| # 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") { |
| visibility = [ "*" ] |
| deps = [] |
| |
| if (!build_with_mozilla) { |
| deps += [ ":libjingle_peerconnection_api" ] |
| } |
| } |
| |
| rtc_source_set("call_api") { |
| visibility = [ "*" ] |
| sources = [ |
| "call/audio_sink.h", |
| ] |
| |
| deps = [ |
| # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done. |
| ":transport_api", |
| "..:webrtc_common", |
| "../rtc_base:rtc_base_approved", |
| "audio:audio_mixer_api", |
| "audio_codecs:audio_codecs_api", |
| ] |
| } |
| |
| rtc_source_set("callfactory_api") { |
| visibility = [ "*" ] |
| sources = [ |
| "call/callfactoryinterface.h", |
| ] |
| } |
| |
| rtc_static_library("libjingle_peerconnection_api") { |
| visibility = [ "*" ] |
| cflags = [] |
| sources = [ |
| "candidate.cc", |
| "candidate.h", |
| "cryptoparams.h", |
| "datachannelinterface.h", |
| "dtmfsenderinterface.h", |
| "jsep.cc", |
| "jsep.h", |
| "jsepicecandidate.h", |
| "jsepsessiondescription.h", |
| "mediaconstraintsinterface.cc", |
| "mediaconstraintsinterface.h", |
| "mediastreaminterface.cc", |
| "mediastreaminterface.h", |
| "mediastreamproxy.h", |
| "mediastreamtrackproxy.h", |
| "mediatypes.cc", |
| "mediatypes.h", |
| "notifier.h", |
| "peerconnectionfactoryproxy.h", |
| "peerconnectioninterface.h", |
| "peerconnectionproxy.h", |
| "proxy.cc", |
| "proxy.h", |
| "rtcerror.cc", |
| "rtcerror.h", |
| "rtp_headers.cc", |
| "rtp_headers.h", |
| "rtpparameters.cc", |
| "rtpparameters.h", |
| "rtpreceiverinterface.cc", |
| "rtpreceiverinterface.h", |
| "rtpsenderinterface.h", |
| "rtptransceiverinterface.h", |
| "setremotedescriptionobserverinterface.h", |
| "statstypes.cc", |
| "statstypes.h", |
| "turncustomizer.h", |
| "umametrics.cc", |
| "umametrics.h", |
| "videosourceproxy.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 = [ |
| ":array_view", |
| ":audio_options_api", |
| ":callfactory_api", |
| ":fec_controller_api", |
| ":libjingle_logging_api", |
| ":optional", |
| ":rtc_stats_api", |
| "audio:audio_mixer_api", |
| "audio_codecs:audio_codecs_api", |
| "transport:bitrate_settings", |
| "transport:network_control", |
| "video:video_frame", |
| |
| # Basically, don't add stuff here. You might break sensitive downstream |
| # targets like pnacl. API should not depend on anything outside of this |
| # file, really. All these should arguably go away in time. |
| "..:typedefs", |
| "..:webrtc_common", |
| "../logging:rtc_event_log_api", |
| "../media:rtc_media_config", |
| "../modules/audio_processing:audio_processing_statistics", |
| "../rtc_base:checks", |
| "../rtc_base:deprecation", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:stringutils", |
| ] |
| |
| if (is_nacl) { |
| # This is needed by .h files included from rtc_base. |
| deps += [ "//native_client_sdk/src/libraries/nacl_io" ] |
| } |
| } |
| |
| rtc_source_set("libjingle_logging_api") { |
| visibility = [ "*" ] |
| sources = [ |
| "rtceventlogoutput.h", |
| ] |
| } |
| |
| rtc_source_set("ortc_api") { |
| visibility = [ "*" ] |
| 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. |
| deps = [ |
| ":libjingle_peerconnection_api", |
| ":optional", |
| "..:webrtc_common", |
| "../rtc_base:rtc_base", |
| ] |
| 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_stats_api") { |
| visibility = [ "*" ] |
| cflags = [] |
| sources = [ |
| "stats/rtcstats.h", |
| "stats/rtcstats_objects.h", |
| "stats/rtcstatscollectorcallback.h", |
| "stats/rtcstatsreport.h", |
| ] |
| |
| deps = [ |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| ] |
| } |
| |
| rtc_source_set("audio_options_api") { |
| visibility = [ "*" ] |
| sources = [ |
| "audio_options.cc", |
| "audio_options.h", |
| ] |
| |
| deps = [ |
| ":optional", |
| "../rtc_base:rtc_base_approved", |
| ] |
| } |
| |
| rtc_source_set("transport_api") { |
| visibility = [ "*" ] |
| sources = [ |
| "call/transport.cc", |
| "call/transport.h", |
| ] |
| } |
| |
| rtc_source_set("fec_controller_api") { |
| visibility = [ "*" ] |
| sources = [ |
| "fec_controller.h", |
| ] |
| |
| deps = [ |
| "..:webrtc_common", |
| "../modules:module_fec_api", |
| ] |
| } |
| |
| # TODO(bugs.webrtc.org/9253): Deprecated, replaced by video:video_frame. |
| # Delete after downstream users are updated. |
| rtc_source_set("video_frame_api") { |
| visibility = [ "*" ] |
| sources = [ |
| "videosinkinterface.h", |
| "videosourceinterface.h", |
| ] |
| |
| public_deps = [ # no-presubmit-check TODO(webrtc:8603) |
| "video:video_frame", |
| ] |
| } |
| |
| # TODO(bugs.webrtc.org/9253): Deprecated, replaced by video:encoded_frame. |
| # Delete after downstream users are updated. |
| rtc_source_set("encoded_frame_api") { |
| visibility = [ "*" ] |
| |
| public_deps = [ # no-presubmit-check TODO(webrtc:8603) |
| "video:encoded_frame", |
| ] |
| } |
| |
| # TODO(bugs.webrtc.org/9253): Deprecated, replaced by video:video_stream_decoder. |
| # Delete after downstream users are updated. |
| rtc_source_set("video_stream_decoder") { |
| visibility = [ "*" ] |
| |
| public_deps = [ # no-presubmit-check TODO(webrtc:8603) |
| "video:video_stream_decoder", |
| ] |
| } |
| |
| # TODO(bugs.webrtc.org/9253): Deprecated, replaced by video:video_stream_decoder_create. |
| # Delete after downstream users are updated. |
| rtc_source_set("video_stream_decoder_create") { |
| visibility = [ "*" ] |
| allow_poison = [ "software_video_codecs" ] # TODO(bugs.webrtc.org/7925): Remove. |
| |
| public_deps = [ # no-presubmit-check TODO(webrtc:8603) |
| "video:video_stream_decoder_create", |
| ] |
| } |
| |
| # TODO(bugs.webrtc.org/9253): Deprecated, replaced by video:video_frame_i420. |
| # Delete after downstream users are updated. |
| rtc_source_set("video_frame_api_i420") { |
| visibility = [ "*" ] |
| public_deps = [ # no-presubmit-check TODO(webrtc:8603) |
| "video:video_frame_i420", |
| ] |
| } |
| |
| rtc_source_set("array_view") { |
| visibility = [ "*" ] |
| sources = [ |
| "array_view.h", |
| ] |
| deps = [ |
| "../rtc_base:checks", |
| "../rtc_base:type_traits", |
| ] |
| } |
| |
| rtc_source_set("optional") { |
| visibility = [ "*" ] |
| sources = [ |
| "optional.cc", |
| "optional.h", |
| ] |
| deps = [ |
| ":array_view", |
| "../rtc_base:checks", |
| "../rtc_base:sanitizer", |
| ] |
| } |
| |
| rtc_source_set("refcountedbase") { |
| visibility = [ "*" ] |
| sources = [ |
| "refcountedbase.h", |
| ] |
| deps = [ |
| "../rtc_base:rtc_base_approved", |
| ] |
| } |
| |
| rtc_source_set("libjingle_peerconnection_test_api") { |
| visibility = [ "*" ] |
| testonly = true |
| sources = [ |
| "test/fakeconstraints.h", |
| ] |
| |
| deps = [ |
| ":libjingle_peerconnection_api", |
| "../rtc_base:rtc_base_approved", |
| ] |
| } |
| |
| # TODO(bugs.webrtc.org/9253): Deprecated, replaced by video:video_bitrate_allocation. |
| # Delete after downstream users are updated. |
| rtc_source_set("video_bitrate_allocation") { |
| visibility = [ "*" ] |
| public_deps = [ # no-presubmit-check TODO(webrtc:8603) |
| "video:video_bitrate_allocation", |
| ] |
| } |
| |
| if (rtc_include_tests) { |
| if (rtc_enable_protobuf) { |
| rtc_source_set("audioproc_f_api") { |
| visibility = [ "*" ] |
| testonly = true |
| sources = [ |
| "test/audioproc_float.cc", |
| "test/audioproc_float.h", |
| ] |
| |
| deps = [ |
| "../modules/audio_processing:audio_processing", |
| "../modules/audio_processing:audioproc_f_impl", |
| ] |
| } |
| } |
| |
| rtc_source_set("videocodec_test_fixture_api") { |
| visibility = [ "*" ] |
| testonly = true |
| sources = [ |
| "test/videocodec_test_fixture.h", |
| "test/videocodec_test_stats.cc", |
| "test/videocodec_test_stats.h", |
| ] |
| deps = [ |
| "..:webrtc_common", |
| "../modules/video_coding:video_codec_interface", |
| "video_codecs:video_codecs_api", |
| ] |
| } |
| |
| rtc_source_set("create_videocodec_test_fixture_api") { |
| visibility = [ "*" ] |
| testonly = true |
| sources = [ |
| "test/create_videocodec_test_fixture.cc", |
| "test/create_videocodec_test_fixture.h", |
| ] |
| deps = [ |
| ":videocodec_test_fixture_api", |
| "../modules/video_coding:video_codecs_test_framework", |
| "../modules/video_coding:videocodec_test_impl", |
| "../rtc_base:rtc_base_approved", |
| "video_codecs:video_codecs_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" ] |
| } |
| } |
| |
| rtc_source_set("mock_audio_mixer") { |
| testonly = true |
| sources = [ |
| "test/mock_audio_mixer.h", |
| ] |
| |
| deps = [ |
| "../test:test_support", |
| "audio:audio_mixer_api", |
| ] |
| } |
| |
| rtc_source_set("mock_rtp") { |
| testonly = true |
| sources = [ |
| "test/mock_rtpreceiver.h", |
| "test/mock_rtpsender.h", |
| ] |
| |
| deps = [ |
| ":libjingle_peerconnection_api", |
| "../test:test_support", |
| ] |
| } |
| |
| rtc_source_set("mock_video_codec_factory") { |
| testonly = true |
| sources = [ |
| "test/mock_video_decoder_factory.h", |
| "test/mock_video_encoder_factory.h", |
| ] |
| |
| deps = [ |
| "../api/video_codecs:video_codecs_api", |
| "../test:test_support", |
| ] |
| } |
| |
| rtc_source_set("fakemetricsobserver") { |
| testonly = true |
| sources = [ |
| "fakemetricsobserver.cc", |
| "fakemetricsobserver.h", |
| ] |
| deps = [ |
| "../media:rtc_media_base", |
| "../rtc_base:checks", |
| "../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 (!build_with_mozilla) { |
| deps += [ ":libjingle_peerconnection_api" ] |
| } |
| } |
| |
| rtc_source_set("rtc_api_unittests") { |
| testonly = true |
| |
| sources = [ |
| "array_view_unittest.cc", |
| "optional_unittest.cc", |
| "ortc/mediadescription_unittest.cc", |
| "ortc/sessiondescription_unittest.cc", |
| "rtcerror_unittest.cc", |
| "rtpparameters_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 = [ |
| ":array_view", |
| ":libjingle_peerconnection_api", |
| ":libjingle_peerconnection_test_api", |
| ":optional", |
| ":ortc_api", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_base_tests_utils", |
| "../test:test_support", |
| "units:units_unittests", |
| ] |
| } |
| } |