blob: 11758780cfaa80752cebcda87cbac41d476060f8 [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")
rtc_library("version") {
sources = [
"version.cc",
"version.h",
]
visibility = [ ":*" ]
}
rtc_library("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 = [
":audio_sender_interface",
":receive_stream_interface",
":rtp_interfaces",
":video_stream_api",
"../api:fec_controller_api",
"../api:field_trials_view",
"../api:frame_transformer_interface",
"../api:network_state_predictor_api",
"../api:rtc_error",
"../api:rtp_headers",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:transport_api",
"../api/adaptation:resource_adaptation_api",
"../api/audio:audio_frame_processor",
"../api/audio:audio_mixer_api",
"../api/audio_codecs:audio_codecs_api",
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
"../api/metronome",
"../api/neteq:neteq_api",
"../api/task_queue",
"../api/transport:bitrate_settings",
"../api/transport:network_control",
"../modules/async_audio_processing",
"../modules/audio_device",
"../modules/audio_processing",
"../modules/audio_processing:api",
"../modules/audio_processing:audio_processing_statistics",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/utility",
"../rtc_base",
"../rtc_base:audio_format_to_string",
"../rtc_base:checks",
"../rtc_base:refcount",
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
"../rtc_base/network:sent_packet",
]
absl_deps = [
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("audio_sender_interface") {
visibility = [ "*" ]
sources = [ "audio_sender.h" ]
deps = [ "../api/audio:audio_frame_api" ]
}
# TODO(nisse): These RTP targets should be moved elsewhere
# when interfaces have stabilized. See also TODO for `mock_rtp_interfaces`.
rtc_library("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 = [
"rtp_config.cc",
"rtp_config.h",
"rtp_packet_sink_interface.h",
"rtp_stream_receiver_controller_interface.h",
"rtp_transport_config.h",
"rtp_transport_controller_send_factory_interface.h",
"rtp_transport_controller_send_interface.h",
]
deps = [
"../api:array_view",
"../api:fec_controller_api",
"../api:field_trials_view",
"../api:frame_transformer_interface",
"../api:network_state_predictor_api",
"../api:rtp_headers",
"../api:rtp_parameters",
"../api/crypto:options",
"../api/rtc_event_log",
"../api/transport:bitrate_settings",
"../api/transport:network_control",
"../api/units:timestamp",
"../common_video:frame_counts",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/utility",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../rtc_base:stringutils",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("rtp_receiver") {
visibility = [ "*" ]
sources = [
"rtp_demuxer.cc",
"rtp_demuxer.h",
"rtp_stream_receiver_controller.cc",
"rtp_stream_receiver_controller.h",
"rtx_receive_stream.cc",
"rtx_receive_stream.h",
]
deps = [
":rtp_interfaces",
"../api:array_view",
"../api:rtp_headers",
"../api:sequence_checker",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
"../rtc_base/containers:flat_map",
"../rtc_base/containers:flat_set",
]
absl_deps = [
"//third_party/abseil-cpp/absl/strings:strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("rtp_sender") {
sources = [
"rtp_payload_params.cc",
"rtp_payload_params.h",
"rtp_transport_controller_send.cc",
"rtp_transport_controller_send.h",
"rtp_transport_controller_send_factory.h",
"rtp_video_sender.cc",
"rtp_video_sender.h",
"rtp_video_sender_interface.h",
]
deps = [
":bitrate_configurator",
":rtp_interfaces",
"../api:array_view",
"../api:bitrate_allocation",
"../api:fec_controller_api",
"../api:field_trials_view",
"../api:network_state_predictor_api",
"../api:rtp_parameters",
"../api:sequence_checker",
"../api:transport_api",
"../api/rtc_event_log",
"../api/transport:field_trial_based_config",
"../api/transport:goog_cc",
"../api/transport:network_control",
"../api/units:data_rate",
"../api/units:time_delta",
"../api/units:timestamp",
"../api/video:video_frame",
"../api/video:video_layers_allocation",
"../api/video:video_rtp_headers",
"../api/video_codecs:video_codecs_api",
"../logging:rtc_event_bwe",
"../modules/congestion_controller",
"../modules/congestion_controller/rtp:control_handler",
"../modules/congestion_controller/rtp:transport_feedback",
"../modules/pacing",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/rtp_rtcp:rtp_video_header",
"../modules/utility",
"../modules/video_coding:chain_diff_calculator",
"../modules/video_coding:codec_globals_headers",
"../modules/video_coding:frame_dependencies_calculator",
"../modules/video_coding:video_codec_interface",
"../rtc_base",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:rate_limiter",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../rtc_base:timeutils",
"../rtc_base/synchronization:mutex",
"../rtc_base/task_utils:repeating_task",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/strings:strings",
"//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
rtc_library("bitrate_configurator") {
sources = [
"rtp_bitrate_configurator.cc",
"rtp_bitrate_configurator.h",
]
deps = [
":rtp_interfaces",
# For api/bitrate_constraints.h
"../api:libjingle_peerconnection_api",
"../api/transport:bitrate_settings",
"../api/units:data_rate",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_library("bitrate_allocator") {
sources = [
"bitrate_allocator.cc",
"bitrate_allocator.h",
]
deps = [
"../api:bitrate_allocation",
"../api:sequence_checker",
"../api/transport:network_control",
"../api/units:data_rate",
"../api/units:time_delta",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:rtc_base_approved",
"../rtc_base:safe_minmax",
"../rtc_base/system:no_unique_address",
"../system_wrappers",
"../system_wrappers:field_trial",
"../system_wrappers:metrics",
]
absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
}
rtc_library("call") {
sources = [
"call.cc",
"call_factory.cc",
"call_factory.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",
":version",
":video_stream_api",
"../api:array_view",
"../api:callfactory_api",
"../api:fec_controller_api",
"../api:field_trials_view",
"../api:rtp_headers",
"../api:rtp_parameters",
"../api:sequence_checker",
"../api:simulated_network_api",
"../api:transport_api",
"../api/rtc_event_log",
"../api/transport:network_control",
"../api/units:time_delta",
"../api/video_codecs:video_codecs_api",
"../audio",
"../logging:rtc_event_audio",
"../logging:rtc_event_rtp_rtcp",
"../logging:rtc_event_video",
"../logging:rtc_stream_config",
"../modules/congestion_controller",
"../modules/pacing",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/utility",
"../modules/video_coding",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:rate_limiter",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../rtc_base:safe_minmax",
"../rtc_base:stringutils",
"../rtc_base:timeutils",
"../rtc_base/experiments:field_trial_parser",
"../rtc_base/network:sent_packet",
"../rtc_base/system:no_unique_address",
"../rtc_base/task_utils:pending_task_safety_flag",
"../system_wrappers",
"../system_wrappers:field_trial",
"../system_wrappers:metrics",
"../video",
"../video:decode_synchronizer",
"adaptation:resource_adaptation",
]
absl_deps = [
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("receive_stream_interface") {
sources = [ "receive_stream.h" ]
deps = [
"../api:frame_transformer_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api/crypto:frame_decryptor_interface",
"../api/transport/rtp:rtp_source",
]
}
rtc_library("video_stream_api") {
sources = [
"video_receive_stream.cc",
"video_receive_stream.h",
"video_send_stream.cc",
"video_send_stream.h",
]
deps = [
":receive_stream_interface",
":rtp_interfaces",
"../api:frame_transformer_interface",
"../api:rtp_headers",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:transport_api",
"../api/adaptation:resource_adaptation_api",
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
"../api/video:recordable_encoded_frame",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
"../api/video:video_stream_encoder",
"../api/video_codecs:video_codecs_api",
"../common_video",
"../common_video:frame_counts",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_library("simulated_network") {
sources = [
"simulated_network.cc",
"simulated_network.h",
]
deps = [
"../api:sequence_checker",
"../api:simulated_network_api",
"../api/units:data_rate",
"../api/units:data_size",
"../api/units:time_delta",
"../api/units:timestamp",
"../rtc_base:checks",
"../rtc_base:macromagic",
"../rtc_base:rtc_base_approved",
"../rtc_base/synchronization:mutex",
]
absl_deps = [ "//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",
]
}
rtc_library("fake_network") {
sources = [
"fake_network_pipe.cc",
"fake_network_pipe.h",
]
deps = [
":call_interfaces",
":simulated_network",
":simulated_packet_receiver",
"../api:rtp_parameters",
"../api:sequence_checker",
"../api:simulated_network_api",
"../api:transport_api",
"../modules/utility",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:rtc_base_approved",
"../rtc_base/synchronization:mutex",
"../system_wrappers",
]
}
if (rtc_include_tests) {
if (!build_with_chromium) {
rtc_library("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",
"rtp_bitrate_configurator_unittest.cc",
"rtp_demuxer_unittest.cc",
"rtp_payload_params_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",
"../api:array_view",
"../api:create_frame_generator",
"../api:mock_audio_mixer",
"../api:rtp_headers",
"../api:rtp_parameters",
"../api:transport_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/rtc_event_log",
"../api/task_queue:default_task_queue_factory",
"../api/test/video:function_video_factory",
"../api/transport:field_trial_based_config",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
"../audio",
"../modules:module_api",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/audio_processing:mocks",
"../modules/congestion_controller",
"../modules/pacing",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:mock_rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/utility:mock_process_thread",
"../modules/video_coding",
"../modules/video_coding:codec_globals_headers",
"../modules/video_coding:video_codec_interface",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:rate_limiter",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_event",
"../rtc_base:safe_conversions",
"../rtc_base:task_queue_for_test",
"../rtc_base:timeutils",
"../rtc_base/synchronization:mutex",
"../system_wrappers",
"../test:audio_codec_mocks",
"../test:direct_transport",
"../test:encoder_settings",
"../test:explicit_key_value_config",
"../test:fake_video_codecs",
"../test:field_trial",
"../test:mock_frame_transformer",
"../test:mock_transport",
"../test:scoped_key_value_config",
"../test:test_common",
"../test:test_support",
"../test:video_test_common",
"../test/scenario",
"../test/time_controller:time_controller",
"../video",
"adaptation:resource_adaptation_test_utilities",
"//testing/gmock",
"//testing/gtest",
]
absl_deps = [
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
rtc_library("call_perf_tests") {
testonly = true
sources = [
"call_perf_tests.cc",
"rampup_tests.cc",
"rampup_tests.h",
]
deps = [
":call_interfaces",
":simulated_network",
":video_stream_api",
"../api:rtc_event_log_output_file",
"../api:simulated_network_api",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/rtc_event_log",
"../api/rtc_event_log:rtc_event_log_factory",
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_bitrate_allocation",
"../api/video_codecs:video_codecs_api",
"../media:rtc_internal_video_codecs",
"../media:rtc_simulcast_encoder_adapter",
"../modules/audio_coding",
"../modules/audio_device",
"../modules/audio_device:audio_device_impl",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_event",
"../rtc_base:stringutils",
"../rtc_base:task_queue_for_test",
"../rtc_base:threading",
"../rtc_base:timeutils",
"../rtc_base/synchronization:mutex",
"../rtc_base/task_utils:pending_task_safety_flag",
"../rtc_base/task_utils:repeating_task",
"../system_wrappers",
"../system_wrappers:metrics",
"../test:direct_transport",
"../test:encoder_settings",
"../test:fake_video_codecs",
"../test:field_trial",
"../test:fileutils",
"../test:null_transport",
"../test:perf_test",
"../test:test_common",
"../test:test_support",
"../test:video_test_common",
"../video",
"//testing/gtest",
]
absl_deps = [ "//third_party/abseil-cpp/absl/flags:flag" ]
}
}
# 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:frame_transformer_interface",
"../api:libjingle_peerconnection_api",
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
"../api/transport:bitrate_settings",
"../modules/pacing",
"../rtc_base",
"../rtc_base:rate_limiter",
"../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_library("fake_network_pipe_unittests") {
testonly = true
sources = [
"fake_network_pipe_unittest.cc",
"simulated_network_unittest.cc",
]
deps = [
":fake_network",
":simulated_network",
"../api/units:data_rate",
"../system_wrappers",
"../test:test_support",
"//testing/gtest",
]
absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
}
}