIT_TIME_T',
     ],
     'configurations': {
@@ -36,6 +35,7 @@
       './source',
       '../../testing/gtest/include',
       '../../third_party/libyuv/include',
+      '../../third_party/webrtc',
     ],
     'dependencies': [
       '<(DEPTH)/base/base.gyp:base',
@@ -50,6 +50,7 @@
         './overrides',
         './source',
         '../../testing/gtest/include',
+        '../../third_party/webrtc',
       ],
       'defines': [
         'FEATURE_ENABLE_SSL',
@@ -57,7 +58,6 @@
         'EXPAT_RELATIVE_PATH',
         'GTEST_RELATIVE_PATH',
         'JSONCPP_RELATIVE_PATH',
-        'WEBRTC_RELATIVE_PATH',
         'NO_MAIN_THREAD_WRAPPING',
         'NO_SOUND_SYSTEM',
       ],