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platform-external-webrtc/call/BUILD.gn
Artem Titov 46c4e60939 Introduce SimulatedNetworkReceiverInterface. 
Introduce SimulatedNetworkReceiverInterface and switch DirectTransport
on this interface. Also switch part of related users on
DefaultNetworkSimulationConfig.

This two changes united into single CL to prevent work duplication.
Most changes were done because of stop including fake_network_pipe.h
into direct_transport.h, so splitting this into 2 CLs will require
first fix all imports of fake_network_pipe.h and then replace them
on new API imports again.

Bug: webrtc:9630
Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6
Reviewed-on: https://webrtc-review.googlesource.com/94762
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24336}
2018-08-20 07:23:41 +00:00

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# 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")
rtc_source_set("call_interfaces") {
sources = [
"audio_receive_stream.cc",
"audio_receive_stream.h",
"audio_send_stream.h",
"audio_state.cc",
"audio_state.h",
"call.h",
"call_config.cc",
"call_config.h",
"flexfec_receive_stream.cc",
"flexfec_receive_stream.h",
"packet_receiver.h",
"syncable.cc",
"syncable.h",
]
if (!build_with_mozilla) {
sources += [ "audio_send_stream.cc" ]
}
deps = [
":rtp_interfaces",
":video_stream_api",
"..:webrtc_common",
"../api:fec_controller_api",
"../api:libjingle_peerconnection_api",
"../api:transport_api",
"../api/audio:audio_mixer_api",
"../api/audio_codecs:audio_codecs_api",
"../api/transport:network_control",
"../modules/audio_device:audio_device",
"../modules/audio_processing:audio_processing",
"../modules/audio_processing:audio_processing_statistics",
"../rtc_base:audio_format_to_string",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/types:optional",
]
}
# TODO(nisse): These RTP targets should be moved elsewhere
# when interfaces have stabilized. See also TODO for |mock_rtp_interfaces|.
rtc_source_set("rtp_interfaces") {
sources = [
"rtcp_packet_sink_interface.h",
"rtp_config.cc",
"rtp_config.h",
"rtp_packet_sink_interface.h",
"rtp_stream_receiver_controller_interface.h",
"rtp_transport_controller_send_interface.h",
]
deps = [
"../api:array_view",
"../api:libjingle_peerconnection_api",
"../api/transport:bitrate_settings",
"../logging:rtc_event_log_api",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("rtp_receiver") {
visibility = [ "*" ]
sources = [
"rtcp_demuxer.cc",
"rtcp_demuxer.h",
"rtp_demuxer.cc",
"rtp_demuxer.h",
"rtp_rtcp_demuxer_helper.cc",
"rtp_rtcp_demuxer_helper.h",
"rtp_stream_receiver_controller.cc",
"rtp_stream_receiver_controller.h",
"rtx_receive_stream.cc",
"rtx_receive_stream.h",
"ssrc_binding_observer.h",
]
deps = [
":rtp_interfaces",
"..:webrtc_common",
"../api:array_view",
"../api:libjingle_peerconnection_api",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("rtp_sender") {
sources = [
"rtp_payload_params.cc",
"rtp_payload_params.h",
"rtp_transport_controller_send.cc",
"rtp_transport_controller_send.h",
"rtp_video_sender.cc",
"rtp_video_sender.h",
"rtp_video_sender_interface.h",
]
deps = [
":bitrate_configurator",
":rtp_interfaces",
"..:webrtc_common",
"../api:transport_api",
"../api/transport:network_control",
"../api/video_codecs:video_codecs_api",
"../logging:rtc_event_log_api",
"../modules/congestion_controller",
"../modules/congestion_controller/rtp:congestion_controller",
"../modules/pacing",
"../modules/rtp_rtcp:rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/rtp_rtcp:rtp_video_header",
"../modules/utility",
"../modules/video_coding:video_codec_interface",
"../rtc_base:checks",
"../rtc_base:rate_limiter",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../system_wrappers:field_trial_api",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("bitrate_configurator") {
sources = [
"rtp_bitrate_configurator.cc",
"rtp_bitrate_configurator.h",
]
deps = [
":rtp_interfaces",
"../api:libjingle_peerconnection_api",
"../api/transport:bitrate_settings",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
]
}
rtc_source_set("bitrate_allocator") {
sources = [
"bitrate_allocator.cc",
"bitrate_allocator.h",
]
deps = [
"../modules/bitrate_controller",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:sequenced_task_checker",
"../system_wrappers",
"../system_wrappers:field_trial_api",
"../system_wrappers:metrics_api",
]
}
rtc_static_library("call") {
sources = [
"call.cc",
"callfactory.cc",
"callfactory.h",
"degraded_call.cc",
"degraded_call.h",
"flexfec_receive_stream_impl.cc",
"flexfec_receive_stream_impl.h",
"receive_time_calculator.cc",
"receive_time_calculator.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 = [
":bitrate_allocator",
":call_interfaces",
":fake_network",
":rtp_interfaces",
":rtp_receiver",
":rtp_sender",
":simulated_network",
":video_stream_api",
"..:webrtc_common",
"../api:callfactory_api",
"../api:simulated_network_api",
"../api:transport_api",
"../api/transport:network_control",
"../audio",
"../logging:rtc_event_audio",
"../logging:rtc_event_log_api",
"../logging:rtc_event_rtp_rtcp",
"../logging:rtc_event_video",
"../logging:rtc_stream_config",
"../modules/bitrate_controller",
"../modules/congestion_controller",
"../modules/pacing",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/utility",
"../modules/video_coding:video_coding",
"../rtc_base:checks",
"../rtc_base:rate_limiter",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../rtc_base:safe_minmax",
"../rtc_base:sequenced_task_checker",
"../rtc_base/synchronization:rw_lock_wrapper",
"../system_wrappers",
"../system_wrappers:field_trial_api",
"../system_wrappers:metrics_api",
"../video",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("video_stream_api") {
sources = [
"video_receive_stream.cc",
"video_receive_stream.h",
"video_send_stream.cc",
"video_send_stream.h",
]
deps = [
":rtp_interfaces",
"../:webrtc_common",
"../api:libjingle_peerconnection_api",
"../api:transport_api",
"../api/video:video_frame",
"../api/video:video_stream_encoder",
"../api/video_codecs:video_codecs_api",
"../common_video:common_video",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("simulated_network") {
sources = [
"simulated_network.cc",
"simulated_network.h",
]
deps = [
"../api:simulated_network_api",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("simulated_packet_receiver") {
sources = [
"simulated_packet_receiver.h",
]
deps = [
":call_interfaces",
"../api:simulated_network_api",
"../modules:module_api",
]
}
rtc_source_set("fake_network") {
sources = [
"fake_network_pipe.cc",
"fake_network_pipe.h",
]
deps = [
":call_interfaces",
":simulated_network",
":simulated_packet_receiver",
"..:webrtc_common",
"../api:simulated_network_api",
"../api:transport_api",
"../modules:module_api",
"../rtc_base:rtc_base_approved",
"../rtc_base:sequenced_task_checker",
"../system_wrappers",
"//third_party/abseil-cpp/absl/memory",
]
}
if (rtc_include_tests) {
rtc_source_set("call_tests") {
testonly = true
sources = [
"bitrate_allocator_unittest.cc",
"bitrate_estimator_tests.cc",
"call_unittest.cc",
"flexfec_receive_stream_unittest.cc",
"receive_time_calculator_unittest.cc",
"rtcp_demuxer_unittest.cc",
"rtp_bitrate_configurator_unittest.cc",
"rtp_demuxer_unittest.cc",
"rtp_payload_params_unittest.cc",
"rtp_rtcp_demuxer_helper_unittest.cc",
"rtp_video_sender_unittest.cc",
"rtx_receive_stream_unittest.cc",
]
deps = [
":bitrate_allocator",
":bitrate_configurator",
":call",
":call_interfaces",
":mock_rtp_interfaces",
":rtp_interfaces",
":rtp_receiver",
":rtp_sender",
"..:webrtc_common",
"../api:array_view",
"../api:libjingle_peerconnection_api",
"../api:mock_audio_mixer",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../audio:audio",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_base",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/audio_processing:mocks",
"../modules/bitrate_controller",
"../modules/congestion_controller",
"../modules/pacing",
"../modules/pacing:mock_paced_sender",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:mock_rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/utility:mock_process_thread",
"../modules/video_coding:video_codec_interface",
"../rtc_base:checks",
"../rtc_base:rate_limiter",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"../test:audio_codec_mocks",
"../test:direct_transport",
"../test:field_trial",
"../test:test_common",
"../test:test_support",
"../test:video_test_common",
"../video:video",
"//testing/gtest",
"//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("call_perf_tests") {
testonly = true
sources = [
"call_perf_tests.cc",
"rampup_tests.cc",
"rampup_tests.h",
]
deps = [
":call_interfaces",
":video_stream_api",
"..:webrtc_common",
"../api:simulated_network_api",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/video:video_bitrate_allocation",
"../api/video_codecs:video_codecs_api",
"../logging:rtc_event_log_api",
"../modules/audio_coding",
"../modules/audio_device",
"../modules/audio_device:audio_device_impl",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/rtp_rtcp",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"../system_wrappers:metrics_default",
"../system_wrappers:runtime_enabled_features_default",
"../test:direct_transport",
"../test:field_trial",
"../test:fileutils",
"../test:perf_test",
"../test:test_common",
"../test:test_support",
"../test:video_test_common",
"../video",
"//testing/gtest",
"//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" ]
}
}
# TODO(eladalon): This should be moved, as with the TODO for |rtp_interfaces|.
rtc_source_set("mock_rtp_interfaces") {
testonly = true
sources = [
"test/mock_rtp_packet_sink_interface.h",
"test/mock_rtp_transport_controller_send.h",
]
deps = [
":rtp_interfaces",
"../api:libjingle_peerconnection_api",
"../modules/congestion_controller",
"../modules/pacing",
"../rtc_base:rate_limiter",
"../rtc_base:rtc_base",
"../test:test_support",
]
}
rtc_source_set("mock_bitrate_allocator") {
testonly = true
sources = [
"test/mock_bitrate_allocator.h",
]
deps = [
":bitrate_allocator",
"//test:test_support",
]
}
rtc_source_set("mock_call_interfaces") {
testonly = true
sources = [
"test/mock_audio_send_stream.h",
]
deps = [
":call_interfaces",
"//test:test_support",
]
}
rtc_test("fake_network_unittests") {
deps = [
":call_interfaces",
":fake_network",
":simulated_network",
"../modules/rtp_rtcp",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"../test:test_common",
"../test:test_main",
"//testing/gtest",
]
sources = [
"test/fake_network_pipe_unittest.cc",
]
}
}
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