| # 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', |
| 'audio_state.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 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', |
| ], |
| }], |
| ], |
| }, |
| |
| ], |
| } |