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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("../build/webrtc.gni") | 9 import("../build/webrtc.gni") |
10 if (is_android) { | 10 if (is_android) { |
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77 "peerconnection.cc", | 77 "peerconnection.cc", |
78 "peerconnection.h", | 78 "peerconnection.h", |
79 "peerconnectionfactory.cc", | 79 "peerconnectionfactory.cc", |
80 "peerconnectionfactory.h", | 80 "peerconnectionfactory.h", |
81 "peerconnectionfactoryproxy.h", | 81 "peerconnectionfactoryproxy.h", |
82 "peerconnectioninterface.h", | 82 "peerconnectioninterface.h", |
83 "peerconnectionproxy.h", | 83 "peerconnectionproxy.h", |
84 "proxy.h", | 84 "proxy.h", |
85 "remoteaudiosource.cc", | 85 "remoteaudiosource.cc", |
86 "remoteaudiosource.h", | 86 "remoteaudiosource.h", |
87 "rtcstats.h", | 87 "rtcstatscollector.cc", |
88 "rtcstats_objects.h", | 88 "rtcstatscollector.h", |
89 "rtcstatsreport.h", | |
90 "rtpparameters.h", | 89 "rtpparameters.h", |
91 "rtpreceiver.cc", | 90 "rtpreceiver.cc", |
92 "rtpreceiver.h", | 91 "rtpreceiver.h", |
93 "rtpreceiverinterface.h", | 92 "rtpreceiverinterface.h", |
94 "rtpsender.cc", | 93 "rtpsender.cc", |
95 "rtpsender.h", | 94 "rtpsender.h", |
96 "rtpsenderinterface.h", | 95 "rtpsenderinterface.h", |
97 "sctputils.cc", | 96 "sctputils.cc", |
98 "sctputils.h", | 97 "sctputils.h", |
99 "statscollector.cc", | 98 "statscollector.cc", |
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118 | 117 |
119 configs += [ ":libjingle_peerconnection_warnings_config" ] | 118 configs += [ ":libjingle_peerconnection_warnings_config" ] |
120 | 119 |
121 if (is_clang) { | 120 if (is_clang) { |
122 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 121 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
123 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 122 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
124 } | 123 } |
125 | 124 |
126 deps = [ | 125 deps = [ |
127 ":call_api", | 126 ":call_api", |
| 127 ":rtc_stats_api", |
128 "../call", | 128 "../call", |
129 "../media", | 129 "../media", |
130 "../pc", | 130 "../pc", |
| 131 "../stats", |
131 ] | 132 ] |
132 | 133 |
133 if (rtc_use_quic) { | 134 if (rtc_use_quic) { |
134 sources += [ | 135 sources += [ |
135 "quicdatachannel.cc", | 136 "quicdatachannel.cc", |
136 "quicdatachannel.h", | 137 "quicdatachannel.h", |
137 "quicdatatransport.cc", | 138 "quicdatatransport.cc", |
138 "quicdatatransport.h", | 139 "quicdatatransport.h", |
139 ] | 140 ] |
140 deps += [ "//third_party/libquic" ] | 141 deps += [ "//third_party/libquic" ] |
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291 "android/java/src/org/webrtc/VideoSource.java", | 292 "android/java/src/org/webrtc/VideoSource.java", |
292 "android/java/src/org/webrtc/VideoTrack.java", | 293 "android/java/src/org/webrtc/VideoTrack.java", |
293 ] | 294 ] |
294 | 295 |
295 deps = [ | 296 deps = [ |
296 "//webrtc/base:base_java", | 297 "//webrtc/base:base_java", |
297 ] | 298 ] |
298 } | 299 } |
299 } | 300 } |
300 | 301 |
| 302 # GYP version: webrtc/api/api.gyp:rtc_stats_api |
| 303 rtc_source_set("rtc_stats_api") { |
| 304 cflags = [] |
| 305 sources = [ |
| 306 "stats/rtcstats.h", |
| 307 "stats/rtcstats_objects.h", |
| 308 "stats/rtcstatsreport.h", |
| 309 ] |
| 310 |
| 311 deps = [ |
| 312 "../base:rtc_base_approved", |
| 313 ] |
| 314 } |
| 315 |
301 if (rtc_include_tests) { | 316 if (rtc_include_tests) { |
302 config("peerconnection_unittests_config") { | 317 config("peerconnection_unittests_config") { |
303 # The warnings below are enabled by default. Since GN orders compiler flags | 318 # The warnings below are enabled by default. Since GN orders compiler flags |
304 # for a target before flags from configs, the only way to disable such | 319 # for a target before flags from configs, the only way to disable such |
305 # warnings is by having them in a separate config, loaded from the target. | 320 # warnings is by having them in a separate config, loaded from the target. |
306 # TODO(kjellander): Make the code compile without disabling these flags. | 321 # TODO(kjellander): Make the code compile without disabling these flags. |
307 # See https://bugs.webrtc.org/3307. | 322 # See https://bugs.webrtc.org/3307. |
308 if (is_clang && is_win) { | 323 if (is_clang && is_win) { |
309 cflags = [ | 324 cflags = [ |
310 # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6267 | 325 # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6267 |
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329 "fakemetricsobserver.h", | 344 "fakemetricsobserver.h", |
330 "jsepsessiondescription_unittest.cc", | 345 "jsepsessiondescription_unittest.cc", |
331 "localaudiosource_unittest.cc", | 346 "localaudiosource_unittest.cc", |
332 "mediaconstraintsinterface_unittest.cc", | 347 "mediaconstraintsinterface_unittest.cc", |
333 "mediastream_unittest.cc", | 348 "mediastream_unittest.cc", |
334 "peerconnection_unittest.cc", | 349 "peerconnection_unittest.cc", |
335 "peerconnectionendtoend_unittest.cc", | 350 "peerconnectionendtoend_unittest.cc", |
336 "peerconnectionfactory_unittest.cc", | 351 "peerconnectionfactory_unittest.cc", |
337 "peerconnectioninterface_unittest.cc", | 352 "peerconnectioninterface_unittest.cc", |
338 "proxy_unittest.cc", | 353 "proxy_unittest.cc", |
| 354 "rtcstatscollector_unittest.cc", |
339 "rtpsenderreceiver_unittest.cc", | 355 "rtpsenderreceiver_unittest.cc", |
340 "statscollector_unittest.cc", | 356 "statscollector_unittest.cc", |
341 "test/fakeaudiocapturemodule.cc", | 357 "test/fakeaudiocapturemodule.cc", |
342 "test/fakeaudiocapturemodule.h", | 358 "test/fakeaudiocapturemodule.h", |
343 "test/fakeaudiocapturemodule_unittest.cc", | 359 "test/fakeaudiocapturemodule_unittest.cc", |
344 "test/fakeconstraints.h", | 360 "test/fakeconstraints.h", |
345 "test/fakedatachannelprovider.h", | 361 "test/fakedatachannelprovider.h", |
346 "test/fakeperiodicvideocapturer.h", | 362 "test/fakeperiodicvideocapturer.h", |
347 "test/fakertccertificategenerator.h", | 363 "test/fakertccertificategenerator.h", |
348 "test/fakevideotrackrenderer.h", | 364 "test/fakevideotrackrenderer.h", |
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446 | 462 |
447 shared_libraries = [ ":libjingle_peerconnection_so" ] | 463 shared_libraries = [ ":libjingle_peerconnection_so" ] |
448 } | 464 } |
449 | 465 |
450 android_resources("libjingle_peerconnection_android_unittest_resources") { | 466 android_resources("libjingle_peerconnection_android_unittest_resources") { |
451 resource_dirs = [ "androidtests/res" ] | 467 resource_dirs = [ "androidtests/res" ] |
452 custom_package = "org.webrtc" | 468 custom_package = "org.webrtc" |
453 } | 469 } |
454 } | 470 } |
455 } | 471 } |
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