# 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", "rtpparametersconversion.cc", "rtpparametersconversion.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:optional", "../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", ] 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", "rtpparametersconversion_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" ] } } }