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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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140 if (!build_with_chromium && is_clang) { | 140 if (!build_with_chromium && is_clang) { |
141 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 141 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
142 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 142 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
143 } | 143 } |
144 | 144 |
145 deps = [ | 145 deps = [ |
146 ":rtc_pc", | 146 ":rtc_pc", |
147 "../api:call_api", | 147 "../api:call_api", |
148 "../api:rtc_stats_api", | 148 "../api:rtc_stats_api", |
149 "../api/video_codecs:video_codecs_api", | 149 "../api/video_codecs:video_codecs_api", |
150 "../call", | |
151 "../media", | 150 "../media", |
152 "../stats", | 151 "../stats", |
153 ] | 152 ] |
154 | 153 |
154 if (rtc_enable_media) { | |
155 deps += [ "../call" ] | |
156 } | |
157 | |
155 public_deps = [ | 158 public_deps = [ |
156 "../api:libjingle_peerconnection_api", | 159 "../api:libjingle_peerconnection_api", |
157 ] | 160 ] |
158 | 161 |
159 if (rtc_use_quic) { | 162 if (rtc_use_quic) { |
160 sources += [ | 163 sources += [ |
161 "quicdatachannel.cc", | 164 "quicdatachannel.cc", |
162 "quicdatachannel.h", | 165 "quicdatachannel.h", |
163 "quicdatatransport.cc", | 166 "quicdatatransport.cc", |
164 "quicdatatransport.h", | 167 "quicdatatransport.h", |
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275 if (!is_win) { | 278 if (!is_win) { |
276 cflags = [ "-Wno-sign-compare" ] | 279 cflags = [ "-Wno-sign-compare" ] |
277 } | 280 } |
278 } | 281 } |
279 | 282 |
280 rtc_test("peerconnection_unittests") { | 283 rtc_test("peerconnection_unittests") { |
281 check_includes = false # TODO(kjellander): Remove (bugs.webrtc.org/6828) | 284 check_includes = false # TODO(kjellander): Remove (bugs.webrtc.org/6828) |
282 testonly = true | 285 testonly = true |
283 sources = [ | 286 sources = [ |
284 "datachannel_unittest.cc", | 287 "datachannel_unittest.cc", |
285 "dtmfsender_unittest.cc", | |
286 "fakemediacontroller.h", | |
287 "iceserverparsing_unittest.cc", | 288 "iceserverparsing_unittest.cc", |
288 "jsepsessiondescription_unittest.cc", | 289 "jsepsessiondescription_unittest.cc", |
289 "localaudiosource_unittest.cc", | |
290 "mediaconstraintsinterface_unittest.cc", | |
291 "mediastream_unittest.cc", | |
292 "peerconnection_integrationtest.cc", | 290 "peerconnection_integrationtest.cc", |
293 "peerconnectionendtoend_unittest.cc", | |
294 "peerconnectionfactory_unittest.cc", | |
295 "peerconnectioninterface_unittest.cc", | |
296 "proxy_unittest.cc", | |
297 "rtcstats_integrationtest.cc", | |
298 "rtcstatscollector_unittest.cc", | |
299 "rtpsenderreceiver_unittest.cc", | |
300 "sctputils_unittest.cc", | |
301 "statscollector_unittest.cc", | |
302 "test/fakeaudiocapturemodule_unittest.cc", | |
303 "test/testsdpstrings.h", | |
304 "trackmediainfomap_unittest.cc", | |
305 "videocapturertracksource_unittest.cc", | |
306 "videotrack_unittest.cc", | |
307 "webrtcsdp_unittest.cc", | |
308 "webrtcsession_unittest.cc", | |
309 ] | 291 ] |
310 | 292 |
293 if (rtc_enable_media) { | |
294 sources += [ | |
295 "dtmfsender_unittest.cc", | |
296 "fakemediacontroller.h", | |
297 "localaudiosource_unittest.cc", | |
298 "mediaconstraintsinterface_unittest.cc", | |
299 "mediastream_unittest.cc", | |
300 "peerconnectionendtoend_unittest.cc", | |
301 "peerconnectionfactory_unittest.cc", | |
302 "peerconnectioninterface_unittest.cc", | |
303 "proxy_unittest.cc", | |
304 "rtcstats_integrationtest.cc", | |
305 "rtcstatscollector_unittest.cc", | |
306 "rtpsenderreceiver_unittest.cc", | |
307 "sctputils_unittest.cc", | |
Taylor Brandstetter
2017/05/03 22:50:57
Does sctputils_unittest.cc need to be here?
Zhi Huang
2017/05/04 01:08:03
No, it doesn't. Actually, if we want to avoid the
| |
308 "statscollector_unittest.cc", | |
309 "test/fakeaudiocapturemodule_unittest.cc", | |
310 "test/testsdpstrings.h", | |
311 "trackmediainfomap_unittest.cc", | |
312 "videocapturertracksource_unittest.cc", | |
313 "videotrack_unittest.cc", | |
314 "webrtcsdp_unittest.cc", | |
315 "webrtcsession_unittest.cc", | |
316 ] | |
317 } | |
318 | |
311 if (rtc_enable_sctp) { | 319 if (rtc_enable_sctp) { |
312 defines = [ "HAVE_SCTP" ] | 320 defines = [ "HAVE_SCTP" ] |
313 } | 321 } |
314 | 322 |
315 configs += [ ":peerconnection_unittests_config" ] | 323 configs += [ ":peerconnection_unittests_config" ] |
316 | 324 |
317 if (!build_with_chromium && is_clang) { | 325 if (!build_with_chromium && is_clang) { |
318 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 326 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
319 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 327 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
320 } | 328 } |
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365 "//testing/gmock", | 373 "//testing/gmock", |
366 ] | 374 ] |
367 | 375 |
368 if (is_android) { | 376 if (is_android) { |
369 deps += [ "//testing/android/native_test:native_test_support" ] | 377 deps += [ "//testing/android/native_test:native_test_support" ] |
370 | 378 |
371 shard_timeout = 900 | 379 shard_timeout = 900 |
372 } | 380 } |
373 } | 381 } |
374 } | 382 } |
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