| # 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", |
| "//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", |
| "../system_wrappers:metrics_default", |
| "../system_wrappers:runtime_enabled_features_default", |
| ] |
| |
| 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" ] |
| } |
| } |
| } |