webrtc_m130/api/BUILD.gn
Sebastian Jansson 6736df1778 Moves BitrateAllocationUpdate to api.
This way it can be forwarded to lower layers. This makes it easier to
add information without having to change signatures of intermediate
classes. This will be used in a later CL to use the link capacity in the
Opus decoder.

Bug: webrtc:9718
Change-Id: I4a4c9d104fedb0e4a0bb7f14d169475940edbf7e
Reviewed-on: https://webrtc-review.googlesource.com/c/111508
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25738}
2018-11-21 19:59:55 +00:00

665 lines
15 KiB
Plaintext

# 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("create_peerconnection_factory") {
visibility = [ "*" ]
sources = [
"create_peerconnection_factory.cc",
"create_peerconnection_factory.h",
]
}
rtc_static_library("libjingle_peerconnection_api") {
visibility = [ "*" ]
cflags = []
sources = [
"asyncresolverfactory.h",
"bitrate_constraints.h",
"candidate.cc",
"candidate.h",
"crypto/cryptooptions.cc",
"crypto/cryptooptions.h",
"crypto/framedecryptorinterface.h",
"crypto/frameencryptorinterface.h",
"cryptoparams.h",
"datachannelinterface.cc",
"datachannelinterface.h",
"dtmfsenderinterface.h",
"jsep.cc",
"jsep.h",
"jsepicecandidate.cc",
"jsepicecandidate.h",
"jsepsessiondescription.h",
"media_transport_interface.cc",
"media_transport_interface.h",
"mediaconstraintsinterface.cc",
"mediaconstraintsinterface.h",
"mediastreaminterface.cc",
"mediastreaminterface.h",
"mediastreamproxy.h",
"mediastreamtrackproxy.h",
"mediatypes.cc",
"mediatypes.h",
"notifier.h",
"peerconnectionfactoryproxy.h",
"peerconnectioninterface.cc",
"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.cc",
"rtpsenderinterface.h",
"rtptransceiverinterface.cc",
"rtptransceiverinterface.h",
"setremotedescriptionobserverinterface.h",
"statstypes.cc",
"statstypes.h",
"turncustomizer.h",
"umametrics.h",
"videosourceproxy.h",
]
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:encoded_image",
"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.
"..: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",
"../rtc_base/system:rtc_export",
]
if (is_nacl) {
# This is needed by .h files included from rtc_base.
deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
}
}
rtc_source_set("scoped_refptr") {
visibility = [ "*" ]
sources = [
"scoped_refptr.h",
]
}
rtc_source_set("video_quality_test_fixture_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/video_quality_test_fixture.h",
]
deps = [
":fec_controller_api",
":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" ]
}
}
rtc_source_set("test_dependency_factory") {
visibility = [ "*" ]
testonly = true
sources = [
"test/test_dependency_factory.cc",
"test/test_dependency_factory.h",
]
deps = [
":video_quality_test_fixture_api",
"../rtc_base:thread_checker",
"//third_party/abseil-cpp/absl/memory",
]
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",
"//third_party/abseil-cpp/absl/memory",
]
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/packettransportinterface.h",
"ortc/rtptransportinterface.h",
"ortc/srtptransportinterface.h",
]
deps = [
":libjingle_peerconnection_api",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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_base/system:rtc_export",
]
}
rtc_source_set("audio_options_api") {
visibility = [ "*" ]
sources = [
"audio_options.cc",
"audio_options.h",
]
deps = [
"../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("transport_api") {
visibility = [ "*" ]
sources = [
"call/transport.cc",
"call/transport.h",
]
}
rtc_source_set("bitrate_allocation") {
visibility = [ "*" ]
sources = [
"call/bitrate_allocation.h",
]
deps = [
"units:data_rate",
"units:time_delta",
]
}
rtc_source_set("simulated_network_api") {
visibility = [ "*" ]
sources = [
"test/simulated_network.h",
]
deps = [
"../rtc_base:criticalsection",
"../rtc_base:rtc_base",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
}
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",
]
}
rtc_source_set("neteq_simulator_api") {
visibility = [ "*" ]
sources = [
"test/neteq_simulator.cc",
"test/neteq_simulator.h",
]
}
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:api",
"../modules/audio_processing:audio_processing",
"../modules/audio_processing:audioproc_f_impl",
]
}
rtc_source_set("neteq_simulator_factory") {
visibility = [ "*" ]
testonly = true
sources = [
"test/neteq_simulator_factory.cc",
"test/neteq_simulator_factory.h",
]
deps = [
":neteq_simulator_api",
"../modules/audio_coding:neteq_test_factory",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/memory",
]
}
}
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",
"//third_party/abseil-cpp/absl/memory",
]
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",
"../rtc_base:stringutils",
"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",
"//third_party/abseil-cpp/absl/memory",
]
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_frame_encryptor") {
testonly = true
sources = [
"test/mock_frame_encryptor.cc",
"test/mock_frame_encryptor.h",
]
deps = [
":libjingle_peerconnection_api",
"../test:test_support",
]
}
rtc_source_set("mock_frame_decryptor") {
testonly = true
sources = [
"test/mock_frame_decryptor.cc",
"test/mock_frame_decryptor.h",
]
deps = [
":libjingle_peerconnection_api",
"../test:test_support",
]
}
rtc_source_set("fake_frame_encryptor") {
testonly = true
sources = [
"test/fake_frame_encryptor.cc",
"test/fake_frame_encryptor.h",
]
deps = [
":array_view",
":libjingle_peerconnection_api",
"..:webrtc_common",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
]
}
rtc_source_set("fake_frame_decryptor") {
testonly = true
sources = [
"test/fake_frame_decryptor.cc",
"test/fake_frame_decryptor.h",
]
deps = [
":array_view",
":libjingle_peerconnection_api",
"..:webrtc_common",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
]
}
rtc_source_set("mock_peerconnectioninterface") {
testonly = true
sources = [
"test/mock_peerconnectioninterface.h",
]
deps = [
":libjingle_peerconnection_api",
"../test:test_support",
]
}
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_bitrate_allocator") {
testonly = true
sources = [
"test/mock_video_bitrate_allocator.h",
]
deps = [
"../api/video:video_bitrate_allocator",
"../test:test_support",
]
}
rtc_source_set("mock_video_bitrate_allocator_factory") {
testonly = true
sources = [
"test/mock_video_bitrate_allocator_factory.h",
]
deps = [
"../api/video:video_bitrate_allocator_factory",
"../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("mock_video_decoder") {
visibility = [ "*" ]
testonly = true
sources = [
"test/mock_video_decoder.cc",
"test/mock_video_decoder.h",
]
deps = [
"../api/video_codecs:video_codecs_api",
"../test:test_support",
]
}
rtc_source_set("mock_video_encoder") {
visibility = [ "*" ]
testonly = true
sources = [
"test/mock_video_encoder.cc",
"test/mock_video_encoder.h",
]
deps = [
"../api/video_codecs:video_codecs_api",
"../test:test_support",
]
}
rtc_source_set("fake_media_transport") {
testonly = true
sources = [
"test/fake_media_transport.h",
]
deps = [
":libjingle_peerconnection_api",
"../rtc_base:checks",
"//third_party/abseil-cpp/absl/memory:memory",
]
}
rtc_source_set("loopback_media_transport") {
testonly = true
sources = [
"test/loopback_media_transport.h",
]
deps = [
":libjingle_peerconnection_api",
"../rtc_base:checks",
"../rtc_base:rtc_base",
]
}
rtc_source_set("rtc_api_unittests") {
testonly = true
sources = [
"array_view_unittest.cc",
"rtcerror_unittest.cc",
"rtpparameters_unittest.cc",
"test/loopback_media_transport_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",
":loopback_media_transport",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../test:test_support",
"units:units_unittests",
]
}
}