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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_library("version") {
+ sources = [
+ "version.cc",
+ "version.h",
+ ]
+ visibility = [ ":*" ]
+}
+
+rtc_library("call_interfaces") {
+ sources = [
+ "audio_receive_stream.cc",
+ "audio_receive_stream.h",
+ "audio_send_stream.cc",
+ "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 += [
+ "call_basic_stats.cc",
+ "call_basic_stats.h",
+ ]
+ }
+
+ 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:rtp_sender_setparameters_callback",
+ "../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",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
+ "../rtc_base:audio_format_to_string",
+ "../rtc_base:checks",
+ "../rtc_base:copy_on_write_buffer",
+ "../rtc_base:network_route",
+ "../rtc_base:refcount",
+ "../rtc_base:stringutils",
+ "../rtc_base/network:sent_packet",
+ ]
+ absl_deps = [
+ "//third_party/abseil-cpp/absl/functional:any_invocable",
+ "//third_party/abseil-cpp/absl/functional:bind_front",
+ "//third_party/abseil-cpp/absl/strings",
+ "//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",
+ "../rtc_base:checks",
+ "../rtc_base:rtc_task_queue",
+ "../rtc_base:stringutils",
+ ]
+ absl_deps = [
+ "//third_party/abseil-cpp/absl/algorithm:container",
+ "//third_party/abseil-cpp/absl/strings",
+ "//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:stringutils",
+ "../rtc_base/containers:flat_map",
+ "../rtc_base/containers:flat_set",
+ "../rtc_base/system:no_unique_address",
+ ]
+ 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/task_queue:pending_task_safety_flag",
+ "../api/task_queue:task_queue",
+ "../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/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:checks",
+ "../rtc_base:event_tracer",
+ "../rtc_base:logging",
+ "../rtc_base:macromagic",
+ "../rtc_base:network_route",
+ "../rtc_base:race_checker",
+ "../rtc_base:random",
+ "../rtc_base:rate_limiter",
+ "../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",
+ ]
+ 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: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/task_queue:pending_task_safety_flag",
+ "../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/video_coding",
+ "../rtc_base:checks",
+ "../rtc_base:copy_on_write_buffer",
+ "../rtc_base:event_tracer",
+ "../rtc_base:logging",
+ "../rtc_base:macromagic",
+ "../rtc_base:rate_limiter",
+ "../rtc_base:rtc_event",
+ "../rtc_base:rtc_task_queue",
+ "../rtc_base:safe_minmax",
+ "../rtc_base:stringutils",
+ "../rtc_base:threading",
+ "../rtc_base:timeutils",
+ "../rtc_base/experiments:field_trial_parser",
+ "../rtc_base/network:sent_packet",
+ "../rtc_base/system:no_unique_address",
+ "../rtc_base/task_utils:repeating_task",
+ "../system_wrappers",
+ "../system_wrappers:field_trial",
+ "../system_wrappers:metrics",
+ "../video",
+ "../video:decode_synchronizer",
+ "../video/config:encoder_config",
+ "adaptation:resource_adaptation",
+ ]
+ absl_deps = [
+ "//third_party/abseil-cpp/absl/functional:bind_front",
+ "//third_party/abseil-cpp/absl/strings",
+ "//third_party/abseil-cpp/absl/types:optional",
+ ]
+ if (build_with_mozilla) { # See Bug 1820869.
+ sources -= [
+ "call_factory.cc",
+ "degraded_call.cc",
+ ]
+ deps -= [
+ ":fake_network",
+ ":simulated_network",
+ ]
+ }
+}
+
+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",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
+ ]
+}
+
+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:rtp_sender_setparameters_callback",
+ "../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:scalability_mode",
+ "../api/video_codecs:video_codecs_api",
+ "../common_video",
+ "../common_video:frame_counts",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
+ "../rtc_base:checks",
+ "../rtc_base:stringutils",
+ "../video/config:encoder_config",
+ ]
+ absl_deps = [
+ "//third_party/abseil-cpp/absl/functional:any_invocable",
+ "//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:race_checker",
+ "../rtc_base:random",
+ "../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",
+ "../api/units:timestamp",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
+ "../rtc_base:checks",
+ "../rtc_base:logging",
+ "../rtc_base:macromagic",
+ "../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/units:timestamp",
+ "../api/video:builtin_video_bitrate_allocator_factory",
+ "../api/video:video_frame",
+ "../api/video:video_rtp_headers",
+ "../audio",
+ "../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/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:random",
+ "../rtc_base:rate_limiter",
+ "../rtc_base:rtc_event",
+ "../rtc_base:safe_conversions",
+ "../rtc_base:task_queue_for_test",
+ "../rtc_base:threading",
+ "../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:frame_generator_capturer",
+ "../test:mock_frame_transformer",
+ "../test:mock_transport",
+ "../test:run_loop",
+ "../test:scoped_key_value_config",
+ "../test:test_common",
+ "../test:test_support",
+ "../test:video_test_constants",
+ "../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/functional:any_invocable",
+ "//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/numerics",
+ "../api/rtc_event_log",
+ "../api/rtc_event_log:rtc_event_log_factory",
+ "../api/task_queue",
+ "../api/task_queue:default_task_queue_factory",
+ "../api/task_queue:pending_task_safety_flag",
+ "../api/test/metrics:global_metrics_logger_and_exporter",
+ "../api/test/metrics:metric",
+ "../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:test_audio_device_module",
+ "../modules/audio_mixer:audio_mixer_impl",
+ "../modules/rtp_rtcp",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
+ "../rtc_base:checks",
+ "../rtc_base:logging",
+ "../rtc_base:macromagic",
+ "../rtc_base:platform_thread",
+ "../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:repeating_task",
+ "../system_wrappers",
+ "../system_wrappers:metrics",
+ "../test:direct_transport",
+ "../test:encoder_settings",
+ "../test:fake_video_codecs",
+ "../test:field_trial",
+ "../test:fileutils",
+ "../test:frame_generator_capturer",
+ "../test:null_transport",
+ "../test:test_common",
+ "../test:test_flags",
+ "../test:test_support",
+ "../test:video_test_common",
+ "../test:video_test_constants",
+ "../video",
+ "../video/config:encoder_config",
+ "//testing/gtest",
+ ]
+ absl_deps = [
+ "//third_party/abseil-cpp/absl/flags:flag",
+ "//third_party/abseil-cpp/absl/strings",
+ ]
+ }
+ }
+
+ # 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:network_route",
+ "../rtc_base:rate_limiter",
+ "../rtc_base/network:sent_packet",
+ "../test:test_support",
+ ]
+ absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+ }
+ 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:simulated_network_api",
+ "../api/units:data_rate",
+ "../api/units:time_delta",
+ "../api/units:timestamp",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
+ "../rtc_base:checks",
+ "../system_wrappers",
+ "../test:test_support",
+ "//testing/gtest",
+ ]
+ absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
+ }
+}