blob: 903df3ba7878d52ad8beaa9eeba201295348fdd8 [file] [log] [blame]
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../webrtc.gni")
rtc_source_set("call_interfaces") {
sources = [
"audio_receive_stream.h",
"audio_send_stream.cc",
"audio_send_stream.h",
"audio_state.h",
"call.h",
"callfactoryinterface.h",
"flexfec_receive_stream.h",
"syncable.cc",
"syncable.h",
]
deps = [
":rtp_interfaces",
":video_stream_api",
"..:webrtc_common",
"../:typedefs",
"../api:audio_mixer_api",
"../api:libjingle_peerconnection_api",
"../api:optional",
"../api:transport_api",
"../api/audio_codecs:audio_codecs_api",
"../modules/audio_processing:audio_processing_statistics",
"../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 = [
"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",
"../rtc_base:rtc_base_approved",
]
}
rtc_source_set("rtp_receiver") {
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 = [
":rtp_interfaces",
"..:webrtc_common",
"../modules/congestion_controller",
"../modules/pacing",
"../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: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",
"flexfec_receive_stream_impl.cc",
"flexfec_receive_stream_impl.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",
":rtp_interfaces",
":rtp_receiver",
":rtp_sender",
":video_stream_api",
"..:webrtc_common",
"../api:optional",
"../api:transport_api",
"../audio",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl",
"../modules/bitrate_controller",
"../modules/congestion_controller",
"../modules/pacing",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/utility",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../rtc_base:sequenced_task_checker",
"../system_wrappers",
"../system_wrappers:metrics_api",
"../video",
]
}
rtc_source_set("video_stream_api") {
# TODO(bugs.webrtc.org/6828): Remove dependency cycle:
# //call:video_stream_api ->
# //media:rtc_media_base ->
# //call:call_interfaces ->
# //call:video_stream_api
check_includes = false
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",
"../:webrtc_common",
"../api:libjingle_peerconnection_api",
"../api:optional",
"../api:transport_api",
"../common_video:common_video",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:rtc_base_approved",
]
}
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",
"rtcp_demuxer_unittest.cc",
"rtp_demuxer_unittest.cc",
"rtp_rtcp_demuxer_helper_unittest.cc",
"rtx_receive_stream_unittest.cc",
]
deps = [
":bitrate_allocator",
":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",
"../logging:rtc_event_log_api",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/bitrate_controller",
"../modules/congestion_controller",
"../modules/congestion_controller:mock_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/gmock",
"//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_mixer:audio_mixer_impl",
"../modules/rtp_rtcp",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"../system_wrappers:metrics_default",
"../test:direct_transport",
"../test:fake_audio_device",
"../test:field_trial",
"../test:test_common",
"../test:test_support",
"../test:video_test_common",
"../video",
"../voice_engine",
"//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 = [
"fake_rtp_transport_controller_send.h",
"test/mock_rtp_packet_sink_interface.h",
]
deps = [
":rtp_interfaces",
"..:webrtc_common",
"../modules/congestion_controller:congestion_controller",
"../modules/pacing:pacing",
"../test:test_support",
"//testing/gmock",
]
}
rtc_source_set("mock_call_interfaces") {
testonly = true
sources = [
"test/mock_audio_send_stream.h",
]
deps = [
":call_interfaces",
"//test:test_support",
]
}
}