blob: 94dd86507fe4ecefb83f1ca082a093848e8e0e2d [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") {
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") {
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",
"mediastreamproxy.h",
"mediastreamtrackproxy.h",
"mediatypes.cc",
"mediatypes.h",
"notifier.h",
"peerconnectionfactoryproxy.h",
"peerconnectionproxy.h",
"proxy.cc",
"proxy.h",
"rtcerror.cc",
"rtcerror.h",
"rtp_headers.cc",
"rtp_headers.h",
"rtpparameters.cc",
"rtpparameters.h",
"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" ]
}
public_deps = [
":libjingle_api_deprecated_headers",
":peerconnection_and_implicit_call_api",
]
deps = [
":array_view",
":optional",
":rtc_stats_api",
":video_frame_api",
"../rtc_base:checks",
"audio_codecs:audio_codecs_api",
# 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",
"../modules/audio_processing:audio_processing_statistics",
"../rtc_base:checks",
"../rtc_base:deprecation",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
]
# 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("peerconnection_and_implicit_call_api") {
# The peerconnectioninterface.h file pulls in call/callfactoryinterface.h
# and the entire call module with it. We need to either get rid of this
# dependency or pull most of call/ into the API. For now, silence the warnings
# this creates since it creates a circular dependency (call very much depends
# on API). See bugs.webrtc.org/7504.
check_includes = false
sources = [
"peerconnectioninterface.h",
]
}
rtc_source_set("libjingle_api_deprecated_headers") {
# We need to include headers from undeclared targets here, since they cause
# circular dependencies. These deprecated headers are going away anyway.
# See http://bugs.webrtc.org/5883.
check_includes = false
sources = [
"mediastreamtrack.h",
"videotracksource.h",
"webrtcsdp.h",
]
}
rtc_source_set("libjingle_logging_api") {
sources = [
"rtceventlogoutput.h",
]
}
rtc_source_set("ortc_api") {
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" ]
}
}
# 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:checks",
"../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/video_content_type.cc",
"video/video_content_type.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:checks",
"../rtc_base:rtc_base_approved",
]
}
rtc_source_set("video_frame_api_i420") {
sources = [
"video/i420_buffer.cc",
"video/i420_buffer.h",
]
deps = [
":video_frame_api",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"//third_party/libyuv",
]
}
rtc_source_set("array_view") {
sources = [
"array_view.h",
]
deps = [
"../rtc_base:checks",
"../rtc_base:type_traits",
]
}
rtc_source_set("optional") {
sources = [
"optional.cc",
"optional.h",
]
deps = [
":array_view",
"../rtc_base:checks",
"../rtc_base:sanitizer",
]
}
rtc_source_set("refcountedbase") {
sources = [
"refcountedbase.h",
]
deps = [
"../rtc_base:rtc_base_approved",
]
}
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("mock_video_codec_factory") {
testonly = true
sources = [
"test/mock_video_decoder_factory.h",
"test/mock_video_encoder_factory.h",
]
public_deps = [
"../api/video_codecs:video_codecs_api",
]
deps = [
"../test:test_support",
"//testing/gmock",
]
}
rtc_source_set("fakemetricsobserver") {
testonly = true
sources = [
"fakemetricsobserver.cc",
"fakemetricsobserver.h",
]
deps = [
":libjingle_peerconnection_api",
"../api:peerconnection_and_implicit_call_api",
"../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" ]
}
}
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",
]
}
}