| # 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': ['call/rtc_event_log.proto',], | 
 |           'variables': { | 
 |             'proto_in_dir': 'call', | 
 |             'proto_out_dir': 'webrtc/call', | 
 |           }, | 
 |         'includes': ['build/protoc.gypi'], | 
 |         }, | 
 |       ], | 
 |     }], | 
 |     ['include_tests==1 and enable_protobuf==1', { | 
 |       'targets': [ | 
 |         { | 
 |           'target_name': 'rtc_event_log2rtp_dump', | 
 |           'type': 'executable', | 
 |           'sources': ['call/rtc_event_log2rtp_dump.cc',], | 
 |           'dependencies': [ | 
 |             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', | 
 |             'rtc_event_log', | 
 |             'rtc_event_log_proto', | 
 |             'test/test.gyp:rtp_test_utils' | 
 |           ], | 
 |         }, | 
 |       ], | 
 |     }], | 
 |   ], | 
 |   'includes': [ | 
 |     'build/common.gypi', | 
 |     'audio/webrtc_audio.gypi', | 
 |     'call/webrtc_call.gypi', | 
 |     'video/webrtc_video.gypi', | 
 |   ], | 
 |   'variables': { | 
 |     'webrtc_all_dependencies': [ | 
 |       'base/base.gyp:*', | 
 |       'sound/sound.gyp:*', | 
 |       'common.gyp:*', | 
 |       'common_audio/common_audio.gyp:*', | 
 |       'common_video/common_video.gyp:*', | 
 |       'modules/modules.gyp:*', | 
 |       'p2p/p2p.gyp:*', | 
 |       'system_wrappers/system_wrappers.gyp:*', | 
 |       'tools/tools.gyp:*', | 
 |       'voice_engine/voice_engine.gyp:*', | 
 |       '<(webrtc_vp8_dir)/vp8.gyp:*', | 
 |       '<(webrtc_vp9_dir)/vp9.gyp:*', | 
 |     ], | 
 |   }, | 
 |   'targets': [ | 
 |     { | 
 |       'target_name': 'webrtc_all', | 
 |       'type': 'none', | 
 |       'dependencies': [ | 
 |         '<@(webrtc_all_dependencies)', | 
 |         'webrtc', | 
 |       ], | 
 |       'conditions': [ | 
 |         ['include_tests==1', { | 
 |           'dependencies': [ | 
 |             'common_video/common_video_unittests.gyp:*', | 
 |             'rtc_unittests', | 
 |             'system_wrappers/system_wrappers_tests.gyp:*', | 
 |             'test/metrics.gyp:*', | 
 |             'test/test.gyp:*', | 
 |             'test/webrtc_test_common.gyp:*', | 
 |             'video_engine/video_engine_core_unittests.gyp:*', | 
 |             'webrtc_tests', | 
 |           ], | 
 |         }], | 
 |       ], | 
 |     }, | 
 |     { | 
 |       'target_name': 'webrtc', | 
 |       'type': 'static_library', | 
 |       'sources': [ | 
 |         'audio_receive_stream.h', | 
 |         'audio_send_stream.h', | 
 |         'call.h', | 
 |         'config.h', | 
 |         'frame_callback.h', | 
 |         'stream.h', | 
 |         'transport.h', | 
 |         'video_receive_stream.h', | 
 |         'video_renderer.h', | 
 |         'video_send_stream.h', | 
 |  | 
 |         '<@(webrtc_audio_sources)', | 
 |         '<@(webrtc_call_sources)', | 
 |         '<@(webrtc_video_sources)', | 
 |       ], | 
 |       'dependencies': [ | 
 |         'common.gyp:*', | 
 |         '<@(webrtc_audio_dependencies)', | 
 |         '<@(webrtc_call_dependencies)', | 
 |         '<@(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', | 
 |       'type': 'static_library', | 
 |       'sources': [ | 
 |         'call/rtc_event_log.cc', | 
 |         'call/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', | 
 |           ], | 
 |         }], | 
 |       ], | 
 |     }, | 
 |  | 
 |   ], | 
 | } |