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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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| 53 "srtpsession.cc", | 53 "srtpsession.cc", |
| 54 "srtpsession.h", | 54 "srtpsession.h", |
| 55 "voicechannel.h", | 55 "voicechannel.h", |
| 56 ] | 56 ] |
| 57 | 57 |
| 58 deps = [ | 58 deps = [ |
| 59 "..:webrtc_common", | 59 "..:webrtc_common", |
| 60 "../api:call_api", | 60 "../api:call_api", |
| 61 "../api:libjingle_peerconnection_api", | 61 "../api:libjingle_peerconnection_api", |
| 62 "../api:ortc_api", | 62 "../api:ortc_api", |
| 63 "../base:rtc_base", | |
| 64 "../base:rtc_task_queue", | |
| 65 "../media:rtc_data", | 63 "../media:rtc_data", |
| 66 "../media:rtc_h264_profile_id", | 64 "../media:rtc_h264_profile_id", |
| 67 "../media:rtc_media_base", | 65 "../media:rtc_media_base", |
| 68 "../p2p:rtc_p2p", | 66 "../p2p:rtc_p2p", |
| 67 "../rtc_base:rtc_base", |
| 68 "../rtc_base:rtc_task_queue", |
| 69 ] | 69 ] |
| 70 | 70 |
| 71 if (rtc_build_libsrtp) { | 71 if (rtc_build_libsrtp) { |
| 72 deps += [ "//third_party/libsrtp" ] | 72 deps += [ "//third_party/libsrtp" ] |
| 73 } | 73 } |
| 74 | 74 |
| 75 public_configs = [ ":rtc_pc_config" ] | 75 public_configs = [ ":rtc_pc_config" ] |
| 76 | 76 |
| 77 if (!build_with_chromium && is_clang) { | 77 if (!build_with_chromium && is_clang) { |
| 78 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 78 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
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| 158 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 158 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 159 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 159 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 160 } | 160 } |
| 161 | 161 |
| 162 deps = [ | 162 deps = [ |
| 163 ":rtc_pc_base", | 163 ":rtc_pc_base", |
| 164 "..:webrtc_common", | 164 "..:webrtc_common", |
| 165 "../api:call_api", | 165 "../api:call_api", |
| 166 "../api:rtc_stats_api", | 166 "../api:rtc_stats_api", |
| 167 "../api/video_codecs:video_codecs_api", | 167 "../api/video_codecs:video_codecs_api", |
| 168 "../base:rtc_base", | |
| 169 "../base:rtc_base_approved", | |
| 170 "../call:call_interfaces", | 168 "../call:call_interfaces", |
| 171 "../logging:rtc_event_log_api", | 169 "../logging:rtc_event_log_api", |
| 172 "../media:rtc_data", | 170 "../media:rtc_data", |
| 173 "../media:rtc_media_base", | 171 "../media:rtc_media_base", |
| 174 "../p2p:rtc_p2p", | 172 "../p2p:rtc_p2p", |
| 173 "../rtc_base:rtc_base", |
| 174 "../rtc_base:rtc_base_approved", |
| 175 "../stats", | 175 "../stats", |
| 176 "../system_wrappers:system_wrappers", | 176 "../system_wrappers:system_wrappers", |
| 177 ] | 177 ] |
| 178 | 178 |
| 179 public_deps = [ | 179 public_deps = [ |
| 180 "../api:libjingle_peerconnection_api", | 180 "../api:libjingle_peerconnection_api", |
| 181 ] | 181 ] |
| 182 } | 182 } |
| 183 | 183 |
| 184 # This target implements CreatePeerConnectionFactory methods that will create a | 184 # This target implements CreatePeerConnectionFactory methods that will create a |
| 185 # PeerConnection will full functionality (audio, video and data). Applications | 185 # PeerConnection will full functionality (audio, video and data). Applications |
| 186 # that wish to reduce their binary size by ommitting functionality they don't | 186 # that wish to reduce their binary size by ommitting functionality they don't |
| 187 # need should use CreateModularCreatePeerConnectionFactory instead, using the | 187 # need should use CreateModularCreatePeerConnectionFactory instead, using the |
| 188 # "peerconnection" build target and other targets specific to their | 188 # "peerconnection" build target and other targets specific to their |
| 189 # requrements. See comment in peerconnectionfactoryinterface.h. | 189 # requrements. See comment in peerconnectionfactoryinterface.h. |
| 190 rtc_static_library("create_pc_factory") { | 190 rtc_static_library("create_pc_factory") { |
| 191 sources = [ | 191 sources = [ |
| 192 "createpeerconnectionfactory.cc", | 192 "createpeerconnectionfactory.cc", |
| 193 ] | 193 ] |
| 194 | 194 |
| 195 deps = [ | 195 deps = [ |
| 196 "../api:audio_mixer_api", | 196 "../api:audio_mixer_api", |
| 197 "../api:libjingle_peerconnection_api", | 197 "../api:libjingle_peerconnection_api", |
| 198 "../api/audio_codecs:audio_codecs_api", | 198 "../api/audio_codecs:audio_codecs_api", |
| 199 "../api/audio_codecs:builtin_audio_decoder_factory", | 199 "../api/audio_codecs:builtin_audio_decoder_factory", |
| 200 "../api/audio_codecs:builtin_audio_encoder_factory", | 200 "../api/audio_codecs:builtin_audio_encoder_factory", |
| 201 "../base:rtc_base", | |
| 202 "../base:rtc_base_approved", | |
| 203 "../call", | 201 "../call", |
| 204 "../call:call_interfaces", | 202 "../call:call_interfaces", |
| 205 "../logging:rtc_event_log_api", | 203 "../logging:rtc_event_log_api", |
| 206 "../media:rtc_audio_video", | 204 "../media:rtc_audio_video", |
| 207 "../modules/audio_device:audio_device", | 205 "../modules/audio_device:audio_device", |
| 208 "../modules/audio_processing:audio_processing", | 206 "../modules/audio_processing:audio_processing", |
| 207 "../rtc_base:rtc_base", |
| 208 "../rtc_base:rtc_base_approved", |
| 209 ] | 209 ] |
| 210 | 210 |
| 211 configs += [ ":libjingle_peerconnection_warnings_config" ] | 211 configs += [ ":libjingle_peerconnection_warnings_config" ] |
| 212 | 212 |
| 213 if (!build_with_chromium && is_clang) { | 213 if (!build_with_chromium && is_clang) { |
| 214 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 214 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 215 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 215 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 216 } | 216 } |
| 217 } | 217 } |
| 218 | 218 |
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| 272 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 272 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 273 } | 273 } |
| 274 | 274 |
| 275 if (is_win) { | 275 if (is_win) { |
| 276 libs = [ "strmiids.lib" ] | 276 libs = [ "strmiids.lib" ] |
| 277 } | 277 } |
| 278 | 278 |
| 279 deps = [ | 279 deps = [ |
| 280 ":libjingle_peerconnection", | 280 ":libjingle_peerconnection", |
| 281 ":rtc_pc", | 281 ":rtc_pc", |
| 282 "../base:rtc_base", | |
| 283 "../base:rtc_base_approved", | |
| 284 "../base:rtc_base_tests_main", | |
| 285 "../base:rtc_base_tests_utils", | |
| 286 "../logging:rtc_event_log_api", | 282 "../logging:rtc_event_log_api", |
| 287 "../media:rtc_media_base", | 283 "../media:rtc_media_base", |
| 288 "../media:rtc_media_tests_utils", | 284 "../media:rtc_media_tests_utils", |
| 289 "../p2p:p2p_test_utils", | 285 "../p2p:p2p_test_utils", |
| 290 "../p2p:rtc_p2p", | 286 "../p2p:rtc_p2p", |
| 287 "../rtc_base:rtc_base", |
| 288 "../rtc_base:rtc_base_approved", |
| 289 "../rtc_base:rtc_base_tests_main", |
| 290 "../rtc_base:rtc_base_tests_utils", |
| 291 "../system_wrappers:metrics_default", | 291 "../system_wrappers:metrics_default", |
| 292 ] | 292 ] |
| 293 | 293 |
| 294 if (rtc_build_libsrtp) { | 294 if (rtc_build_libsrtp) { |
| 295 deps += [ "//third_party/libsrtp" ] | 295 deps += [ "//third_party/libsrtp" ] |
| 296 } | 296 } |
| 297 | 297 |
| 298 if (is_android) { | 298 if (is_android) { |
| 299 deps += [ "//testing/android/native_test:native_test_support" ] | 299 deps += [ "//testing/android/native_test:native_test_support" ] |
| 300 } | 300 } |
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| 318 "test/peerconnectiontestwrapper.h", | 318 "test/peerconnectiontestwrapper.h", |
| 319 "test/rtcstatsobtainer.h", | 319 "test/rtcstatsobtainer.h", |
| 320 "test/testsdpstrings.h", | 320 "test/testsdpstrings.h", |
| 321 ] | 321 ] |
| 322 | 322 |
| 323 deps = [ | 323 deps = [ |
| 324 ":libjingle_peerconnection", | 324 ":libjingle_peerconnection", |
| 325 "..:webrtc_common", | 325 "..:webrtc_common", |
| 326 "../api:libjingle_peerconnection_test_api", | 326 "../api:libjingle_peerconnection_test_api", |
| 327 "../api:rtc_stats_api", | 327 "../api:rtc_stats_api", |
| 328 "../base:rtc_base", | |
| 329 "../base:rtc_base_approved", | |
| 330 "../base:rtc_base_tests_utils", | |
| 331 "../call:call_interfaces", | 328 "../call:call_interfaces", |
| 332 "../logging:rtc_event_log_api", | 329 "../logging:rtc_event_log_api", |
| 333 "../media:rtc_media", | 330 "../media:rtc_media", |
| 334 "../media:rtc_media_tests_utils", | 331 "../media:rtc_media_tests_utils", |
| 335 "../modules/audio_device:audio_device", | 332 "../modules/audio_device:audio_device", |
| 336 "../p2p:p2p_test_utils", | 333 "../p2p:p2p_test_utils", |
| 334 "../rtc_base:rtc_base", |
| 335 "../rtc_base:rtc_base_approved", |
| 336 "../rtc_base:rtc_base_tests_utils", |
| 337 "../test:test_support", | 337 "../test:test_support", |
| 338 "//testing/gmock", | 338 "//testing/gmock", |
| 339 ] | 339 ] |
| 340 | 340 |
| 341 if (!build_with_chromium && is_clang) { | 341 if (!build_with_chromium && is_clang) { |
| 342 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 342 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 343 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 343 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 344 } | 344 } |
| 345 } | 345 } |
| 346 | 346 |
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| 435 "../sdk/android:libjingle_peerconnection_jni", | 435 "../sdk/android:libjingle_peerconnection_jni", |
| 436 "//testing/android/native_test:native_test_support", | 436 "//testing/android/native_test:native_test_support", |
| 437 ] | 437 ] |
| 438 } | 438 } |
| 439 | 439 |
| 440 deps += [ | 440 deps += [ |
| 441 ":libjingle_peerconnection", | 441 ":libjingle_peerconnection", |
| 442 ":pc_test_utils", | 442 ":pc_test_utils", |
| 443 "..:webrtc_common", | 443 "..:webrtc_common", |
| 444 "../api:fakemetricsobserver", | 444 "../api:fakemetricsobserver", |
| 445 "../base:rtc_base_tests_main", | |
| 446 "../base:rtc_base_tests_utils", | |
| 447 "../media:rtc_media_tests_utils", | 445 "../media:rtc_media_tests_utils", |
| 448 "../pc:rtc_pc", | 446 "../pc:rtc_pc", |
| 447 "../rtc_base:rtc_base_tests_main", |
| 448 "../rtc_base:rtc_base_tests_utils", |
| 449 "../system_wrappers:metrics_default", | 449 "../system_wrappers:metrics_default", |
| 450 "../test:audio_codec_mocks", | 450 "../test:audio_codec_mocks", |
| 451 "//testing/gmock", | 451 "//testing/gmock", |
| 452 ] | 452 ] |
| 453 | 453 |
| 454 if (is_android) { | 454 if (is_android) { |
| 455 deps += [ "//testing/android/native_test:native_test_support" ] | 455 deps += [ "//testing/android/native_test:native_test_support" ] |
| 456 | 456 |
| 457 shard_timeout = 900 | 457 shard_timeout = 900 |
| 458 } | 458 } |
| 459 } | 459 } |
| 460 } | 460 } |
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