|  | # Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | 
|  | # | 
|  | # Use of this source code is governed by a BSD-style license | 
|  | # that can be found in the LICENSE file in the root of the source | 
|  | # tree. An additional intellectual property rights grant can be found | 
|  | # in the file PATENTS.  All contributing project authors may | 
|  | # be found in the AUTHORS file in the root of the source tree. | 
|  |  | 
|  | import("../webrtc.gni") | 
|  | if (is_android) { | 
|  | import("//build/config/android/config.gni") | 
|  | import("//build/config/android/rules.gni") | 
|  | } | 
|  |  | 
|  | rtc_static_library("ortc") { | 
|  | defines = [] | 
|  | sources = [ | 
|  | "ortcfactory.cc", | 
|  | "ortcfactory.h", | 
|  | "ortcrtpreceiveradapter.cc", | 
|  | "ortcrtpreceiveradapter.h", | 
|  | "ortcrtpsenderadapter.cc", | 
|  | "ortcrtpsenderadapter.h", | 
|  | "rtptransportadapter.cc", | 
|  | "rtptransportadapter.h", | 
|  | "rtptransportcontrolleradapter.cc", | 
|  | "rtptransportcontrolleradapter.h", | 
|  | ] | 
|  |  | 
|  | # TODO(deadbeef): Create a separate target for the common things ORTC and | 
|  | # PeerConnection code shares, so that ortc can depend on that instead of | 
|  | # libjingle_peerconnection. | 
|  | deps = [ | 
|  | "../api:libjingle_peerconnection_api", | 
|  | "../api:ortc_api", | 
|  | "../api/video_codecs:builtin_video_decoder_factory", | 
|  | "../api/video_codecs:builtin_video_encoder_factory", | 
|  | "../call:call_interfaces", | 
|  | "../call:rtp_sender", | 
|  | "../logging:rtc_event_log_api", | 
|  | "../logging:rtc_event_log_impl_base", | 
|  | "../media:rtc_audio_video", | 
|  | "../media:rtc_media", | 
|  | "../media:rtc_media_base", | 
|  | "../modules/audio_processing:audio_processing", | 
|  | "../p2p:rtc_p2p", | 
|  | "../pc:libjingle_peerconnection", | 
|  | "../pc:peerconnection", | 
|  | "../pc:rtc_pc", | 
|  | "../pc:rtc_pc_base", | 
|  | "../rtc_base:checks", | 
|  | "../rtc_base:rtc_base", | 
|  | "../rtc_base:rtc_base_approved", | 
|  | "../rtc_base/third_party/sigslot", | 
|  | "//third_party/abseil-cpp/absl/memory", | 
|  | "//third_party/abseil-cpp/absl/types:optional", | 
|  | ] | 
|  |  | 
|  | if (!build_with_chromium && is_clang) { | 
|  | # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 
|  | suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 
|  | } | 
|  | } | 
|  |  | 
|  | if (rtc_include_tests) { | 
|  | rtc_test("ortc_unittests") { | 
|  | testonly = true | 
|  |  | 
|  | sources = [ | 
|  | "ortcfactory_integrationtest.cc", | 
|  | "ortcfactory_unittest.cc", | 
|  | "ortcrtpreceiver_unittest.cc", | 
|  | "ortcrtpsender_unittest.cc", | 
|  | "rtptransport_unittest.cc", | 
|  | "rtptransportcontroller_unittest.cc", | 
|  | "srtptransport_unittest.cc", | 
|  | "testrtpparameters.cc", | 
|  | "testrtpparameters.h", | 
|  | ] | 
|  |  | 
|  | deps = [ | 
|  | ":ortc", | 
|  | "../api:libjingle_peerconnection_api", | 
|  | "../api:ortc_api", | 
|  | "../api/audio_codecs:builtin_audio_decoder_factory", | 
|  | "../api/audio_codecs:builtin_audio_encoder_factory", | 
|  | "../media:rtc_media_tests_utils", | 
|  | "../p2p:p2p_test_utils", | 
|  | "../p2p:rtc_p2p", | 
|  | "../pc:pc_test_utils", | 
|  | "../pc:peerconnection", | 
|  | "../rtc_base:rtc_base", | 
|  | "../rtc_base:rtc_base_approved", | 
|  | "../rtc_base:rtc_base_tests_main", | 
|  | "../rtc_base:rtc_base_tests_utils", | 
|  | "../rtc_base/system:arch", | 
|  | ] | 
|  |  | 
|  | if (!build_with_chromium && is_clang) { | 
|  | # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 
|  | suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 
|  | } | 
|  |  | 
|  | if (is_android) { | 
|  | deps += [ "//testing/android/native_test:native_test_support" ] | 
|  | } | 
|  | } | 
|  | } |