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platform-external-webrtc/call/BUILD.gn
Jiawei Ou c2ebe21ba9 Reland "Use the factory instead of using the builtin code path in VideoCodecInitializer" 
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782

This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.

Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.

One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.

Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
2018-11-08 19:10:47 +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:api",
"../modules/audio_processing:audio_processing",
"../modules/audio_processing:audio_processing_statistics",
"../rtc_base:audio_format_to_string",
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base/network:sent_packet",
"//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") {
# Client code SHOULD NOT USE THIS TARGET, but for now it needs to be public
# because there exists client code that uses it.
# TODO(bugs.webrtc.org/9808): Move to private visibility as soon as that
# client code gets updated.
visibility = [ "*" ]
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:fec_controller_api",
"../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:fec_controller_api",
"../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",
"//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",
"../system_wrappers:metrics",
]
}
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",
]
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/experiments:field_trial_parser",
"../rtc_base/synchronization:rw_lock_wrapper",
"../system_wrappers",
"../system_wrappers:field_trial",
"../system_wrappers:metrics",
"../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",
"../api/units:data_rate",
"../api/units:data_size",
"../api/units:time_delta",
"../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",
"../modules/utility",
"../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",
":simulated_network",
"..: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",
"../modules/video_coding:video_coding",
"../rtc_base:checks",
"../rtc_base:rate_limiter",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"../test:audio_codec_mocks",
"../test:direct_transport",
"../test:fake_video_codecs",
"../test:field_trial",
"../test:test_common",
"../test:test_support",
"../test:video_test_common",
"../video:video",
"//testing/gmock",
"//testing/gtest",
"//third_party/abseil-cpp/absl/memory",
"//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("call_perf_tests") {
testonly = true
sources = [
"call_perf_tests.cc",
"rampup_tests.cc",
"rampup_tests.h",
]
deps = [
":call_interfaces",
":simulated_network",
":video_stream_api",
"..:webrtc_common",
"../api:simulated_network_api",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_bitrate_allocation",
"../api/video_codecs:video_codecs_api",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_output",
"../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",
"../test:direct_transport",
"../test:fake_video_codecs",
"../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",
"../rtc_base/network:sent_packet",
"../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") {
sources = [
"test/fake_network_pipe_unittest.cc",
]
deps = [
":call_interfaces",
":fake_network",
":simulated_network",
"../modules/rtp_rtcp",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"../test:test_common",
"../test:test_main",
"../test:test_support",
"//testing/gtest",
]
}
}
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