273d0296a4
This enables PeerConnection tests to use LoopbackMediaTransport to test data-channel-over-media-transport code. Also changes LoopbackMediaTransport to invoke callbacks asynchronously. This is more accurate, as these callbacks are triggered by network events. The caller should not block while the callback executes. Since LoopbackMediaTransport is used for testing, it provides a FlushAsyncInvokes() method which may be used to ensure that callbacks occur deterministically (eg. before checking that data has been received). Bug: webrtc:9719 Change-Id: Ib8ea9aebf4a0ad3d5934a6fe4ab33432c68523fd Tbr: stefan@webrtc.org Reviewed-on: https://webrtc-review.googlesource.com/c/109060 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25489}
648 lines
15 KiB
Plaintext
648 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("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("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/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_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("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: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_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",
|
|
"ortc/mediadescription_unittest.cc",
|
|
"ortc/sessiondescription_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",
|
|
":ortc_api",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../test:test_support",
|
|
"units:units_unittests",
|
|
]
|
|
}
|
|
}
|