diff --git a/audio/BUILD.gn b/audio/BUILD.gn index cb83e98aa4..fa8b029f83 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -162,6 +162,7 @@ if (rtc_include_tests) { ":audio_end_to_end_test", ":channel_receive_unittest", "../api:array_view", + "../api:bitrate_allocation", "../api:frame_transformer_factory", "../api:frame_transformer_interface", "../api:libjingle_peerconnection_api", @@ -173,6 +174,7 @@ if (rtc_include_tests) { "../api:mock_transformable_audio_frame", "../api:rtp_headers", "../api:scoped_refptr", + "../api:transport_api", "../api/audio:audio_frame_api", "../api/audio:audio_processing_statistics", "../api/audio_codecs:audio_codecs_api", @@ -180,10 +182,13 @@ if (rtc_include_tests) { "../api/audio_codecs/opus:audio_decoder_opus", "../api/audio_codecs/opus:audio_encoder_opus", "../api/crypto:frame_decryptor_interface", + "../api/crypto:options", "../api/environment", "../api/environment:environment_factory", "../api/task_queue:default_task_queue_factory", "../api/task_queue/test:mock_task_queue_base", + "../api/transport:bitrate_settings", + "../api/units:data_rate", "../api/units:time_delta", "../api/units:timestamp", "../call:mock_bitrate_allocator", diff --git a/audio/channel_send_unittest.cc b/audio/channel_send_unittest.cc index ac82fbd052..2260c810c4 100644 --- a/audio/channel_send_unittest.cc +++ b/audio/channel_send_unittest.cc @@ -10,19 +10,40 @@ #include "audio/channel_send.h" +#include +#include +#include +#include #include +#include +#include "api/array_view.h" #include "api/audio/audio_frame.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_encoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "api/call/bitrate_allocation.h" +#include "api/call/transport.h" +#include "api/crypto/crypto_options.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" +#include "api/frame_transformer_interface.h" +#include "api/make_ref_counted.h" +#include "api/rtp_headers.h" #include "api/scoped_refptr.h" #include "api/test/mock_frame_transformer.h" #include "api/test/mock_transformable_audio_frame.h" +#include "api/transport/bitrate_settings.h" +#include "api/units/data_rate.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" +#include "call/rtp_transport_config.h" #include "call/rtp_transport_controller_send.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "rtc_base/gunit.h" +#include "test/gmock.h" #include "test/gtest.h" #include "test/mock_transport.h" #include "test/scoped_key_value_config.h" diff --git a/call/BUILD.gn b/call/BUILD.gn index fd387d980b..1094b48002 100644 --- a/call/BUILD.gn +++ b/call/BUILD.gn @@ -42,8 +42,10 @@ rtc_library("call_interfaces") { ":rtp_interfaces", ":video_stream_api", "../api:fec_controller_api", + "../api:field_trials_view", "../api:frame_transformer_interface", "../api:network_state_predictor_api", + "../api:ref_count", "../api:rtp_headers", "../api:rtp_parameters", "../api:rtp_sender_interface", @@ -56,6 +58,7 @@ rtc_library("call_interfaces") { "../api/audio:audio_processing", "../api/audio:audio_processing_statistics", "../api/audio_codecs:audio_codecs_api", + "../api/crypto:frame_decryptor_interface", "../api/crypto:frame_encryptor_interface", "../api/crypto:options", "../api/environment", @@ -64,6 +67,8 @@ rtc_library("call_interfaces") { "../api/task_queue", "../api/transport:bitrate_settings", "../api/transport:network_control", + "../api/units:time_delta", + "../api/units:timestamp", "../modules/async_audio_processing", "../modules/audio_device", "../modules/audio_processing", @@ -76,6 +81,7 @@ rtc_library("call_interfaces") { "../rtc_base:refcount", "../rtc_base:stringutils", "../rtc_base/network:sent_packet", + "../video/config:encoder_config", "//third_party/abseil-cpp/absl/functional:any_invocable", "//third_party/abseil-cpp/absl/strings:string_view", ] @@ -112,6 +118,7 @@ rtc_library("rtp_interfaces") { "../api:rtp_headers", "../api:rtp_packet_sender", "../api:rtp_parameters", + "../api:scoped_refptr", "../api/crypto:options", "../api/environment", "../api/transport:bandwidth_estimation_settings", @@ -147,6 +154,7 @@ rtc_library("rtp_receiver") { "../modules/rtp_rtcp:rtp_rtcp_format", "../rtc_base:checks", "../rtc_base:logging", + "../rtc_base:macromagic", "../rtc_base:stringutils", "../rtc_base/containers:flat_map", "../rtc_base/containers:flat_set", @@ -173,25 +181,43 @@ rtc_library("rtp_sender") { "../api:bitrate_allocation", "../api:fec_controller_api", "../api:field_trials_view", + "../api:frame_transformer_interface", "../api:network_state_predictor_api", + "../api:rtp_headers", + "../api:rtp_packet_sender", "../api:rtp_parameters", + "../api:scoped_refptr", "../api:sequence_checker", "../api:transport_api", + "../api/crypto:options", "../api/environment", "../api/rtc_event_log", "../api/task_queue:pending_task_safety_flag", "../api/task_queue:task_queue", + "../api/transport:bandwidth_estimation_settings", + "../api/transport:bitrate_settings", "../api/transport:field_trial_based_config", "../api/transport:goog_cc", "../api/transport:network_control", + "../api/transport/rtp:dependency_descriptor", "../api/units:data_rate", + "../api/units:data_size", + "../api/units:frequency", "../api/units:time_delta", "../api/units:timestamp", + "../api/video:encoded_image", + "../api/video:render_resolution", + "../api/video:video_bitrate_allocation", + "../api/video:video_codec_constants", "../api/video:video_frame", + "../api/video:video_frame_type", "../api/video:video_layers_allocation", "../api/video:video_rtp_headers", "../api/video_codecs:video_codecs_api", + "../common_video:frame_counts", + "../common_video/generic_frame_descriptor", "../logging:rtc_event_bwe", + "../modules:module_fec_api", "../modules/congestion_controller", "../modules/congestion_controller/rtp:control_handler", "../modules/congestion_controller/rtp:transport_feedback", @@ -211,8 +237,12 @@ rtc_library("rtp_sender") { "../rtc_base:race_checker", "../rtc_base:random", "../rtc_base:rate_limiter", + "../rtc_base:safe_conversions", "../rtc_base:timeutils", + "../rtc_base/experiments:field_trial_parser", + "../rtc_base/network:sent_packet", "../rtc_base/synchronization:mutex", + "../rtc_base/system:no_unique_address", "../rtc_base/task_utils:repeating_task", "//third_party/abseil-cpp/absl/algorithm:container", "//third_party/abseil-cpp/absl/container:inlined_vector", @@ -252,6 +282,8 @@ rtc_library("bitrate_allocator") { "../api/units:time_delta", "../rtc_base:checks", "../rtc_base:logging", + "../rtc_base:macromagic", + "../rtc_base:safe_conversions", "../rtc_base:safe_minmax", "../rtc_base/experiments:field_trial_parser", "../rtc_base/system:no_unique_address", @@ -275,6 +307,7 @@ rtc_library("call") { ":bitrate_allocator", ":call_interfaces", ":fake_network", + ":receive_stream_interface", ":rtp_interfaces", ":rtp_receiver", ":rtp_sender", @@ -286,14 +319,21 @@ rtc_library("call") { "../api:field_trials_view", "../api:rtp_headers", "../api:rtp_parameters", + "../api:scoped_refptr", "../api:sequence_checker", "../api:simulated_network_api", "../api:transport_api", + "../api/adaptation:resource_adaptation_api", "../api/environment", "../api/rtc_event_log", + "../api/task_queue", "../api/task_queue:pending_task_safety_flag", + "../api/transport:bitrate_settings", "../api/transport:network_control", + "../api/units:data_rate", + "../api/units:data_size", "../api/units:time_delta", + "../api/units:timestamp", "../api/video_codecs:video_codecs_api", "../audio", "../logging:rtc_event_audio", @@ -305,6 +345,7 @@ rtc_library("call") { "../modules/rtp_rtcp", "../modules/rtp_rtcp:rtp_rtcp_format", "../modules/video_coding", + "../modules/video_coding:nack_requester", "../rtc_base:checks", "../rtc_base:copy_on_write_buffer", "../rtc_base:event_tracer", @@ -337,6 +378,7 @@ rtc_source_set("receive_stream_interface") { sources = [ "receive_stream.h" ] deps = [ "../api:frame_transformer_interface", + "../api:rtp_headers", "../api:rtp_parameters", "../api:scoped_refptr", "../api/crypto:frame_decryptor_interface", @@ -362,8 +404,10 @@ rtc_library("video_stream_api") { "../api:scoped_refptr", "../api:transport_api", "../api/adaptation:resource_adaptation_api", + "../api/crypto:frame_decryptor_interface", "../api/crypto:frame_encryptor_interface", "../api/crypto:options", + "../api/units:time_delta", "../api/video:recordable_encoded_frame", "../api/video:video_frame", "../api/video:video_rtp_headers", @@ -401,6 +445,7 @@ rtc_library("fake_network") { deps = [ ":call_interfaces", ":simulated_packet_receiver", + "../api:array_view", "../api:rtp_parameters", "../api:sequence_checker", "../api:simulated_network_api", @@ -408,6 +453,7 @@ rtc_library("fake_network") { "../api/units:timestamp", "../modules/rtp_rtcp:rtp_rtcp_format", "../rtc_base:checks", + "../rtc_base:copy_on_write_buffer", "../rtc_base:logging", "../rtc_base:macromagic", "../rtc_base/synchronization:mutex", @@ -443,23 +489,44 @@ if (rtc_include_tests) { ":rtp_receiver", ":rtp_sender", ":simulated_network", + ":video_stream_api", "../api:array_view", + "../api:bitrate_allocation", "../api:create_frame_generator", + "../api:frame_transformer_interface", + "../api:make_ref_counted", "../api:mock_audio_mixer", "../api:mock_frame_transformer", "../api:rtp_headers", "../api:rtp_parameters", + "../api:scoped_refptr", + "../api:simulated_network_api", "../api:transport_api", + "../api/adaptation:resource_adaptation_api", "../api/audio_codecs:builtin_audio_decoder_factory", + "../api/crypto:options", "../api/environment", "../api/environment:environment_factory", + "../api/test/network_emulation", "../api/test/video:function_video_factory", + "../api/transport:bitrate_settings", "../api/transport:field_trial_based_config", + "../api/transport:network_control", + "../api/transport/rtp:dependency_descriptor", + "../api/units:data_rate", + "../api/units:data_size", + "../api/units:time_delta", "../api/units:timestamp", "../api/video:builtin_video_bitrate_allocator_factory", + "../api/video:encoded_image", + "../api/video:video_codec_constants", "../api/video:video_frame", + "../api/video:video_frame_type", "../api/video:video_rtp_headers", + "../api/video_codecs:video_codecs_api", "../audio", + "../common_video:frame_counts", + "../common_video/generic_frame_descriptor", "../modules/audio_device:mock_audio_device", "../modules/audio_mixer", "../modules/audio_mixer:audio_mixer_impl", @@ -469,9 +536,11 @@ if (rtc_include_tests) { "../modules/rtp_rtcp", "../modules/rtp_rtcp:mock_rtp_rtcp", "../modules/rtp_rtcp:rtp_rtcp_format", + "../modules/rtp_rtcp:rtp_video_header", "../modules/video_coding", "../modules/video_coding:codec_globals_headers", "../modules/video_coding:video_codec_interface", + "../rtc_base:buffer", "../rtc_base:checks", "../rtc_base:logging", "../rtc_base:macromagic", @@ -500,6 +569,7 @@ if (rtc_include_tests) { "../test/scenario", "../test/time_controller:time_controller", "../video", + "../video/config:encoder_config", "adaptation:resource_adaptation_test_utilities", "//testing/gmock", "//testing/gtest", @@ -522,10 +592,18 @@ if (rtc_include_tests) { ":call_interfaces", ":simulated_network", ":video_stream_api", + "../api:array_view", + "../api:field_trials_view", + "../api:make_ref_counted", "../api:rtc_event_log_output_file", + "../api:rtp_parameters", + "../api:scoped_refptr", + "../api:sequence_checker", "../api:simulated_network_api", "../api/audio:audio_device", + "../api/audio:audio_processing", "../api/audio_codecs:builtin_audio_encoder_factory", + "../api/environment", "../api/numerics", "../api/rtc_event_log", "../api/rtc_event_log:rtc_event_log_factory", @@ -533,9 +611,15 @@ if (rtc_include_tests) { "../api/task_queue:pending_task_safety_flag", "../api/test/metrics:global_metrics_logger_and_exporter", "../api/test/metrics:metric", + "../api/test/video:function_video_factory", + "../api/transport:bitrate_settings", "../api/units:data_rate", + "../api/units:time_delta", + "../api/units:timestamp", "../api/video:builtin_video_bitrate_allocator_factory", "../api/video:video_bitrate_allocation", + "../api/video:video_bitrate_allocator_factory", + "../api/video:video_frame", "../api/video_codecs:video_codecs_api", "../media:rtc_internal_video_codecs", "../media:rtc_simulcast_encoder_adapter", @@ -635,6 +719,7 @@ if (rtc_include_tests) { "../api/units:timestamp", "../modules/rtp_rtcp:rtp_rtcp_format", "../rtc_base:checks", + "../rtc_base:copy_on_write_buffer", "../system_wrappers", "../test:test_support", "../test/network:simulated_network", diff --git a/call/DEPS b/call/DEPS index 236b37e82a..daff3e9efe 100644 --- a/call/DEPS +++ b/call/DEPS @@ -31,6 +31,25 @@ specific_include_rules = { ], "simulated_network\.h": [ "+test/network/simulated_network.h", - ] - + ], + "rtp_payload_params\.cc": [ + "+common_video/generic_frame_descriptor", + ], + "rtp_payload_params\.h": [ + "+common_video/generic_frame_descriptor", + ], + "rtp_payload_params_unittest\.cc": [ + "+common_video/generic_frame_descriptor", + ], + "rtp_video_sender\.cc": [ + "+common_video/frame_counts.h", + "+common_video/generic_frame_descriptor", + ], + "rtp_video_sender.h": [ + "+common_video/frame_counts.h", + ], + "rtp_video_sender_unittest.cc": [ + "+common_video/frame_counts.h", + "+common_video/generic_frame_descriptor", + ], } diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h index 5154360afb..d91e68caa9 100644 --- a/call/audio_receive_stream.h +++ b/call/audio_receive_stream.h @@ -11,16 +11,23 @@ #ifndef CALL_AUDIO_RECEIVE_STREAM_H_ #define CALL_AUDIO_RECEIVE_STREAM_H_ +#include +#include #include -#include #include #include -#include +#include "api/audio_codecs/audio_codec_pair_id.h" #include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/call/transport.h" #include "api/crypto/crypto_options.h" -#include "api/rtp_parameters.h" +#include "api/crypto/frame_decryptor_interface.h" +#include "api/frame_transformer_interface.h" +#include "api/rtp_headers.h" +#include "api/scoped_refptr.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "call/receive_stream.h" #include "call/rtp_config.h" diff --git a/call/audio_send_stream.cc b/call/audio_send_stream.cc index a36050a9f7..bd7300d56d 100644 --- a/call/audio_send_stream.cc +++ b/call/audio_send_stream.cc @@ -12,6 +12,11 @@ #include +#include + +#include "api/audio_codecs/audio_format.h" +#include "api/call/transport.h" +#include "rtc_base/string_encode.h" #include "rtc_base/strings/audio_format_to_string.h" #include "rtc_base/strings/string_builder.h" diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h index 3794dfb6f8..540956c1fd 100644 --- a/call/audio_send_stream.h +++ b/call/audio_send_stream.h @@ -11,7 +11,7 @@ #ifndef CALL_AUDIO_SEND_STREAM_H_ #define CALL_AUDIO_SEND_STREAM_H_ -#include +#include #include #include #include @@ -25,11 +25,12 @@ #include "api/crypto/crypto_options.h" #include "api/crypto/frame_encryptor_interface.h" #include "api/frame_transformer_interface.h" +#include "api/rtp_headers.h" #include "api/rtp_parameters.h" #include "api/rtp_sender_interface.h" #include "api/scoped_refptr.h" +#include "api/units/time_delta.h" #include "call/audio_sender.h" -#include "call/rtp_config.h" #include "modules/rtp_rtcp/include/report_block_data.h" namespace webrtc { diff --git a/call/audio_state.h b/call/audio_state.h index 3a8e7c8b48..7c17231c30 100644 --- a/call/audio_state.h +++ b/call/audio_state.h @@ -13,9 +13,9 @@ #include "api/audio/audio_device.h" #include "api/audio/audio_mixer.h" #include "api/audio/audio_processing.h" +#include "api/ref_count.h" #include "api/scoped_refptr.h" #include "modules/async_audio_processing/async_audio_processing.h" -#include "rtc_base/ref_count.h" namespace webrtc { diff --git a/call/bitrate_allocator.cc b/call/bitrate_allocator.cc index 1b6d63ab47..cd4260f833 100644 --- a/call/bitrate_allocator.cc +++ b/call/bitrate_allocator.cc @@ -13,17 +13,26 @@ #include #include -#include +#include +#include +#include +#include +#include #include +#include #include "absl/algorithm/container.h" +#include "api/call/bitrate_allocation.h" +#include "api/field_trials_view.h" +#include "api/sequence_checker.h" +#include "api/transport/network_types.h" #include "api/units/data_rate.h" #include "api/units/time_delta.h" #include "rtc_base/checks.h" #include "rtc_base/experiments/field_trial_parser.h" #include "rtc_base/logging.h" +#include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/numerics/safe_minmax.h" -#include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" namespace webrtc { diff --git a/call/bitrate_allocator.h b/call/bitrate_allocator.h index 33dd0e9ed7..270a34e1f7 100644 --- a/call/bitrate_allocator.h +++ b/call/bitrate_allocator.h @@ -13,10 +13,7 @@ #include -#include -#include -#include -#include +#include #include #include "api/call/bitrate_allocation.h" @@ -25,6 +22,7 @@ #include "api/transport/network_types.h" #include "api/units/data_rate.h" #include "rtc_base/system/no_unique_address.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/call/bitrate_allocator_unittest.cc b/call/bitrate_allocator_unittest.cc index 7b4fff3e2b..b8f0abf524 100644 --- a/call/bitrate_allocator_unittest.cc +++ b/call/bitrate_allocator_unittest.cc @@ -11,11 +11,18 @@ #include "call/bitrate_allocator.h" #include +#include #include -#include +#include +#include #include "absl/strings/string_view.h" -#include "system_wrappers/include/clock.h" +#include "api/call/bitrate_allocation.h" +#include "api/transport/network_types.h" +#include "api/units/data_rate.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "rtc_base/numerics/safe_conversions.h" #include "test/explicit_key_value_config.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/call/bitrate_estimator_tests.cc b/call/bitrate_estimator_tests.cc index 379f947794..d66e4cc115 100644 --- a/call/bitrate_estimator_tests.cc +++ b/call/bitrate_estimator_tests.cc @@ -11,11 +11,20 @@ #include #include #include +#include #include +#include #include "absl/strings/string_view.h" +#include "api/rtp_parameters.h" #include "api/test/create_frame_generator.h" +#include "api/test/simulated_network.h" +#include "api/test/video/function_video_decoder_factory.h" +#include "api/video/video_codec_type.h" +#include "api/video_codecs/sdp_video_format.h" #include "call/call.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "rtc_base/checks.h" #include "rtc_base/event.h" #include "rtc_base/logging.h" @@ -25,10 +34,10 @@ #include "test/call_test.h" #include "test/encoder_settings.h" #include "test/fake_decoder.h" -#include "test/fake_encoder.h" #include "test/frame_generator_capturer.h" #include "test/gtest.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { diff --git a/call/call.cc b/call/call.cc index aa48491436..0b1ca6f237 100644 --- a/call/call.cc +++ b/call/call.cc @@ -19,29 +19,47 @@ #include #include #include +#include #include #include #include "absl/functional/bind_front.h" #include "absl/strings/string_view.h" +#include "api/adaptation/resource.h" +#include "api/environment/environment.h" +#include "api/fec_controller.h" +#include "api/field_trials_view.h" #include "api/media_types.h" #include "api/rtc_event_log/rtc_event_log.h" +#include "api/rtp_headers.h" +#include "api/scoped_refptr.h" #include "api/sequence_checker.h" #include "api/task_queue/pending_task_safety_flag.h" +#include "api/task_queue/task_queue_base.h" +#include "api/transport/bitrate_settings.h" #include "api/transport/network_control.h" +#include "api/transport/network_types.h" +#include "api/units/data_rate.h" +#include "api/units/data_size.h" #include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "audio/audio_receive_stream.h" #include "audio/audio_send_stream.h" #include "audio/audio_state.h" #include "call/adaptation/broadcast_resource_listener.h" #include "call/bitrate_allocator.h" +#include "call/call_config.h" +#include "call/flexfec_receive_stream.h" #include "call/flexfec_receive_stream_impl.h" #include "call/packet_receiver.h" +#include "call/receive_stream.h" #include "call/receive_time_calculator.h" +#include "call/rtp_config.h" #include "call/rtp_stream_receiver_controller.h" -#include "call/rtp_transport_controller_send.h" #include "call/rtp_transport_controller_send_factory.h" #include "call/version.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h" #include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h" #include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h" @@ -50,17 +68,20 @@ #include "logging/rtc_event_log/rtc_stream_config.h" #include "modules/congestion_controller/include/receive_side_congestion_controller.h" #include "modules/rtp_rtcp/include/flexfec_receiver.h" -#include "modules/rtp_rtcp/include/rtp_header_extension_map.h" -#include "modules/rtp_rtcp/source/byte_io.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_header_extensions.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "modules/rtp_rtcp/source/rtp_util.h" #include "modules/video_coding/fec_controller_default.h" +#include "modules/video_coding/nack_requester.h" #include "rtc_base/checks.h" +#include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/logging.h" +#include "rtc_base/network/sent_packet.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/system/no_unique_address.h" #include "rtc_base/task_utils/repeating_task.h" +#include "rtc_base/thread.h" #include "rtc_base/thread_annotations.h" #include "rtc_base/time_utils.h" #include "rtc_base/trace_event.h" @@ -68,6 +89,8 @@ #include "system_wrappers/include/cpu_info.h" #include "system_wrappers/include/metrics.h" #include "video/call_stats2.h" +#include "video/config/video_encoder_config.h" +#include "video/decode_synchronizer.h" #include "video/send_delay_stats.h" #include "video/stats_counter.h" #include "video/video_receive_stream2.h" diff --git a/call/call.h b/call/call.h index 6ada035a22..6c6bc46c2a 100644 --- a/call/call.h +++ b/call/call.h @@ -10,26 +10,29 @@ #ifndef CALL_CALL_H_ #define CALL_CALL_H_ -#include +#include #include #include -#include #include "absl/strings/string_view.h" #include "api/adaptation/resource.h" +#include "api/fec_controller.h" +#include "api/field_trials_view.h" #include "api/media_types.h" +#include "api/rtp_headers.h" +#include "api/scoped_refptr.h" #include "api/task_queue/task_queue_base.h" +#include "api/transport/bitrate_settings.h" #include "call/audio_receive_stream.h" #include "call/audio_send_stream.h" #include "call/call_config.h" #include "call/flexfec_receive_stream.h" #include "call/packet_receiver.h" +#include "call/rtp_transport_controller_send_interface.h" #include "call/video_receive_stream.h" #include "call/video_send_stream.h" -#include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/network/sent_packet.h" -#include "rtc_base/network_route.h" -#include "rtc_base/ref_count.h" +#include "video/config/video_encoder_config.h" namespace webrtc { diff --git a/call/call_config.cc b/call/call_config.cc index 19d91cf1b6..6a8117603d 100644 --- a/call/call_config.cc +++ b/call/call_config.cc @@ -12,6 +12,7 @@ #include "api/environment/environment.h" #include "api/task_queue/task_queue_base.h" +#include "call/rtp_transport_config.h" namespace webrtc { diff --git a/call/call_config.h b/call/call_config.h index 74c15509e8..d15651da4e 100644 --- a/call/call_config.h +++ b/call/call_config.h @@ -11,14 +11,18 @@ #define CALL_CALL_CONFIG_H_ #include +#include #include "api/environment/environment.h" #include "api/fec_controller.h" #include "api/metronome/metronome.h" #include "api/neteq/neteq_factory.h" #include "api/network_state_predictor.h" +#include "api/scoped_refptr.h" +#include "api/task_queue/task_queue_base.h" #include "api/transport/bitrate_settings.h" #include "api/transport/network_control.h" +#include "api/units/time_delta.h" #include "call/audio_state.h" #include "call/rtp_transport_config.h" #include "call/rtp_transport_controller_send_factory_interface.h" diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc index ff92a52901..a9db79c2f7 100644 --- a/call/call_perf_tests.cc +++ b/call/call_perf_tests.cc @@ -9,42 +9,68 @@ */ #include +#include +#include +#include +#include #include +#include #include #include +#include +#include #include "absl/flags/flag.h" #include "absl/strings/string_view.h" +#include "api/array_view.h" #include "api/audio/audio_device.h" +#include "api/audio/audio_processing.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "api/environment/environment.h" +#include "api/field_trials_view.h" +#include "api/make_ref_counted.h" #include "api/numerics/samples_stats_counter.h" -#include "api/rtc_event_log/rtc_event_log.h" +#include "api/rtp_parameters.h" +#include "api/scoped_refptr.h" #include "api/task_queue/pending_task_safety_flag.h" #include "api/task_queue/task_queue_base.h" #include "api/test/metrics/global_metrics_logger_and_exporter.h" #include "api/test/metrics/metric.h" #include "api/test/simulated_network.h" +#include "api/test/video/function_video_encoder_factory.h" +#include "api/transport/bitrate_settings.h" #include "api/units/data_rate.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "api/video/builtin_video_bitrate_allocator_factory.h" #include "api/video/video_bitrate_allocation.h" +#include "api/video/video_bitrate_allocator_factory.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "api/video/video_source_interface.h" +#include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_encoder.h" +#include "api/video_codecs/video_encoder_factory.h" +#include "call/audio_receive_stream.h" +#include "call/audio_send_stream.h" +#include "call/audio_state.h" #include "call/call.h" +#include "call/call_config.h" #include "call/fake_network_pipe.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "media/engine/internal_encoder_factory.h" #include "media/engine/simulcast_encoder_adapter.h" -#include "modules/audio_coding/include/audio_coding_module.h" #include "modules/audio_device/include/test_audio_device.h" #include "modules/audio_mixer/audio_mixer_impl.h" -#include "modules/rtp_rtcp/source/rtp_packet.h" #include "rtc_base/checks.h" +#include "rtc_base/event.h" #include "rtc_base/logging.h" -#include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_queue_for_test.h" #include "rtc_base/thread.h" -#include "rtc_base/thread_annotations.h" #include "system_wrappers/include/metrics.h" #include "test/call_test.h" -#include "test/direct_transport.h" #include "test/drifting_clock.h" #include "test/encoder_settings.h" #include "test/fake_encoder.h" @@ -52,14 +78,12 @@ #include "test/frame_generator_capturer.h" #include "test/gtest.h" #include "test/network/simulated_network.h" -#include "test/null_transport.h" #include "test/rtp_rtcp_observer.h" #include "test/test_flags.h" #include "test/testsupport/file_utils.h" #include "test/video_encoder_proxy_factory.h" #include "test/video_test_constants.h" #include "video/config/video_encoder_config.h" -#include "video/transport_adapter.h" using webrtc::test::DriftingClock; diff --git a/call/call_unittest.cc b/call/call_unittest.cc index aa0b08fb5b..9e06795ed2 100644 --- a/call/call_unittest.cc +++ b/call/call_unittest.cc @@ -10,33 +10,46 @@ #include "call/call.h" +#include #include -#include #include +#include #include +#include #include "absl/strings/string_view.h" -#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/adaptation/resource.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" +#include "api/make_ref_counted.h" #include "api/media_types.h" +#include "api/scoped_refptr.h" #include "api/test/mock_audio_mixer.h" #include "api/test/video/function_video_encoder_factory.h" #include "api/units/timestamp.h" #include "api/video/builtin_video_bitrate_allocator_factory.h" +#include "api/video_codecs/sdp_video_format.h" #include "audio/audio_receive_stream.h" #include "audio/audio_send_stream.h" #include "call/adaptation/test/fake_resource.h" #include "call/adaptation/test/mock_resource_listener.h" +#include "call/audio_receive_stream.h" +#include "call/audio_send_stream.h" #include "call/audio_state.h" +#include "call/call_config.h" +#include "call/flexfec_receive_stream.h" +#include "call/video_send_stream.h" #include "modules/audio_device/include/mock_audio_device.h" #include "modules/audio_processing/include/mock_audio_processing.h" -#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "test/fake_encoder.h" +#include "test/gmock.h" #include "test/gtest.h" #include "test/mock_audio_decoder_factory.h" #include "test/mock_transport.h" #include "test/run_loop.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { diff --git a/call/fake_network_pipe.cc b/call/fake_network_pipe.cc index 474d2ac246..7d64b78dfb 100644 --- a/call/fake_network_pipe.cc +++ b/call/fake_network_pipe.cc @@ -13,15 +13,23 @@ #include #include +#include +#include +#include #include #include #include +#include "api/array_view.h" +#include "api/call/transport.h" #include "api/media_types.h" +#include "api/test/simulated_network.h" #include "api/units/timestamp.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "rtc_base/checks.h" +#include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/logging.h" +#include "rtc_base/synchronization/mutex.h" #include "system_wrappers/include/clock.h" namespace webrtc { diff --git a/call/fake_network_pipe.h b/call/fake_network_pipe.h index ea4b1afeff..dabc4b9255 100644 --- a/call/fake_network_pipe.h +++ b/call/fake_network_pipe.h @@ -11,18 +11,19 @@ #ifndef CALL_FAKE_NETWORK_PIPE_H_ #define CALL_FAKE_NETWORK_PIPE_H_ +#include +#include #include #include #include -#include -#include -#include -#include +#include +#include "api/array_view.h" #include "api/call/transport.h" #include "api/test/simulated_network.h" #include "call/simulated_packet_receiver.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" +#include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" diff --git a/call/fake_network_pipe_unittest.cc b/call/fake_network_pipe_unittest.cc index 66e43f69b7..b2775355b4 100644 --- a/call/fake_network_pipe_unittest.cc +++ b/call/fake_network_pipe_unittest.cc @@ -10,9 +10,14 @@ #include "call/fake_network_pipe.h" +#include +#include +#include #include #include +#include +#include "api/test/simulated_network.h" #include "api/units/data_rate.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" @@ -20,6 +25,7 @@ #include "modules/rtp_rtcp/source/rtp_header_extensions.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "rtc_base/checks.h" +#include "rtc_base/copy_on_write_buffer.h" #include "system_wrappers/include/clock.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/call/flexfec_receive_stream.cc b/call/flexfec_receive_stream.cc index ab6dde37b4..27261b8bfe 100644 --- a/call/flexfec_receive_stream.cc +++ b/call/flexfec_receive_stream.cc @@ -10,6 +10,7 @@ #include "call/flexfec_receive_stream.h" +#include "api/call/transport.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/call/flexfec_receive_stream.h b/call/flexfec_receive_stream.h index c5ac0f9fb6..eb70e206ec 100644 --- a/call/flexfec_receive_stream.h +++ b/call/flexfec_receive_stream.h @@ -18,7 +18,6 @@ #include "api/call/transport.h" #include "api/rtp_headers.h" -#include "api/rtp_parameters.h" #include "call/receive_stream.h" #include "call/rtp_packet_sink_interface.h" #include "modules/rtp_rtcp/include/receive_statistics.h" diff --git a/call/flexfec_receive_stream_impl.cc b/call/flexfec_receive_stream_impl.cc index 053cf67a96..bb4122a83f 100644 --- a/call/flexfec_receive_stream_impl.cc +++ b/call/flexfec_receive_stream_impl.cc @@ -13,13 +13,13 @@ #include #include +#include #include -#include #include "api/array_view.h" -#include "api/call/transport.h" #include "api/environment/environment.h" -#include "api/rtp_parameters.h" +#include "api/sequence_checker.h" +#include "call/flexfec_receive_stream.h" #include "call/rtp_stream_receiver_controller_interface.h" #include "modules/rtp_rtcp/include/flexfec_receiver.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" diff --git a/call/flexfec_receive_stream_impl.h b/call/flexfec_receive_stream_impl.h index 623560382c..2e1b9c452c 100644 --- a/call/flexfec_receive_stream_impl.h +++ b/call/flexfec_receive_stream_impl.h @@ -11,14 +11,17 @@ #ifndef CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_ #define CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_ +#include #include -#include #include "api/environment/environment.h" +#include "api/rtp_headers.h" +#include "api/sequence_checker.h" #include "call/flexfec_receive_stream.h" #include "call/rtp_packet_sink_interface.h" #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" #include "rtc_base/system/no_unique_address.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/call/flexfec_receive_stream_unittest.cc b/call/flexfec_receive_stream_unittest.cc index b591d5cb9b..b0a3420b6f 100644 --- a/call/flexfec_receive_stream_unittest.cc +++ b/call/flexfec_receive_stream_unittest.cc @@ -18,14 +18,11 @@ #include "api/call/transport.h" #include "api/environment/environment_factory.h" #include "api/rtp_headers.h" -#include "api/rtp_parameters.h" #include "call/flexfec_receive_stream_impl.h" #include "call/rtp_stream_receiver_controller.h" -#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h" #include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" #include "modules/rtp_rtcp/source/byte_io.h" -#include "modules/rtp_rtcp/source/rtp_header_extensions.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "rtc_base/thread.h" #include "test/gmock.h" diff --git a/call/packet_receiver.h b/call/packet_receiver.h index 6d223efdfa..974c435534 100644 --- a/call/packet_receiver.h +++ b/call/packet_receiver.h @@ -13,7 +13,6 @@ #include "absl/functional/any_invocable.h" #include "api/media_types.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" -#include "rtc_base/checks.h" #include "rtc_base/copy_on_write_buffer.h" namespace webrtc { diff --git a/call/rampup_tests.cc b/call/rampup_tests.cc index 103e998079..4d3946c2b9 100644 --- a/call/rampup_tests.cc +++ b/call/rampup_tests.cc @@ -10,26 +10,48 @@ #include "call/rampup_tests.h" +#include +#include #include +#include +#include +#include #include "absl/flags/flag.h" #include "absl/strings/string_view.h" +#include "api/field_trials_view.h" +#include "api/make_ref_counted.h" +#include "api/rtc_event_log/rtc_event_log.h" #include "api/rtc_event_log/rtc_event_log_factory.h" #include "api/rtc_event_log_output_file.h" +#include "api/rtp_parameters.h" +#include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" #include "api/test/metrics/global_metrics_logger_and_exporter.h" #include "api/test/metrics/metric.h" +#include "api/test/simulated_network.h" +#include "api/transport/bitrate_settings.h" #include "api/units/data_rate.h" +#include "api/units/time_delta.h" +#include "api/video/video_codec_type.h" +#include "api/video_codecs/sdp_video_format.h" +#include "call/audio_receive_stream.h" +#include "call/audio_send_stream.h" +#include "call/call.h" #include "call/fake_network_pipe.h" +#include "call/flexfec_receive_stream.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "rtc_base/checks.h" -#include "rtc_base/logging.h" -#include "rtc_base/platform_thread.h" #include "rtc_base/string_encode.h" #include "rtc_base/task_queue_for_test.h" -#include "rtc_base/time_utils.h" +#include "rtc_base/task_utils/repeating_task.h" +#include "test/call_test.h" #include "test/encoder_settings.h" #include "test/gtest.h" +#include "test/rtp_rtcp_observer.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" ABSL_FLAG(std::string, ramp_dump_name, diff --git a/call/rampup_tests.h b/call/rampup_tests.h index e09986d6ec..259aa79954 100644 --- a/call/rampup_tests.h +++ b/call/rampup_tests.h @@ -11,21 +11,27 @@ #ifndef CALL_RAMPUP_TESTS_H_ #define CALL_RAMPUP_TESTS_H_ +#include +#include #include -#include #include -#include #include #include "absl/strings/string_view.h" -#include "api/rtc_event_log/rtc_event_log.h" #include "api/task_queue/task_queue_base.h" #include "api/test/metrics/metric.h" #include "api/test/simulated_network.h" +#include "api/transport/bitrate_settings.h" +#include "call/audio_receive_stream.h" +#include "call/audio_send_stream.h" #include "call/call.h" -#include "rtc_base/event.h" +#include "call/flexfec_receive_stream.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "rtc_base/task_utils/repeating_task.h" #include "test/call_test.h" +#include "test/rtp_rtcp_observer.h" +#include "video/config/video_encoder_config.h" namespace webrtc { diff --git a/call/receive_stream.h b/call/receive_stream.h index 287ddc47a8..14655b7590 100644 --- a/call/receive_stream.h +++ b/call/receive_stream.h @@ -11,11 +11,12 @@ #ifndef CALL_RECEIVE_STREAM_H_ #define CALL_RECEIVE_STREAM_H_ +#include #include #include "api/crypto/frame_decryptor_interface.h" #include "api/frame_transformer_interface.h" -#include "api/media_types.h" +#include "api/rtp_headers.h" #include "api/scoped_refptr.h" #include "api/transport/rtp/rtp_source.h" diff --git a/call/receive_time_calculator.cc b/call/receive_time_calculator.cc index 417168b15d..e73e5ff86f 100644 --- a/call/receive_time_calculator.cc +++ b/call/receive_time_calculator.cc @@ -10,10 +10,12 @@ #include "call/receive_time_calculator.h" +#include #include #include -#include +#include "api/field_trials_view.h" +#include "api/units/time_delta.h" #include "rtc_base/experiments/field_trial_parser.h" #include "rtc_base/numerics/safe_minmax.h" diff --git a/call/rtp_bitrate_configurator.cc b/call/rtp_bitrate_configurator.cc index f0a36ad6ef..3e7912b6bc 100644 --- a/call/rtp_bitrate_configurator.cc +++ b/call/rtp_bitrate_configurator.cc @@ -11,7 +11,10 @@ #include "call/rtp_bitrate_configurator.h" #include +#include +#include "api/transport/bitrate_settings.h" +#include "api/units/data_rate.h" #include "rtc_base/checks.h" namespace { diff --git a/call/rtp_bitrate_configurator_unittest.cc b/call/rtp_bitrate_configurator_unittest.cc index 0b019ce8d7..a034e2c048 100644 --- a/call/rtp_bitrate_configurator_unittest.cc +++ b/call/rtp_bitrate_configurator_unittest.cc @@ -10,7 +10,9 @@ #include "call/rtp_bitrate_configurator.h" #include +#include +#include "api/transport/bitrate_settings.h" #include "test/gtest.h" namespace webrtc { diff --git a/call/rtp_config.cc b/call/rtp_config.cc index 43b615c94a..4c9afeb472 100644 --- a/call/rtp_config.cc +++ b/call/rtp_config.cc @@ -10,10 +10,17 @@ #include "call/rtp_config.h" +#include +#include #include +#include +#include +#include +#include #include "absl/algorithm/container.h" #include "api/array_view.h" +#include "api/rtp_headers.h" #include "rtc_base/checks.h" #include "rtc_base/strings/string_builder.h" diff --git a/call/rtp_demuxer.cc b/call/rtp_demuxer.cc index 525acf7b99..5dce864e7a 100644 --- a/call/rtp_demuxer.cc +++ b/call/rtp_demuxer.cc @@ -10,11 +10,19 @@ #include "call/rtp_demuxer.h" +#include +#include +#include +#include +#include + #include "absl/strings/string_view.h" #include "call/rtp_packet_sink_interface.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_header_extensions.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "rtc_base/checks.h" +#include "rtc_base/containers/flat_set.h" #include "rtc_base/logging.h" #include "rtc_base/strings/string_builder.h" diff --git a/call/rtp_demuxer.h b/call/rtp_demuxer.h index 80427b82a7..ad0103eafb 100644 --- a/call/rtp_demuxer.h +++ b/call/rtp_demuxer.h @@ -11,10 +11,10 @@ #ifndef CALL_RTP_DEMUXER_H_ #define CALL_RTP_DEMUXER_H_ +#include #include #include #include -#include #include "absl/strings/string_view.h" #include "rtc_base/containers/flat_map.h" diff --git a/call/rtp_demuxer_unittest.cc b/call/rtp_demuxer_unittest.cc index e460af8536..40a615e93c 100644 --- a/call/rtp_demuxer_unittest.cc +++ b/call/rtp_demuxer_unittest.cc @@ -10,13 +10,14 @@ #include "call/rtp_demuxer.h" +#include +#include #include #include #include #include "absl/strings/string_view.h" #include "call/test/mock_rtp_packet_sink_interface.h" -#include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "modules/rtp_rtcp/source/rtp_header_extensions.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "rtc_base/arraysize.h" diff --git a/call/rtp_payload_params.cc b/call/rtp_payload_params.cc index 00e7509bca..edcf3f74fe 100644 --- a/call/rtp_payload_params.cc +++ b/call/rtp_payload_params.cc @@ -13,16 +13,30 @@ #include #include +#include +#include #include "absl/container/inlined_vector.h" #include "absl/strings/match.h" #include "absl/types/variant.h" +#include "api/field_trials_view.h" +#include "api/transport/rtp/dependency_descriptor.h" +#include "api/video/encoded_image.h" +#include "api/video/render_resolution.h" +#include "api/video/video_codec_constants.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_frame_type.h" #include "api/video/video_timing.h" +#include "call/rtp_config.h" +#include "common_video/generic_frame_descriptor/generic_frame_info.h" +#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h" +#include "modules/rtp_rtcp/source/rtp_video_header.h" #include "modules/video_coding/codecs/h264/include/h264_globals.h" #include "modules/video_coding/codecs/interface/common_constants.h" #include "modules/video_coding/codecs/vp8/include/vp8_globals.h" #include "modules/video_coding/codecs/vp9/include/vp9_globals.h" #include "modules/video_coding/frame_dependencies_calculator.h" +#include "modules/video_coding/include/video_codec_interface.h" #include "rtc_base/arraysize.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" diff --git a/call/rtp_payload_params.h b/call/rtp_payload_params.h index 870e3c302d..ea585c0cd5 100644 --- a/call/rtp_payload_params.h +++ b/call/rtp_payload_params.h @@ -12,12 +12,17 @@ #define CALL_RTP_PAYLOAD_PARAMS_H_ #include +#include +#include #include #include #include "api/field_trials_view.h" +#include "api/transport/rtp/dependency_descriptor.h" +#include "api/video/encoded_image.h" #include "api/video_codecs/video_encoder.h" #include "call/rtp_config.h" +#include "common_video/generic_frame_descriptor/generic_frame_info.h" #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h" #include "modules/rtp_rtcp/source/rtp_video_header.h" #include "modules/video_coding/chain_diff_calculator.h" diff --git a/call/rtp_payload_params_unittest.cc b/call/rtp_payload_params_unittest.cc index 15096cf5c6..5c9f8eb426 100644 --- a/call/rtp_payload_params_unittest.cc +++ b/call/rtp_payload_params_unittest.cc @@ -10,8 +10,7 @@ #include "call/rtp_payload_params.h" -#include - +#include #include #include #include @@ -19,14 +18,22 @@ #include "absl/container/inlined_vector.h" #include "absl/types/variant.h" #include "api/transport/field_trial_based_config.h" +#include "api/transport/rtp/dependency_descriptor.h" +#include "api/video/color_space.h" +#include "api/video/encoded_image.h" +#include "api/video/video_codec_constants.h" +#include "api/video/video_codec_type.h" #include "api/video/video_content_type.h" +#include "api/video/video_frame_type.h" #include "api/video/video_rotation.h" -#include "modules/video_coding/codecs/h264/include/h264_globals.h" +#include "call/rtp_config.h" +#include "common_video/generic_frame_descriptor/generic_frame_info.h" +#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h" +#include "modules/rtp_rtcp/source/rtp_video_header.h" #include "modules/video_coding/codecs/interface/common_constants.h" #include "modules/video_coding/codecs/vp8/include/vp8_globals.h" #include "modules/video_coding/codecs/vp9/include/vp9_globals.h" #include "modules/video_coding/include/video_codec_interface.h" -#include "test/explicit_key_value_config.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/scoped_key_value_config.h" diff --git a/call/rtp_stream_receiver_controller.cc b/call/rtp_stream_receiver_controller.cc index 993a4fc76e..9367b87a57 100644 --- a/call/rtp_stream_receiver_controller.cc +++ b/call/rtp_stream_receiver_controller.cc @@ -10,8 +10,12 @@ #include "call/rtp_stream_receiver_controller.h" +#include #include +#include "api/sequence_checker.h" +#include "call/rtp_packet_sink_interface.h" +#include "call/rtp_stream_receiver_controller_interface.h" #include "rtc_base/logging.h" namespace webrtc { diff --git a/call/rtp_stream_receiver_controller.h b/call/rtp_stream_receiver_controller.h index 1040632639..1954c4fc2f 100644 --- a/call/rtp_stream_receiver_controller.h +++ b/call/rtp_stream_receiver_controller.h @@ -10,12 +10,14 @@ #ifndef CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_ #define CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_ +#include #include #include "api/sequence_checker.h" #include "call/rtp_demuxer.h" #include "call/rtp_stream_receiver_controller_interface.h" #include "modules/rtp_rtcp/include/recovered_packet_receiver.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/call/rtp_stream_receiver_controller_interface.h b/call/rtp_stream_receiver_controller_interface.h index 793d0bc145..51ae6554b9 100644 --- a/call/rtp_stream_receiver_controller_interface.h +++ b/call/rtp_stream_receiver_controller_interface.h @@ -10,6 +10,7 @@ #ifndef CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_ #define CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_ +#include #include #include "call/rtp_packet_sink_interface.h" diff --git a/call/rtp_transport_config.h b/call/rtp_transport_config.h index b310710005..5664b628eb 100644 --- a/call/rtp_transport_config.h +++ b/call/rtp_transport_config.h @@ -11,7 +11,6 @@ #ifndef CALL_RTP_TRANSPORT_CONFIG_H_ #define CALL_RTP_TRANSPORT_CONFIG_H_ -#include #include #include "api/environment/environment.h" diff --git a/call/rtp_transport_controller_send.cc b/call/rtp_transport_controller_send.cc index 34cf06d13d..9ff82f4d78 100644 --- a/call/rtp_transport_controller_send.cc +++ b/call/rtp_transport_controller_send.cc @@ -9,30 +9,53 @@ */ #include "call/rtp_transport_controller_send.h" +#include #include +#include #include #include +#include #include #include -#include "absl/strings/match.h" #include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/call/transport.h" +#include "api/fec_controller.h" +#include "api/frame_transformer_interface.h" +#include "api/rtp_packet_sender.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" #include "api/task_queue/pending_task_safety_flag.h" #include "api/task_queue/task_queue_base.h" +#include "api/transport/bandwidth_estimation_settings.h" +#include "api/transport/bitrate_settings.h" #include "api/transport/goog_cc_factory.h" +#include "api/transport/network_control.h" #include "api/transport/network_types.h" #include "api/units/data_rate.h" +#include "api/units/data_size.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" +#include "call/rtp_config.h" +#include "call/rtp_transport_config.h" +#include "call/rtp_transport_controller_send_interface.h" #include "call/rtp_video_sender.h" -#include "logging/rtc_event_log/events/rtc_event_remote_estimate.h" +#include "call/rtp_video_sender_interface.h" #include "logging/rtc_event_log/events/rtc_event_route_change.h" +#include "modules/congestion_controller/rtp/control_handler.h" +#include "modules/pacing/packet_router.h" +#include "modules/rtp_rtcp/include/report_block_data.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" -#include "modules/rtp_rtcp/source/rtp_header_extensions.h" +#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" #include "rtc_base/checks.h" +#include "rtc_base/experiments/field_trial_parser.h" #include "rtc_base/logging.h" +#include "rtc_base/network/sent_packet.h" +#include "rtc_base/network_route.h" #include "rtc_base/rate_limiter.h" +#include "rtc_base/task_utils/repeating_task.h" namespace webrtc { namespace { diff --git a/call/rtp_transport_controller_send.h b/call/rtp_transport_controller_send.h index 602feb3081..4725b38f62 100644 --- a/call/rtp_transport_controller_send.h +++ b/call/rtp_transport_controller_send.h @@ -11,22 +11,33 @@ #ifndef CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_ #define CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_ -#include +#include #include #include #include +#include #include #include #include "absl/strings/string_view.h" +#include "api/array_view.h" #include "api/environment/environment.h" -#include "api/network_state_predictor.h" +#include "api/fec_controller.h" +#include "api/frame_transformer_interface.h" +#include "api/scoped_refptr.h" #include "api/sequence_checker.h" +#include "api/task_queue/pending_task_safety_flag.h" #include "api/task_queue/task_queue_base.h" -#include "api/task_queue/task_queue_factory.h" +#include "api/transport/bandwidth_estimation_settings.h" +#include "api/transport/bitrate_settings.h" #include "api/transport/network_control.h" +#include "api/transport/network_types.h" #include "api/units/data_rate.h" +#include "api/units/data_size.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "call/rtp_bitrate_configurator.h" +#include "call/rtp_config.h" #include "call/rtp_transport_config.h" #include "call/rtp_transport_controller_send_interface.h" #include "call/rtp_video_sender.h" @@ -34,12 +45,14 @@ #include "modules/congestion_controller/rtp/transport_feedback_adapter.h" #include "modules/congestion_controller/rtp/transport_feedback_demuxer.h" #include "modules/pacing/packet_router.h" -#include "modules/pacing/rtp_packet_pacer.h" #include "modules/pacing/task_queue_paced_sender.h" +#include "modules/rtp_rtcp/include/report_block_data.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "rtc_base/experiments/field_trial_parser.h" #include "rtc_base/network_route.h" -#include "rtc_base/race_checker.h" +#include "rtc_base/rate_limiter.h" #include "rtc_base/task_utils/repeating_task.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { class FrameEncryptorInterface; diff --git a/call/rtp_transport_controller_send_factory.h b/call/rtp_transport_controller_send_factory.h index cd5a3c58ae..2a1c2c8ae7 100644 --- a/call/rtp_transport_controller_send_factory.h +++ b/call/rtp_transport_controller_send_factory.h @@ -12,10 +12,11 @@ #define CALL_RTP_TRANSPORT_CONTROLLER_SEND_FACTORY_H_ #include -#include +#include "call/rtp_transport_config.h" #include "call/rtp_transport_controller_send.h" #include "call/rtp_transport_controller_send_factory_interface.h" +#include "call/rtp_transport_controller_send_interface.h" namespace webrtc { class RtpTransportControllerSendFactory diff --git a/call/rtp_transport_controller_send_interface.h b/call/rtp_transport_controller_send_interface.h index 13eede6de0..a80dd83bc5 100644 --- a/call/rtp_transport_controller_send_interface.h +++ b/call/rtp_transport_controller_send_interface.h @@ -16,24 +16,23 @@ #include #include #include -#include -#include #include "absl/strings/string_view.h" #include "api/crypto/crypto_options.h" #include "api/fec_controller.h" #include "api/frame_transformer_interface.h" #include "api/rtp_packet_sender.h" +#include "api/scoped_refptr.h" #include "api/transport/bandwidth_estimation_settings.h" #include "api/transport/bitrate_settings.h" #include "api/transport/network_control.h" +#include "api/transport/network_types.h" #include "api/units/timestamp.h" #include "call/rtp_config.h" #include "common_video/frame_counts.h" #include "modules/rtp_rtcp/include/report_block_data.h" #include "modules/rtp_rtcp/include/rtcp_statistics.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "modules/rtp_rtcp/source/rtp_packet_received.h" namespace rtc { struct SentPacket; diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc index 431e5c0501..e97112c617 100644 --- a/call/rtp_video_sender.cc +++ b/call/rtp_video_sender.cc @@ -11,26 +11,62 @@ #include "call/rtp_video_sender.h" #include +#include +#include +#include +#include #include +#include #include #include +#include #include "absl/algorithm/container.h" #include "absl/strings/match.h" #include "absl/strings/string_view.h" #include "api/array_view.h" -#include "api/task_queue/task_queue_factory.h" -#include "api/transport/field_trial_based_config.h" +#include "api/call/bitrate_allocation.h" +#include "api/crypto/crypto_options.h" +#include "api/environment/environment.h" +#include "api/fec_controller.h" +#include "api/field_trials_view.h" +#include "api/frame_transformer_interface.h" +#include "api/rtp_headers.h" +#include "api/rtp_parameters.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" +#include "api/transport/rtp/dependency_descriptor.h" +#include "api/units/data_rate.h" +#include "api/units/data_size.h" +#include "api/units/frequency.h" #include "api/units/time_delta.h" +#include "api/video/encoded_image.h" +#include "api/video/video_bitrate_allocation.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_frame_type.h" +#include "api/video/video_layers_allocation.h" #include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_encoder.h" +#include "call/rtp_config.h" +#include "call/rtp_payload_params.h" #include "call/rtp_transport_controller_send_interface.h" +#include "common_video/frame_counts.h" +#include "common_video/generic_frame_descriptor/generic_frame_info.h" +#include "modules/include/module_fec_types.h" #include "modules/pacing/packet_router.h" +#include "modules/rtp_rtcp/include/flexfec_sender.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" #include "modules/rtp_rtcp/source/rtp_sender.h" +#include "modules/rtp_rtcp/source/rtp_sender_video.h" +#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h" +#include "modules/rtp_rtcp/source/ulpfec_generator.h" +#include "modules/rtp_rtcp/source/video_fec_generator.h" #include "modules/video_coding/include/video_codec_interface.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/numerics/safe_conversions.h" +#include "rtc_base/synchronization/mutex.h" #include "rtc_base/trace_event.h" namespace webrtc { diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h index 3cb2e9ee50..4ebd51381d 100644 --- a/call/rtp_video_sender.h +++ b/call/rtp_video_sender.h @@ -11,31 +11,43 @@ #ifndef CALL_RTP_VIDEO_SENDER_H_ #define CALL_RTP_VIDEO_SENDER_H_ +#include +#include #include #include #include -#include #include #include "api/array_view.h" +#include "api/call/bitrate_allocation.h" #include "api/call/transport.h" +#include "api/crypto/crypto_options.h" #include "api/environment/environment.h" #include "api/fec_controller.h" -#include "api/fec_controller_override.h" +#include "api/frame_transformer_interface.h" +#include "api/scoped_refptr.h" #include "api/sequence_checker.h" +#include "api/units/data_rate.h" +#include "api/units/data_size.h" +#include "api/units/frequency.h" +#include "api/video/encoded_image.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_layers_allocation.h" #include "api/video_codecs/video_encoder.h" #include "call/rtp_config.h" #include "call/rtp_payload_params.h" #include "call/rtp_transport_controller_send_interface.h" #include "call/rtp_video_sender_interface.h" -#include "modules/rtp_rtcp/include/flexfec_sender.h" +#include "common_video/frame_counts.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" #include "modules/rtp_rtcp/source/rtp_sender.h" #include "modules/rtp_rtcp/source/rtp_sender_video.h" #include "modules/rtp_rtcp/source/rtp_sequence_number_map.h" -#include "modules/rtp_rtcp/source/rtp_video_header.h" +#include "modules/rtp_rtcp/source/video_fec_generator.h" #include "rtc_base/rate_limiter.h" #include "rtc_base/synchronization/mutex.h" +#include "rtc_base/system/no_unique_address.h" #include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/call/rtp_video_sender_interface.h b/call/rtp_video_sender_interface.h index 76f7de74d1..5fe4ad37b6 100644 --- a/call/rtp_video_sender_interface.h +++ b/call/rtp_video_sender_interface.h @@ -11,18 +11,19 @@ #ifndef CALL_RTP_VIDEO_SENDER_INTERFACE_H_ #define CALL_RTP_VIDEO_SENDER_INTERFACE_H_ +#include +#include #include -#include #include #include "api/array_view.h" #include "api/call/bitrate_allocation.h" #include "api/fec_controller_override.h" #include "api/video/video_layers_allocation.h" +#include "api/video_codecs/video_encoder.h" #include "call/rtp_config.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_sequence_number_map.h" -#include "modules/video_coding/include/video_codec_interface.h" namespace webrtc { class VideoBitrateAllocation; diff --git a/call/rtp_video_sender_unittest.cc b/call/rtp_video_sender_unittest.cc index 98cd307e39..2c8e9d3707 100644 --- a/call/rtp_video_sender_unittest.cc +++ b/call/rtp_video_sender_unittest.cc @@ -10,31 +10,65 @@ #include "call/rtp_video_sender.h" -#include +#include +#include +#include #include -#include -#include +#include +#include -#include "absl/functional/any_invocable.h" +#include "api/array_view.h" +#include "api/call/bitrate_allocation.h" +#include "api/call/transport.h" +#include "api/crypto/crypto_options.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" +#include "api/frame_transformer_interface.h" +#include "api/make_ref_counted.h" +#include "api/rtp_parameters.h" +#include "api/scoped_refptr.h" #include "api/test/mock_frame_transformer.h" +#include "api/test/network_emulation/network_emulation_interfaces.h" +#include "api/transport/bitrate_settings.h" +#include "api/transport/rtp/dependency_descriptor.h" +#include "api/units/data_rate.h" +#include "api/units/data_size.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "api/video/encoded_image.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_frame_type.h" +#include "api/video_codecs/video_encoder.h" +#include "call/rtp_config.h" +#include "call/rtp_transport_config.h" #include "call/rtp_transport_controller_send.h" +#include "call/rtp_transport_controller_send_interface.h" +#include "call/video_send_stream.h" +#include "common_video/frame_counts.h" +#include "common_video/generic_frame_descriptor/generic_frame_info.h" +#include "modules/rtp_rtcp/include/report_block_data.h" +#include "modules/rtp_rtcp/include/rtcp_statistics.h" +#include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/byte_io.h" #include "modules/rtp_rtcp/source/rtcp_packet/nack.h" #include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h" #include "modules/rtp_rtcp/source/rtp_packet.h" +#include "modules/rtp_rtcp/source/rtp_sender_video.h" +#include "modules/video_coding/codecs/interface/common_constants.h" #include "modules/video_coding/fec_controller_default.h" #include "modules/video_coding/include/video_codec_interface.h" +#include "rtc_base/buffer.h" #include "rtc_base/rate_limiter.h" #include "test/explicit_key_value_config.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/mock_transport.h" #include "test/scenario/scenario.h" +#include "test/scenario/scenario_config.h" #include "test/scoped_key_value_config.h" #include "test/time_controller/simulated_time_controller.h" +#include "video/config/video_encoder_config.h" #include "video/send_statistics_proxy.h" namespace webrtc { diff --git a/call/rtx_receive_stream.cc b/call/rtx_receive_stream.cc index 6c5fa3f859..c7f5c7c0de 100644 --- a/call/rtx_receive_stream.cc +++ b/call/rtx_receive_stream.cc @@ -12,9 +12,13 @@ #include +#include +#include #include #include "api/array_view.h" +#include "api/sequence_checker.h" +#include "call/rtp_packet_sink_interface.h" #include "modules/rtp_rtcp/include/receive_statistics.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" diff --git a/call/rtx_receive_stream.h b/call/rtx_receive_stream.h index 79b03d306b..e98055ae3b 100644 --- a/call/rtx_receive_stream.h +++ b/call/rtx_receive_stream.h @@ -17,6 +17,7 @@ #include "api/sequence_checker.h" #include "call/rtp_packet_sink_interface.h" #include "rtc_base/system/no_unique_address.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/call/rtx_receive_stream_unittest.cc b/call/rtx_receive_stream_unittest.cc index b06990820f..e6aee48be4 100644 --- a/call/rtx_receive_stream_unittest.cc +++ b/call/rtx_receive_stream_unittest.cc @@ -10,8 +10,17 @@ #include "call/rtx_receive_stream.h" +#include +#include +#include +#include + +#include "api/array_view.h" +#include "api/units/timestamp.h" +#include "api/video/video_rotation.h" #include "call/test/mock_rtp_packet_sink_interface.h" #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_header_extensions.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "test/gmock.h" diff --git a/call/simulated_packet_receiver.h b/call/simulated_packet_receiver.h index e70937d7e6..629ca10f9c 100644 --- a/call/simulated_packet_receiver.h +++ b/call/simulated_packet_receiver.h @@ -11,7 +11,9 @@ #ifndef CALL_SIMULATED_PACKET_RECEIVER_H_ #define CALL_SIMULATED_PACKET_RECEIVER_H_ -#include "api/test/simulated_network.h" +#include +#include + #include "call/packet_receiver.h" namespace webrtc { diff --git a/call/video_receive_stream.cc b/call/video_receive_stream.cc index 8d88ce23c6..c03b053113 100644 --- a/call/video_receive_stream.cc +++ b/call/video_receive_stream.cc @@ -10,6 +10,14 @@ #include "call/video_receive_stream.h" +#include +#include +#include +#include + +#include "api/call/transport.h" +#include "api/rtp_headers.h" +#include "api/video_codecs/sdp_video_format.h" #include "rtc_base/strings/string_builder.h" namespace webrtc { diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h index 370bc168fd..08ac664edc 100644 --- a/call/video_receive_stream.h +++ b/call/video_receive_stream.h @@ -12,8 +12,10 @@ #define CALL_VIDEO_RECEIVE_STREAM_H_ #include +#include #include #include +#include #include #include #include @@ -21,8 +23,11 @@ #include "api/call/transport.h" #include "api/crypto/crypto_options.h" +#include "api/crypto/frame_decryptor_interface.h" +#include "api/frame_transformer_interface.h" #include "api/rtp_headers.h" -#include "api/rtp_parameters.h" +#include "api/scoped_refptr.h" +#include "api/units/time_delta.h" #include "api/video/recordable_encoded_frame.h" #include "api/video/video_content_type.h" #include "api/video/video_frame.h" @@ -34,7 +39,6 @@ #include "common_video/frame_counts.h" #include "modules/rtp_rtcp/include/rtcp_statistics.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "rtc_base/checks.h" namespace webrtc { diff --git a/call/video_send_stream.cc b/call/video_send_stream.cc index e8532a7a26..20f6cefc3e 100644 --- a/call/video_send_stream.cc +++ b/call/video_send_stream.cc @@ -10,9 +10,13 @@ #include "call/video_send_stream.h" +#include +#include #include -#include "api/crypto/frame_encryptor_interface.h" +#include "api/call/transport.h" +#include "api/video_codecs/video_encoder.h" +#include "rtc_base/checks.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/strings/string_format.h" diff --git a/call/video_send_stream.h b/call/video_send_stream.h index bd2209612e..3bf13c5e74 100644 --- a/call/video_send_stream.h +++ b/call/video_send_stream.h @@ -27,10 +27,10 @@ #include "api/scoped_refptr.h" #include "api/video/video_content_type.h" #include "api/video/video_frame.h" -#include "api/video/video_sink_interface.h" #include "api/video/video_source_interface.h" #include "api/video/video_stream_encoder_settings.h" #include "api/video_codecs/scalability_mode.h" +#include "api/video_codecs/video_encoder_factory.h" #include "call/rtp_config.h" #include "common_video/frame_counts.h" #include "common_video/include/quality_limitation_reason.h" diff --git a/test/scenario/BUILD.gn b/test/scenario/BUILD.gn index 8f9234aaf5..f055bbdd93 100644 --- a/test/scenario/BUILD.gn +++ b/test/scenario/BUILD.gn @@ -96,6 +96,7 @@ if (rtc_include_tests && !build_with_chromium) { "../../api/rtc_event_log", "../../api/rtc_event_log:rtc_event_log_factory", "../../api/task_queue", + "../../api/test/network_emulation", "../../api/test/video:function_video_factory", "../../api/transport:network_control", "../../api/units:data_rate", @@ -133,7 +134,9 @@ if (rtc_include_tests && !build_with_chromium) { "../../modules/video_coding/svc:scalability_mode_util", "../../rtc_base:checks", "../../rtc_base:copy_on_write_buffer", + "../../rtc_base:macromagic", "../../rtc_base:net_helper", + "../../rtc_base:network_route", "../../rtc_base:refcount", "../../rtc_base:rtc_base_tests_utils", "../../rtc_base:rtc_event", diff --git a/test/scenario/network_node.h b/test/scenario/network_node.h index 57d683c038..21ee5a1667 100644 --- a/test/scenario/network_node.h +++ b/test/scenario/network_node.h @@ -10,18 +10,22 @@ #ifndef TEST_SCENARIO_NETWORK_NODE_H_ #define TEST_SCENARIO_NETWORK_NODE_H_ -#include -#include +#include +#include #include -#include -#include +#include "api/array_view.h" #include "api/call/transport.h" #include "api/sequence_checker.h" +#include "api/test/network_emulation/network_emulation_interfaces.h" +#include "api/units/data_size.h" #include "api/units/timestamp.h" #include "call/call.h" -#include "rtc_base/copy_on_write_buffer.h" +#include "rtc_base/network_route.h" +#include "rtc_base/socket_address.h" #include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread_annotations.h" +#include "system_wrappers/include/clock.h" #include "test/network/network_emulation.h" #include "test/network/simulated_network.h" #include "test/scenario/column_printer.h"