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Side by Side Diff: webrtc/modules/rtp_rtcp/BUILD.gn

Issue 2400993002: Fix chromium-style warnings. (Closed)
Patch Set: Created 4 years, 2 months ago
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1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2014 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 import("../../build/webrtc.gni") 9 import("../../build/webrtc.gni")
10 10
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126 "source/rtp_payload_registry.cc", 126 "source/rtp_payload_registry.cc",
127 "source/rtp_receiver_audio.cc", 127 "source/rtp_receiver_audio.cc",
128 "source/rtp_receiver_audio.h", 128 "source/rtp_receiver_audio.h",
129 "source/rtp_receiver_impl.cc", 129 "source/rtp_receiver_impl.cc",
130 "source/rtp_receiver_impl.h", 130 "source/rtp_receiver_impl.h",
131 "source/rtp_receiver_strategy.cc", 131 "source/rtp_receiver_strategy.cc",
132 "source/rtp_receiver_strategy.h", 132 "source/rtp_receiver_strategy.h",
133 "source/rtp_receiver_video.cc", 133 "source/rtp_receiver_video.cc",
134 "source/rtp_receiver_video.h", 134 "source/rtp_receiver_video.h",
135 "source/rtp_rtcp_config.h", 135 "source/rtp_rtcp_config.h",
136 "source/rtp_rtcp_defines_nullimpl.cc",
danilchap 2016/10/07 12:42:49 may be call this files null_rtp_callbacks
terelius 2016/10/07 15:03:37 My reasoning for choosing this name is that the fi
danilchap 2016/10/07 17:22:00 might make sense to move the remaining object into
137 "source/rtp_rtcp_defines_nullimpl.h",
136 "source/rtp_rtcp_impl.cc", 138 "source/rtp_rtcp_impl.cc",
137 "source/rtp_rtcp_impl.h", 139 "source/rtp_rtcp_impl.h",
138 "source/rtp_sender.cc", 140 "source/rtp_sender.cc",
139 "source/rtp_sender.h", 141 "source/rtp_sender.h",
140 "source/rtp_sender_audio.cc", 142 "source/rtp_sender_audio.cc",
141 "source/rtp_sender_audio.h", 143 "source/rtp_sender_audio.h",
142 "source/rtp_sender_video.cc", 144 "source/rtp_sender_video.cc",
143 "source/rtp_sender_video.h", 145 "source/rtp_sender_video.h",
144 "source/rtp_utility.cc", 146 "source/rtp_utility.cc",
145 "source/rtp_utility.h", 147 "source/rtp_utility.h",
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195 "test/testFec/test_packet_masks_metrics.cc", 197 "test/testFec/test_packet_masks_metrics.cc",
196 ] 198 ]
197 199
198 deps = [ 200 deps = [
199 ":rtp_rtcp", 201 ":rtp_rtcp",
200 "//testing/gtest", 202 "//testing/gtest",
201 "//webrtc/test:test_support_main", 203 "//webrtc/test:test_support_main",
202 ] 204 ]
203 } # test_packet_masks_metrics 205 } # test_packet_masks_metrics
204 } 206 }
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