1 # Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 # 3 # Use of this source code is governed by a BSD-style license 4 # that can be found in the LICENSE file in the root of the source 5 # tree. An additional intellectual property rights grant can be found 6 # in the file PATENTS. All contributing project authors may 7 # be found in the AUTHORS file in the root of the source tree. 8 { 9 'conditions': [ 10 ['include_tests==1', { 11 'includes': [ 12 'webrtc_tests.gypi', 13 ], 14 }], 15 ], 16 'includes': [ 17 'build/common.gypi', 18 'video/webrtc_video.gypi', 19 ], 20 'variables': { 21 'webrtc_all_dependencies': [ 22 'base/base.gyp:*', 23 'sound/sound.gyp:*', 24 'common.gyp:*', 25 'common_audio/common_audio.gyp:*', 26 'common_video/common_video.gyp:*', 27 'libjingle/xmllite/xmllite.gyp:*', 28 'modules/modules.gyp:*', 29 'system_wrappers/source/system_wrappers.gyp:*', 30 'video_engine/video_engine.gyp:*', 31 'voice_engine/voice_engine.gyp:*', 32 '<(webrtc_vp8_dir)/vp8.gyp:*', 33 ], 34 }, 35 'targets': [ 36 { 37 'target_name': 'webrtc_all', 38 'type': 'none', 39 'dependencies': [ 40 '<@(webrtc_all_dependencies)', 41 'webrtc', 42 ], 43 'conditions': [ 44 ['include_tests==1', { 45 'dependencies': [ 46 'common_video/common_video_unittests.gyp:*', 47 'libjingle/xmllite/xmllite_tests.gyp:*', 48 'sound/sound_tests.gyp:*', 49 'system_wrappers/source/system_wrappers_tests.gyp:*', 50 'test/metrics.gyp:*', 51 'test/test.gyp:*', 52 'test/webrtc_test_common.gyp:webrtc_test_common_unittests', 53 'tools/tools.gyp:*', 54 'webrtc_tests', 55 'rtc_unittests', 56 ], 57 }], 58 ], 59 }, 60 { 61 # TODO(pbos): This is intended to contain audio parts as well as soon as 62 # VoiceEngine moves to the same new API format. 63 'target_name': 'webrtc', 64 'type': 'static_library', 65 'sources': [ 66 'call.h', 67 'config.h', 68 'experiments.h', 69 'frame_callback.h', 70 'transport.h', 71 'video_receive_stream.h', 72 'video_renderer.h', 73 'video_send_stream.h', 74 75 '<@(webrtc_video_sources)', 76 ], 77 'dependencies': [ 78 'common.gyp:*', 79 '<@(webrtc_video_dependencies)', 80 ], 81 'conditions': [ 82 # TODO(andresp): Chromium libpeerconnection should link directly with 83 # this and no if conditions should be needed on webrtc build files. 84 ['build_with_chromium==1', { 85 'dependencies': [ 86 '<(webrtc_root)/modules/modules.gyp:video_capture_module_impl', 87 '<(webrtc_root)/modules/modules.gyp:video_render_module_impl', 88 ], 89 }], 90 ], 91 }, 92 ], 93 } 94
