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Issue 1365043002: Set RtcpSender transport at construction. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase + cleanup Created 5 years, 2 months ago
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1 # Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
2 # 2 #
3 # Use of this source code is governed by a BSD-style license 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 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 5 # tree. An additional intellectual property rights grant can be found
6 # in the file PATENTS. All contributing project authors may 6 # in the file PATENTS. All contributing project authors may
7 # be found in the AUTHORS file in the root of the source tree. 7 # be found in the AUTHORS file in the root of the source tree.
8 8
9 { 9 {
10 'includes': [ 10 'includes': [
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86 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', 86 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
87 '<(webrtc_root)/common.gyp:webrtc_common', 87 '<(webrtc_root)/common.gyp:webrtc_common',
88 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', 88 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
89 '<(webrtc_root)/modules/modules.gyp:video_capture', 89 '<(webrtc_root)/modules/modules.gyp:video_capture',
90 '<(webrtc_root)/modules/video_coding/codecs/vp8/vp8.gyp:webrtc_vp8', 90 '<(webrtc_root)/modules/video_coding/codecs/vp8/vp8.gyp:webrtc_vp8',
91 '<(webrtc_root)/modules/video_coding/codecs/vp9/vp9.gyp:webrtc_vp9', 91 '<(webrtc_root)/modules/video_coding/codecs/vp9/vp9.gyp:webrtc_vp9',
92 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers' , 92 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers' ,
93 '<(webrtc_root)/test/test.gyp:frame_generator', 93 '<(webrtc_root)/test/test.gyp:frame_generator',
94 '<(webrtc_root)/test/test.gyp:rtp_test_utils', 94 '<(webrtc_root)/test/test.gyp:rtp_test_utils',
95 '<(webrtc_root)/test/test.gyp:test_support_main', 95 '<(webrtc_root)/test/test.gyp:test_support_main',
96 '<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
96 '<(webrtc_root)/tools/tools.gyp:agc_test_utils', 97 '<(webrtc_root)/tools/tools.gyp:agc_test_utils',
97 ], 98 ],
98 'sources': [ 99 'sources': [
99 'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc', 100 'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc',
100 'audio_coding/main/acm2/acm_receiver_unittest.cc', 101 'audio_coding/main/acm2/acm_receiver_unittest.cc',
101 'audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc', 102 'audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc',
102 'audio_coding/main/acm2/audio_coding_module_unittest.cc', 103 'audio_coding/main/acm2/audio_coding_module_unittest.cc',
103 'audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc', 104 'audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc',
104 'audio_coding/main/acm2/call_statistics_unittest.cc', 105 'audio_coding/main/acm2/call_statistics_unittest.cc',
105 'audio_coding/main/acm2/codec_owner_unittest.cc', 106 'audio_coding/main/acm2/codec_owner_unittest.cc',
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545 'sources': [ 546 'sources': [
546 'video_render_tests.isolate', 547 'video_render_tests.isolate',
547 ], 548 ],
548 }, 549 },
549 ], 550 ],
550 }], 551 }],
551 ], 552 ],
552 }], # include_tests 553 }], # include_tests
553 ], # conditions 554 ], # conditions
554 } 555 }
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