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Issue 3014543002: Fix Gn untracked headers in webrtc/call
Patch Set: added missing dependency Created 3 years, 3 months ago
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1 # Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2015 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("../webrtc.gni") 9 import("../webrtc.gni")
10 10
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81 "rtp_transport_controller_send.h", 81 "rtp_transport_controller_send.h",
82 ] 82 ]
83 deps = [ 83 deps = [
84 ":rtp_interfaces", 84 ":rtp_interfaces",
85 "..:webrtc_common", 85 "..:webrtc_common",
86 "../modules/congestion_controller", 86 "../modules/congestion_controller",
87 "../rtc_base:rtc_base_approved", 87 "../rtc_base:rtc_base_approved",
88 ] 88 ]
89 } 89 }
90 90
91 rtc_source_set("bitrate_allocator_header") {
mbonadei 2017/09/19 09:00:42 Is it possible to move this into the "call" target
kjellander_webrtc 2017/09/19 16:01:43 Yeah, let's ask owners for advice here.
92 sources = [
93 "bitrate_allocator.h",
94 ]
95 deps = [
96 "../rtc_base:sequenced_task_checker",
97 ]
98 }
99
91 rtc_static_library("call") { 100 rtc_static_library("call") {
92 sources = [ 101 sources = [
93 "bitrate_allocator.cc", 102 "bitrate_allocator.cc",
94 "call.cc", 103 "call.cc",
95 "callfactory.cc", 104 "callfactory.cc",
96 "callfactory.h", 105 "callfactory.h",
97 "flexfec_receive_stream_impl.cc", 106 "flexfec_receive_stream_impl.cc",
98 "flexfec_receive_stream_impl.h", 107 "flexfec_receive_stream_impl.h",
99 ] 108 ]
100 109
101 if (!build_with_chromium && is_clang) { 110 if (!build_with_chromium && is_clang) {
102 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 111 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
103 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 112 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
104 } 113 }
105 114
106 public_deps = [ 115 public_deps = [
107 ":call_interfaces", 116 ":call_interfaces",
108 "../api:call_api", 117 "../api:call_api",
109 "../api:libjingle_peerconnection_api", 118 "../api:libjingle_peerconnection_api",
110 ] 119 ]
111 120
112 deps = [ 121 deps = [
122 ":bitrate_allocator_header",
113 ":call_interfaces", 123 ":call_interfaces",
114 ":rtp_interfaces", 124 ":rtp_interfaces",
115 ":rtp_receiver", 125 ":rtp_receiver",
116 ":rtp_sender", 126 ":rtp_sender",
117 ":video_stream_api", 127 ":video_stream_api",
118 "..:webrtc_common", 128 "..:webrtc_common",
119 "../api:optional", 129 "../api:optional",
120 "../api:transport_api", 130 "../api:transport_api",
121 "../audio", 131 "../audio",
122 "../logging:rtc_event_log_api", 132 "../logging:rtc_event_log_api",
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161 # Skip restricting visibility on mobile platforms since the tests on those 171 # Skip restricting visibility on mobile platforms since the tests on those
162 # gets additional generated targets which would require many lines here to 172 # gets additional generated targets which would require many lines here to
163 # cover (which would be confusing to read and hard to maintain). 173 # cover (which would be confusing to read and hard to maintain).
164 if (!is_android && !is_ios) { 174 if (!is_android && !is_ios) {
165 visibility = [ "..:video_engine_tests" ] 175 visibility = [ "..:video_engine_tests" ]
166 } 176 }
167 sources = [ 177 sources = [
168 "bitrate_allocator_unittest.cc", 178 "bitrate_allocator_unittest.cc",
169 "bitrate_estimator_tests.cc", 179 "bitrate_estimator_tests.cc",
170 "call_unittest.cc", 180 "call_unittest.cc",
181 "fake_rtp_transport_controller_send.h",
171 "flexfec_receive_stream_unittest.cc", 182 "flexfec_receive_stream_unittest.cc",
172 "rtcp_demuxer_unittest.cc", 183 "rtcp_demuxer_unittest.cc",
173 "rtp_demuxer_unittest.cc", 184 "rtp_demuxer_unittest.cc",
174 "rtp_rtcp_demuxer_helper_unittest.cc", 185 "rtp_rtcp_demuxer_helper_unittest.cc",
175 "rtx_receive_stream_unittest.cc", 186 "rtx_receive_stream_unittest.cc",
176 ] 187 ]
177 deps = [ 188 deps = [
189 ":bitrate_allocator_header",
178 ":call", 190 ":call",
179 ":mock_rtp_interfaces", 191 ":mock_rtp_interfaces",
180 ":rtp_interfaces", 192 ":rtp_interfaces",
181 ":rtp_receiver", 193 ":rtp_receiver",
182 ":rtp_sender", 194 ":rtp_sender",
183 "..:webrtc_common", 195 "..:webrtc_common",
184 "../api:array_view", 196 "../api:array_view",
185 "../api:mock_audio_mixer", 197 "../api:mock_audio_mixer",
186 "../logging:rtc_event_log_api", 198 "../logging:rtc_event_log_api",
187 "../modules/audio_device:mock_audio_device", 199 "../modules/audio_device:mock_audio_device",
188 "../modules/audio_mixer", 200 "../modules/audio_mixer",
189 "../modules/bitrate_controller", 201 "../modules/bitrate_controller",
202 "../modules/congestion_controller:congestion_controller",
190 "../modules/congestion_controller:mock_congestion_controller", 203 "../modules/congestion_controller:mock_congestion_controller",
191 "../modules/pacing", 204 "../modules/pacing",
192 "../modules/pacing:mock_paced_sender", 205 "../modules/pacing:mock_paced_sender",
193 "../modules/rtp_rtcp", 206 "../modules/rtp_rtcp",
194 "../modules/rtp_rtcp:mock_rtp_rtcp", 207 "../modules/rtp_rtcp:mock_rtp_rtcp",
195 "../modules/utility:mock_process_thread", 208 "../modules/utility:mock_process_thread",
196 "../rtc_base:rtc_base_approved", 209 "../rtc_base:rtc_base_approved",
197 "../system_wrappers", 210 "../system_wrappers",
198 "../test:audio_codec_mocks", 211 "../test:audio_codec_mocks",
199 "../test:direct_transport", 212 "../test:direct_transport",
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258 sources = [ 271 sources = [
259 "test/mock_rtp_packet_sink_interface.h", 272 "test/mock_rtp_packet_sink_interface.h",
260 ] 273 ]
261 deps = [ 274 deps = [
262 ":rtp_interfaces", 275 ":rtp_interfaces",
263 "../test:test_support", 276 "../test:test_support",
264 "//testing/gmock", 277 "//testing/gmock",
265 ] 278 ]
266 } 279 }
267 } 280 }
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