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Side by Side Diff: webrtc/pc/BUILD.gn

Issue 2854123003: Build WebRTC with data channel only. (Closed)
Patch Set: Merge. Created 3 years, 7 months ago
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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) {
(...skipping 129 matching lines...) Expand 10 before | Expand all | Expand 10 after
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",
(...skipping 110 matching lines...) Expand 10 before | Expand all | Expand 10 after
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 }
(...skipping 44 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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