| # Copyright (c) 2015 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") |
| } |
| |
| group("api") { |
| public_deps = [ |
| ":libjingle_peerconnection_api", |
| ] |
| } |
| |
| rtc_source_set("call_api") { |
| sources = [ |
| "call/audio_sink.h", |
| ] |
| |
| deps = [ |
| # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done. |
| ":audio_mixer_api", |
| ":transport_api", |
| "..:webrtc_common", |
| "../rtc_base:rtc_base_approved", |
| "audio_codecs:audio_codecs_api", |
| ] |
| } |
| |
| rtc_static_library("libjingle_peerconnection_api") { |
| # Cannot have GN check enabled since that would introduce dependency cycles |
| # TODO(kjellander): Remove (bugs.webrtc.org/7504) |
| check_includes = false |
| cflags = [] |
| sources = [ |
| "datachannel.h", |
| "datachannelinterface.h", |
| "dtmfsenderinterface.h", |
| "jsep.h", |
| "jsepicecandidate.h", |
| "jsepsessiondescription.h", |
| "mediaconstraintsinterface.cc", |
| "mediaconstraintsinterface.h", |
| "mediastream.h", |
| "mediastreaminterface.cc", |
| "mediastreaminterface.h", |
| "mediastreamproxy.h", |
| "mediastreamtrack.h", |
| "mediastreamtrackproxy.h", |
| "mediatypes.cc", |
| "mediatypes.h", |
| "notifier.h", |
| "peerconnectionfactoryproxy.h", |
| "peerconnectioninterface.h", |
| "peerconnectionproxy.h", |
| "proxy.h", |
| "rtcerror.cc", |
| "rtcerror.h", |
| "rtpparameters.h", |
| "rtpreceiverinterface.h", |
| "rtpsender.h", |
| "rtpsenderinterface.h", |
| "statstypes.cc", |
| "statstypes.h", |
| "streamcollection.h", |
| "umametrics.h", |
| "videosourceproxy.h", |
| "videotracksource.h", |
| "webrtcsdp.h", |
| ] |
| |
| 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" ] |
| } |
| |
| deps = [ |
| ":rtc_stats_api", |
| "..:webrtc_common", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_base_approved", |
| "audio_codecs:audio_codecs_api", |
| ] |
| |
| # This is needed until bugs.webrtc.org/7504 is removed so this target can |
| # properly depend on ../media:rtc_media_base |
| # TODO(kjellander): Remove this dependency. |
| if (is_nacl) { |
| deps += [ "//native_client_sdk/src/libraries/nacl_io" ] |
| } |
| } |
| |
| rtc_source_set("ortc_api") { |
| check_includes = false # TODO(deadbeef): Remove (bugs.webrtc.org/6828) |
| sources = [ |
| "ortc/mediadescription.cc", |
| "ortc/mediadescription.h", |
| "ortc/ortcfactoryinterface.h", |
| "ortc/ortcrtpreceiverinterface.h", |
| "ortc/ortcrtpsenderinterface.h", |
| "ortc/packettransportinterface.h", |
| "ortc/rtptransportcontrollerinterface.h", |
| "ortc/rtptransportinterface.h", |
| "ortc/sessiondescription.cc", |
| "ortc/sessiondescription.h", |
| "ortc/srtptransportinterface.h", |
| "ortc/udptransportinterface.h", |
| ] |
| |
| # For mediastreaminterface.h, etc. |
| # TODO(deadbeef): Create a separate target for the common things ORTC and |
| # PeerConnection code shares, so that ortc_api can depend on that instead of |
| # libjingle_peerconnection_api. |
| public_deps = [ |
| ":libjingle_peerconnection_api", |
| ] |
| 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" ] |
| } |
| } |
| |
| # TODO(ossu): Remove once downstream projects have updated. |
| rtc_source_set("libjingle_peerconnection") { |
| public_deps = [ |
| "../pc:libjingle_peerconnection", |
| ] |
| } |
| |
| rtc_source_set("rtc_stats_api") { |
| cflags = [] |
| sources = [ |
| "stats/rtcstats.h", |
| "stats/rtcstats_objects.h", |
| "stats/rtcstatscollectorcallback.h", |
| "stats/rtcstatsreport.h", |
| ] |
| |
| deps = [ |
| "../rtc_base:rtc_base_approved", |
| ] |
| } |
| |
| rtc_source_set("audio_mixer_api") { |
| sources = [ |
| "audio/audio_mixer.h", |
| ] |
| |
| deps = [ |
| "../modules:module_api", |
| "../rtc_base:rtc_base_approved", |
| ] |
| } |
| |
| rtc_source_set("transport_api") { |
| sources = [ |
| "call/transport.h", |
| ] |
| } |
| |
| rtc_source_set("video_frame_api") { |
| sources = [ |
| "video/i420_buffer.cc", |
| "video/i420_buffer.h", |
| "video/video_frame.cc", |
| "video/video_frame.h", |
| "video/video_frame_buffer.cc", |
| "video/video_frame_buffer.h", |
| "video/video_rotation.h", |
| "video/video_timing.cc", |
| "video/video_timing.h", |
| ] |
| |
| deps = [ |
| "../rtc_base:rtc_base_approved", |
| "../system_wrappers", |
| ] |
| |
| # TODO(nisse): This logic is duplicated in multiple places. |
| # Define in a single place. |
| if (rtc_build_libyuv) { |
| deps += [ "$rtc_libyuv_dir" ] |
| public_deps = [ |
| "$rtc_libyuv_dir", |
| ] |
| } else { |
| # Need to add a directory normally exported by libyuv. |
| include_dirs = [ "$rtc_libyuv_dir/include" ] |
| } |
| } |
| |
| rtc_source_set("libjingle_peerconnection_test_api") { |
| testonly = true |
| sources = [ |
| "test/fakeconstraints.h", |
| ] |
| |
| public_deps = [ |
| ":libjingle_peerconnection_api", |
| ] |
| |
| deps = [ |
| "../rtc_base:rtc_base_approved", |
| ] |
| } |
| |
| if (rtc_include_tests) { |
| rtc_source_set("mock_audio_mixer") { |
| testonly = true |
| sources = [ |
| "test/mock_audio_mixer.h", |
| ] |
| |
| public_deps = [ |
| ":audio_mixer_api", |
| ] |
| |
| deps = [ |
| "../test:test_support", |
| "//testing/gmock", |
| ] |
| } |
| |
| rtc_source_set("fakemetricsobserver") { |
| testonly = true |
| sources = [ |
| "fakemetricsobserver.cc", |
| "fakemetricsobserver.h", |
| ] |
| deps = [ |
| ":libjingle_peerconnection_api", |
| "../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" ] |
| } |
| } |
| |
| rtc_source_set("rtc_api_unittests") { |
| testonly = true |
| |
| # Skip restricting visibility on mobile platforms since the tests on those |
| # gets additional generated targets which would require many lines here to |
| # cover (which would be confusing to read and hard to maintain). |
| if (!is_android && !is_ios) { |
| visibility = [ "..:rtc_unittests" ] |
| } |
| sources = [ |
| "ortc/mediadescription_unittest.cc", |
| "ortc/sessiondescription_unittest.cc", |
| "rtcerror_unittest.cc", |
| ] |
| |
| 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" ] |
| } |
| |
| deps = [ |
| ":libjingle_peerconnection_api", |
| ":ortc_api", |
| "../test:test_support", |
| ] |
| } |
| } |