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| 1 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 1 # Copyright (c) 2016 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 if (is_android) { | 10 if (is_android) { |
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| 47 "rtcpmuxfilter.cc", | 47 "rtcpmuxfilter.cc", |
| 48 "rtcpmuxfilter.h", | 48 "rtcpmuxfilter.h", |
| 49 "rtptransport.cc", | 49 "rtptransport.cc", |
| 50 "rtptransport.h", | 50 "rtptransport.h", |
| 51 "srtpfilter.cc", | 51 "srtpfilter.cc", |
| 52 "srtpfilter.h", | 52 "srtpfilter.h", |
| 53 "voicechannel.h", | 53 "voicechannel.h", |
| 54 ] | 54 ] |
| 55 | 55 |
| 56 deps = [ | 56 deps = [ |
| 57 "..:webrtc_common", |
| 57 "../api:call_api", | 58 "../api:call_api", |
| 59 "../api:libjingle_peerconnection_api", |
| 60 "../api:ortc_api", |
| 58 "../base:rtc_base", | 61 "../base:rtc_base", |
| 62 "../common_video:common_video", |
| 59 "../media", | 63 "../media", |
| 64 "../p2p:rtc_p2p", |
| 60 ] | 65 ] |
| 61 | 66 |
| 62 if (rtc_build_libsrtp) { | 67 if (rtc_build_libsrtp) { |
| 63 deps += [ "//third_party/libsrtp" ] | 68 deps += [ "//third_party/libsrtp" ] |
| 64 } | 69 } |
| 65 | 70 |
| 66 public_configs = [ ":rtc_pc_config" ] | 71 public_configs = [ ":rtc_pc_config" ] |
| 67 | 72 |
| 68 if (!build_with_chromium && is_clang) { | 73 if (!build_with_chromium && is_clang) { |
| 69 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 74 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
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| 135 | 140 |
| 136 configs += [ ":libjingle_peerconnection_warnings_config" ] | 141 configs += [ ":libjingle_peerconnection_warnings_config" ] |
| 137 | 142 |
| 138 if (!build_with_chromium && is_clang) { | 143 if (!build_with_chromium && is_clang) { |
| 139 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 144 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 140 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 145 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 141 } | 146 } |
| 142 | 147 |
| 143 deps = [ | 148 deps = [ |
| 144 ":rtc_pc", | 149 ":rtc_pc", |
| 150 "..:webrtc_common", |
| 145 "../api:call_api", | 151 "../api:call_api", |
| 146 "../api:rtc_stats_api", | 152 "../api:rtc_stats_api", |
| 153 "../api/audio_codecs:builtin_audio_decoder_factory", |
| 154 "../api/audio_codecs:builtin_audio_encoder_factory", |
| 147 "../api/video_codecs:video_codecs_api", | 155 "../api/video_codecs:video_codecs_api", |
| 156 "../base:rtc_base", |
| 157 "../base:rtc_base_approved", |
| 148 "../call", | 158 "../call", |
| 159 "../logging:rtc_event_log_api", |
| 149 "../media", | 160 "../media", |
| 161 "../modules/audio_device:audio_device", |
| 162 "../p2p:rtc_p2p", |
| 150 "../stats", | 163 "../stats", |
| 164 "../system_wrappers:system_wrappers", |
| 151 ] | 165 ] |
| 152 | 166 |
| 153 public_deps = [ | 167 public_deps = [ |
| 154 "../api:libjingle_peerconnection_api", | 168 "../api:libjingle_peerconnection_api", |
| 155 ] | 169 ] |
| 156 | 170 |
| 157 if (rtc_use_quic) { | 171 if (rtc_use_quic) { |
| 158 sources += [ | 172 sources += [ |
| 159 "quicdatachannel.cc", | 173 "quicdatachannel.cc", |
| 160 "quicdatachannel.h", | 174 "quicdatachannel.h", |
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| 201 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 215 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 202 } | 216 } |
| 203 | 217 |
| 204 if (is_win) { | 218 if (is_win) { |
| 205 libs = [ "strmiids.lib" ] | 219 libs = [ "strmiids.lib" ] |
| 206 } | 220 } |
| 207 | 221 |
| 208 deps = [ | 222 deps = [ |
| 209 ":libjingle_peerconnection", | 223 ":libjingle_peerconnection", |
| 210 ":rtc_pc", | 224 ":rtc_pc", |
| 225 "../base:rtc_base", |
| 226 "../base:rtc_base_approved", |
| 211 "../base:rtc_base_tests_main", | 227 "../base:rtc_base_tests_main", |
| 212 "../base:rtc_base_tests_utils", | 228 "../base:rtc_base_tests_utils", |
| 229 "../logging:rtc_event_log_api", |
| 230 "../media:rtc_media_base", |
| 213 "../media:rtc_media_tests_utils", | 231 "../media:rtc_media_tests_utils", |
| 232 "../p2p:p2p_test_utils", |
| 233 "../p2p:rtc_p2p", |
| 214 "../system_wrappers:metrics_default", | 234 "../system_wrappers:metrics_default", |
| 215 ] | 235 ] |
| 216 | 236 |
| 217 if (rtc_build_libsrtp) { | 237 if (rtc_build_libsrtp) { |
| 218 deps += [ "//third_party/libsrtp" ] | 238 deps += [ "//third_party/libsrtp" ] |
| 219 } | 239 } |
| 220 | 240 |
| 221 if (is_android) { | 241 if (is_android) { |
| 222 deps += [ "//testing/android/native_test:native_test_support" ] | 242 deps += [ "//testing/android/native_test:native_test_support" ] |
| 223 } | 243 } |
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| 238 "test/mock_webrtcsession.h", | 258 "test/mock_webrtcsession.h", |
| 239 "test/mockpeerconnectionobservers.h", | 259 "test/mockpeerconnectionobservers.h", |
| 240 "test/peerconnectiontestwrapper.cc", | 260 "test/peerconnectiontestwrapper.cc", |
| 241 "test/peerconnectiontestwrapper.h", | 261 "test/peerconnectiontestwrapper.h", |
| 242 "test/rtcstatsobtainer.h", | 262 "test/rtcstatsobtainer.h", |
| 243 "test/testsdpstrings.h", | 263 "test/testsdpstrings.h", |
| 244 ] | 264 ] |
| 245 | 265 |
| 246 deps = [ | 266 deps = [ |
| 247 ":libjingle_peerconnection", | 267 ":libjingle_peerconnection", |
| 268 "..:webrtc_common", |
| 269 "../api:libjingle_peerconnection_test_api", |
| 270 "../api:rtc_stats_api", |
| 271 "../base:rtc_base", |
| 272 "../base:rtc_base_approved", |
| 248 "../base:rtc_base_tests_utils", | 273 "../base:rtc_base_tests_utils", |
| 274 "../media:rtc_media", |
| 275 "../media:rtc_media_tests_utils", |
| 276 "../modules/audio_device:audio_device", |
| 277 "../p2p:p2p_test_utils", |
| 278 "../test:test_support", |
| 249 "//testing/gmock", | 279 "//testing/gmock", |
| 250 ] | 280 ] |
| 251 | 281 |
| 252 if (!build_with_chromium && is_clang) { | 282 if (!build_with_chromium && is_clang) { |
| 253 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 283 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 254 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 284 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 255 } | 285 } |
| 256 } | 286 } |
| 257 | 287 |
| 258 config("peerconnection_unittests_config") { | 288 config("peerconnection_unittests_config") { |
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| 362 "//testing/gmock", | 392 "//testing/gmock", |
| 363 ] | 393 ] |
| 364 | 394 |
| 365 if (is_android) { | 395 if (is_android) { |
| 366 deps += [ "//testing/android/native_test:native_test_support" ] | 396 deps += [ "//testing/android/native_test:native_test_support" ] |
| 367 | 397 |
| 368 shard_timeout = 900 | 398 shard_timeout = 900 |
| 369 } | 399 } |
| 370 } | 400 } |
| 371 } | 401 } |
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