| # Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| # |
| # Use of this source code is governed by a BSD-style license |
| # that can be found in the LICENSE file in the root of the source |
| # tree. An additional intellectual property rights grant can be found |
| # in the file PATENTS. All contributing project authors may |
| # be found in the AUTHORS file in the root of the source tree. |
| |
| import("../webrtc.gni") |
| |
| rtc_source_set("call_interfaces") { |
| sources = [ |
| "audio_receive_stream.cc", |
| "audio_receive_stream.h", |
| "audio_send_stream.h", |
| "audio_state.cc", |
| "audio_state.h", |
| "call.h", |
| "call_config.cc", |
| "call_config.h", |
| "flexfec_receive_stream.cc", |
| "flexfec_receive_stream.h", |
| "syncable.cc", |
| "syncable.h", |
| ] |
| if (!build_with_mozilla) { |
| sources += [ "audio_send_stream.cc" ] |
| } |
| deps = [ |
| ":rtp_interfaces", |
| ":video_stream_api", |
| "..:webrtc_common", |
| "../:typedefs", |
| "../api:fec_controller_api", |
| "../api:libjingle_peerconnection_api", |
| "../api:optional", |
| "../api:transport_api", |
| "../api/audio:audio_mixer_api", |
| "../api/audio_codecs:audio_codecs_api", |
| "../modules/audio_device:audio_device", |
| "../modules/audio_processing:audio_processing", |
| "../modules/audio_processing:audio_processing_statistics", |
| "../rtc_base:audio_format_to_string", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_base_approved", |
| ] |
| } |
| |
| # TODO(nisse): These RTP targets should be moved elsewhere |
| # when interfaces have stabilized. See also TODO for |mock_rtp_interfaces|. |
| rtc_source_set("rtp_interfaces") { |
| sources = [ |
| "bitrate_constraints.cc", |
| "bitrate_constraints.h", |
| "rtcp_packet_sink_interface.h", |
| "rtp_config.cc", |
| "rtp_config.h", |
| "rtp_packet_sink_interface.h", |
| "rtp_stream_receiver_controller_interface.h", |
| "rtp_transport_controller_send_interface.h", |
| ] |
| deps = [ |
| "../api:array_view", |
| "../api:optional", |
| "../rtc_base:rtc_base_approved", |
| ] |
| } |
| |
| rtc_source_set("rtp_receiver") { |
| visibility = [ "*" ] |
| sources = [ |
| "rtcp_demuxer.cc", |
| "rtcp_demuxer.h", |
| "rtp_demuxer.cc", |
| "rtp_demuxer.h", |
| "rtp_rtcp_demuxer_helper.cc", |
| "rtp_rtcp_demuxer_helper.h", |
| "rtp_stream_receiver_controller.cc", |
| "rtp_stream_receiver_controller.h", |
| "rtx_receive_stream.cc", |
| "rtx_receive_stream.h", |
| "ssrc_binding_observer.h", |
| ] |
| deps = [ |
| ":rtp_interfaces", |
| "..:webrtc_common", |
| "../api:array_view", |
| "../api:libjingle_peerconnection_api", |
| "../api:optional", |
| "../modules/rtp_rtcp", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| ] |
| } |
| |
| rtc_source_set("rtp_sender") { |
| sources = [ |
| "rtp_transport_controller_send.cc", |
| "rtp_transport_controller_send.h", |
| ] |
| deps = [ |
| ":bitrate_configurator", |
| ":rtp_interfaces", |
| "..:webrtc_common", |
| "../modules/congestion_controller", |
| "../modules/congestion_controller/network_control", |
| "../modules/congestion_controller/rtp:congestion_controller", |
| "../modules/pacing", |
| "../modules/utility", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_base_approved", |
| "../system_wrappers:field_trial_api", |
| ] |
| } |
| |
| rtc_source_set("bitrate_configurator") { |
| sources = [ |
| "rtp_bitrate_configurator.cc", |
| "rtp_bitrate_configurator.h", |
| ] |
| deps = [ |
| ":rtp_interfaces", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| ] |
| } |
| |
| rtc_source_set("bitrate_allocator") { |
| sources = [ |
| "bitrate_allocator.cc", |
| "bitrate_allocator.h", |
| ] |
| deps = [ |
| "../modules/bitrate_controller", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:sequenced_task_checker", |
| "../system_wrappers", |
| "../system_wrappers:field_trial_api", |
| "../system_wrappers:metrics_api", |
| ] |
| if (!build_with_chromium && is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| } |
| } |
| |
| rtc_static_library("call") { |
| sources = [ |
| "call.cc", |
| "callfactory.cc", |
| "callfactory.h", |
| "degraded_call.cc", |
| "degraded_call.h", |
| "flexfec_receive_stream_impl.cc", |
| "flexfec_receive_stream_impl.h", |
| "receive_time_calculator.cc", |
| "receive_time_calculator.h", |
| ] |
| |
| if (!build_with_chromium && is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| } |
| |
| deps = [ |
| ":bitrate_allocator", |
| ":call_interfaces", |
| ":fake_network", |
| ":rtp_interfaces", |
| ":rtp_receiver", |
| ":rtp_sender", |
| ":video_stream_api", |
| "..:webrtc_common", |
| "../api:callfactory_api", |
| "../api:optional", |
| "../api:transport_api", |
| "../audio", |
| "../logging:rtc_event_audio", |
| "../logging:rtc_event_log_api", |
| "../logging:rtc_event_rtp_rtcp", |
| "../logging:rtc_event_video", |
| "../logging:rtc_stream_config", |
| "../modules/bitrate_controller", |
| "../modules/congestion_controller", |
| "../modules/congestion_controller/network_control", |
| "../modules/pacing", |
| "../modules/rtp_rtcp", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../modules/utility", |
| "../modules/video_coding:video_coding", |
| "../rtc_base:checks", |
| "../rtc_base:rate_limiter", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base:safe_minmax", |
| "../rtc_base:sequenced_task_checker", |
| "../rtc_base/synchronization:rw_lock_wrapper", |
| "../system_wrappers", |
| "../system_wrappers:field_trial_api", |
| "../system_wrappers:metrics_api", |
| "../video", |
| ] |
| } |
| |
| rtc_source_set("video_stream_api") { |
| sources = [ |
| "video_config.cc", |
| "video_config.h", |
| "video_receive_stream.cc", |
| "video_receive_stream.h", |
| "video_send_stream.cc", |
| "video_send_stream.h", |
| ] |
| deps = [ |
| ":rtp_interfaces", |
| "../:typedefs", |
| "../:webrtc_common", |
| "../api:libjingle_peerconnection_api", |
| "../api:optional", |
| "../api:transport_api", |
| "../api:video_frame_api", |
| "../api/video_codecs:video_codecs_api", |
| "../common_video:common_video", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| ] |
| } |
| |
| rtc_source_set("fake_network") { |
| sources = [ |
| "fake_network_pipe.cc", |
| "fake_network_pipe.h", |
| ] |
| deps = [ |
| ":call_interfaces", |
| "..:typedefs", |
| "..:webrtc_common", |
| "../api:transport_api", |
| "../modules:module_api", |
| "../modules/rtp_rtcp", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:sequenced_task_checker", |
| "../system_wrappers", |
| ] |
| |
| if (!build_with_chromium && is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| } |
| } |
| |
| if (rtc_include_tests) { |
| rtc_source_set("call_tests") { |
| testonly = true |
| |
| sources = [ |
| "bitrate_allocator_unittest.cc", |
| "bitrate_estimator_tests.cc", |
| "call_unittest.cc", |
| "flexfec_receive_stream_unittest.cc", |
| "receive_time_calculator_unittest.cc", |
| "rtcp_demuxer_unittest.cc", |
| "rtp_bitrate_configurator_unittest.cc", |
| "rtp_demuxer_unittest.cc", |
| "rtp_rtcp_demuxer_helper_unittest.cc", |
| "rtx_receive_stream_unittest.cc", |
| ] |
| deps = [ |
| ":bitrate_allocator", |
| ":bitrate_configurator", |
| ":call", |
| ":call_interfaces", |
| ":mock_rtp_interfaces", |
| ":rtp_interfaces", |
| ":rtp_receiver", |
| ":rtp_sender", |
| "..:webrtc_common", |
| "../api:array_view", |
| "../api:libjingle_peerconnection_api", |
| "../api:mock_audio_mixer", |
| "../api/audio_codecs:builtin_audio_decoder_factory", |
| "../audio:audio", |
| "../logging:rtc_event_log_api", |
| "../logging:rtc_event_log_impl_base", |
| "../modules/audio_device:mock_audio_device", |
| "../modules/audio_mixer", |
| "../modules/audio_mixer:audio_mixer_impl", |
| "../modules/audio_processing:mocks", |
| "../modules/bitrate_controller", |
| "../modules/congestion_controller", |
| "../modules/pacing", |
| "../modules/pacing:mock_paced_sender", |
| "../modules/rtp_rtcp", |
| "../modules/rtp_rtcp:mock_rtp_rtcp", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../modules/utility:mock_process_thread", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../system_wrappers", |
| "../test:audio_codec_mocks", |
| "../test:direct_transport", |
| "../test:test_common", |
| "../test:test_support", |
| "../test:video_test_common", |
| "//testing/gtest", |
| ] |
| if (!build_with_chromium && is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| } |
| } |
| |
| rtc_source_set("call_perf_tests") { |
| testonly = true |
| |
| sources = [ |
| "call_perf_tests.cc", |
| "rampup_tests.cc", |
| "rampup_tests.h", |
| ] |
| deps = [ |
| ":call_interfaces", |
| ":video_stream_api", |
| "..:webrtc_common", |
| "../api/audio_codecs:builtin_audio_encoder_factory", |
| "../logging:rtc_event_log_api", |
| "../modules/audio_coding", |
| "../modules/audio_device", |
| "../modules/audio_device:audio_device_impl", |
| "../modules/audio_mixer:audio_mixer_impl", |
| "../modules/rtp_rtcp", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../system_wrappers", |
| "../system_wrappers:metrics_default", |
| "../system_wrappers:runtime_enabled_features_default", |
| "../test:direct_transport", |
| "../test:field_trial", |
| "../test:fileutils", |
| "../test:perf_test", |
| "../test:test_common", |
| "../test:test_support", |
| "../test:video_test_common", |
| "../video", |
| "//testing/gtest", |
| ] |
| if (!build_with_chromium && is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| } |
| } |
| |
| # TODO(eladalon): This should be moved, as with the TODO for |rtp_interfaces|. |
| rtc_source_set("mock_rtp_interfaces") { |
| testonly = true |
| |
| sources = [ |
| "test/mock_rtp_packet_sink_interface.h", |
| "test/mock_rtp_transport_controller_send.h", |
| ] |
| deps = [ |
| ":rtp_interfaces", |
| "../modules/congestion_controller", |
| "../modules/pacing", |
| "../rtc_base:rate_limiter", |
| "../rtc_base:rtc_base", |
| "../test:test_support", |
| ] |
| } |
| |
| rtc_source_set("mock_call_interfaces") { |
| testonly = true |
| |
| sources = [ |
| "test/mock_audio_send_stream.h", |
| ] |
| deps = [ |
| ":call_interfaces", |
| "//test:test_support", |
| ] |
| } |
| |
| rtc_test("fake_network_unittests") { |
| deps = [ |
| ":call_interfaces", |
| ":fake_network", |
| "../modules/rtp_rtcp", |
| "../rtc_base:rtc_base_approved", |
| "../system_wrappers", |
| "../test:test_common", |
| "../test:test_main", |
| "//testing/gtest", |
| ] |
| sources = [ |
| "test/fake_network_pipe_unittest.cc", |
| ] |
| if (!build_with_chromium && is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| } |
| } |
| } |
