|  | # 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: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/neteq:neteq_api", | 
|  | "../api/task_queue", | 
|  | "../api/transport:bitrate_settings", | 
|  | "../api/transport:network_control", | 
|  | "../api/transport:webrtc_key_value_config", | 
|  | "../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:rtc_base_approved", | 
|  | "../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: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/transport:webrtc_key_value_config", | 
|  | "../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", | 
|  | ] | 
|  | 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:rtc_base_approved", | 
|  | "../rtc_base/containers:flat_map", | 
|  | "../rtc_base/containers:flat_set", | 
|  | ] | 
|  | absl_deps = [ "//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: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/transport:webrtc_key_value_config", | 
|  | "../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:rate_limiter", | 
|  | "../rtc_base:rtc_base_approved", | 
|  | "../rtc_base:rtc_task_queue", | 
|  | "../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: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: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:rate_limiter", | 
|  | "../rtc_base:rtc_base_approved", | 
|  | "../rtc_base:rtc_task_queue", | 
|  | "../rtc_base:safe_minmax", | 
|  | "../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", | 
|  | "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", | 
|  | ] | 
|  | 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: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: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:rate_limiter", | 
|  | "../rtc_base:rtc_base_approved", | 
|  | "../rtc_base:task_queue_for_test", | 
|  | "../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: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/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:rtc_base_approved", | 
|  | "../rtc_base:task_queue_for_test", | 
|  | "../rtc_base:threading", | 
|  | "../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" ] | 
|  | } | 
|  | } |