Includes webrtc/build/protoc.gypi instead of build/protoc.gypi

webrtc/webrtc.gyp

 # Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 {
   'conditions': [
     ['include_tests==1', {
       'includes': [
         'libjingle/xmllite/xmllite_tests.gypi',
         'libjingle/xmpp/xmpp_tests.gypi',
         'p2p/p2p_tests.gypi',
         'sound/sound_tests.gypi',
         'webrtc_tests.gypi',
       ],
     }],
+    ['enable_protobuf==1', {
+      'targets': [
+        {
+          # This target should only be built if enable_protobuf is defined
+          'target_name': 'rtc_event_log_proto',
+          'type': 'static_library',
+          'sources': ['video/rtc_event_log.proto',],
+          'variables': {
+            'proto_in_dir': 'video',
+            'proto_out_dir': 'webrtc/video',
+          },
+        'includes': ['build/protoc.gypi'],
+        },
+      ],
+    }],
   ],
   'includes': [
     'build/common.gypi',
     'video/webrtc_video.gypi',
   ],
   'variables': {
     'webrtc_all_dependencies': [
       'base/base.gyp:*',
       'sound/sound.gyp:*',
       'common.gyp:*',
@@ skipping 44 matching lines @@
         'transport.h',
         'video_receive_stream.h',
         'video_renderer.h',
         'video_send_stream.h',
 
         '<@(webrtc_video_sources)',
       ],
       'dependencies': [
         'common.gyp:*',
         '<@(webrtc_video_dependencies)',
+        'rtc_event_log',
       ],
       'conditions': [
         # TODO(andresp): Chromium libpeerconnection should link directly with
         # this and no if conditions should be needed on webrtc build files.
         ['build_with_chromium==1', {
           'dependencies': [
             '<(webrtc_root)/modules/modules.gyp:video_capture',
             '<(webrtc_root)/modules/modules.gyp:video_render',
           ],
         }],
       ],
     },
+    {
+      'target_name': 'rtc_event_log',

[kjellander_webrtc, 2015/09/15 13:26:33]

Putting this target here, combined with
https://codereview.webrtc.org/1267683002/ introduces more circular dependencies
to WebRTC, which something we would rather try to get rid of (see
https://code.google.com/p/webrtc/issues/detail?id=4243).

Is it maybe possible to move the rtc_event_log into webrtc/tools instead?

+      'type': 'static_library',
+      'sources': [
+        'video/rtc_event_log.cc',
+        'video/rtc_event_log.h',
+      ],
+      'conditions': [
+        # If enable_protobuf is defined, we want to compile the protobuf
+        # and add rtc_event_log.pb.h and rtc_event_log.pb.cc to the sources.
+        ['enable_protobuf==1', {
+          'dependencies': [
+            'rtc_event_log_proto',
+          ],
+          'defines': [
+            'ENABLE_RTC_EVENT_LOG',
+          ],
+        }],
+      ],
+    },
+
   ],
 }