blob: 6ec5f23fd441e1d0c7711450de4785c669d30906 [file] [log] [blame]
# 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 = [
"bitrate_constraints.h",
"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",
":rtc_stats_api",
"audio:audio_mixer_api",
"audio_codecs:audio_codecs_api",
"transport:bitrate_settings",
"transport:network_control",
"video:video_frame",
"//third_party/abseil-cpp/absl/types:optional",
# 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("video_quality_test_fixture_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/video_quality_test_fixture.h",
]
deps = [
":libjingle_peerconnection_api",
":simulated_network_api",
"../call:fake_network",
"../call:rtp_interfaces",
"../test:test_common",
"../test:video_test_common",
"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" ]
}
}
if (rtc_include_tests) {
rtc_source_set("create_video_quality_test_fixture_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/create_video_quality_test_fixture.cc",
"test/create_video_quality_test_fixture.h",
]
deps = [
":fec_controller_api",
":video_quality_test_fixture_api",
"../rtc_base:ptr_util",
"../video:video_quality_test",
]
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("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",
"..:webrtc_common",
"../rtc_base:rtc_base",
"//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" ]
}
}
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 = [
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("transport_api") {
visibility = [ "*" ]
sources = [
"call/transport.cc",
"call/transport.h",
]
}
rtc_source_set("simulated_network_api") {
visibility = [ "*" ]
sources = [
"test/simulated_network.h",
]
deps = [
":optional",
"../rtc_base:criticalsection",
"../rtc_base:rtc_base",
]
}
rtc_source_set("fec_controller_api") {
visibility = [ "*" ]
sources = [
"fec_controller.h",
]
deps = [
"..:webrtc_common",
"../modules:module_fec_api",
]
}
rtc_source_set("array_view") {
visibility = [ "*" ]
sources = [
"array_view.h",
]
deps = [
"../rtc_base:checks",
"../rtc_base:type_traits",
]
}
# TODO(bugs.webrtc.org/9078): Deprecated, replaced by absl/types:optional.
# Delete after webrtc and downstreams users are updated.
rtc_source_set("optional") {
visibility = [ "*" ]
sources = [
"optional.h",
]
deps = [
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
}
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("simulcast_test_fixture_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/simulcast_test_fixture.h",
]
}
rtc_source_set("create_simulcast_test_fixture_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/create_simulcast_test_fixture.cc",
"test/create_simulcast_test_fixture.h",
]
deps = [
":simulcast_test_fixture_api",
"../modules/video_coding:simulcast_test_fixture_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("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",
# TODO(bugs.webrtc.org/8821): Remove optional_unittests when webrtc starts
# running absl unittest on each commit.
"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",
]
}
}