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

Issue 2981013002: Introduce RtpTransportInternal and SrtpTransport. (Closed)
Patch Set: Created 3 years, 5 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 30 matching lines...) Expand all
41 "externalhmac.cc", 41 "externalhmac.cc",
42 "externalhmac.h", 42 "externalhmac.h",
43 "mediamonitor.cc", 43 "mediamonitor.cc",
44 "mediamonitor.h", 44 "mediamonitor.h",
45 "mediasession.cc", 45 "mediasession.cc",
46 "mediasession.h", 46 "mediasession.h",
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 "rtptransportinternal.h",
51 "srtpfilter.cc", 52 "srtpfilter.cc",
52 "srtpfilter.h", 53 "srtpfilter.h",
53 "srtpsession.cc", 54 "srtpsession.cc",
54 "srtpsession.h", 55 "srtpsession.h",
56 "srtptransport.cc",
57 "srtptransport.h",
55 "voicechannel.h", 58 "voicechannel.h",
56 ] 59 ]
57 60
58 deps = [ 61 deps = [
59 "..:webrtc_common", 62 "..:webrtc_common",
60 "../api:call_api", 63 "../api:call_api",
61 "../api:libjingle_peerconnection_api", 64 "../api:libjingle_peerconnection_api",
62 "../api:ortc_api", 65 "../api:ortc_api",
63 "../base:rtc_base", 66 "../base:rtc_base",
64 "../base:rtc_task_queue", 67 "../base:rtc_task_queue",
(...skipping 186 matching lines...) Expand 10 before | Expand all | Expand 10 after
251 testonly = true 254 testonly = true
252 255
253 sources = [ 256 sources = [
254 "bundlefilter_unittest.cc", 257 "bundlefilter_unittest.cc",
255 "channel_unittest.cc", 258 "channel_unittest.cc",
256 "channelmanager_unittest.cc", 259 "channelmanager_unittest.cc",
257 "currentspeakermonitor_unittest.cc", 260 "currentspeakermonitor_unittest.cc",
258 "mediasession_unittest.cc", 261 "mediasession_unittest.cc",
259 "rtcpmuxfilter_unittest.cc", 262 "rtcpmuxfilter_unittest.cc",
260 "rtptransport_unittest.cc", 263 "rtptransport_unittest.cc",
264 "rtptransporttestutil.h",
261 "srtpfilter_unittest.cc", 265 "srtpfilter_unittest.cc",
262 "srtpsession_unittest.cc", 266 "srtpsession_unittest.cc",
263 "srtptestutil.h", 267 "srtptestutil.h",
268 "srtptransport_unittest.cc",
264 ] 269 ]
265 270
266 include_dirs = [ "//third_party/libsrtp/srtp" ] 271 include_dirs = [ "//third_party/libsrtp/srtp" ]
267 272
268 configs += [ ":rtc_pc_unittests_config" ] 273 configs += [ ":rtc_pc_unittests_config" ]
269 274
270 if (!build_with_chromium && is_clang) { 275 if (!build_with_chromium && is_clang) {
271 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 276 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
272 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 277 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
273 } 278 }
(...skipping 177 matching lines...) Expand 10 before | Expand all | Expand 10 after
451 "//testing/gmock", 456 "//testing/gmock",
452 ] 457 ]
453 458
454 if (is_android) { 459 if (is_android) {
455 deps += [ "//testing/android/native_test:native_test_support" ] 460 deps += [ "//testing/android/native_test:native_test_support" ]
456 461
457 shard_timeout = 900 462 shard_timeout = 900
458 } 463 }
459 } 464 }
460 } 465 }
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