mirror of
https://github.com/klzgrad/naiveproxy.git
synced 2024-11-28 08:16:09 +03:00
282 lines
7.5 KiB
Plaintext
282 lines
7.5 KiB
Plaintext
# 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",
|
|
"../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:optional",
|
|
"../modules/rtp_rtcp",
|
|
"../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: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" ]
|
|
}
|
|
}
|
|
|
|
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" ]
|
|
}
|
|
|
|
public_deps = [
|
|
":call_interfaces",
|
|
"../api:call_api",
|
|
"../api:libjingle_peerconnection_api",
|
|
]
|
|
|
|
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/utility",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../rtc_base:sequenced_task_checker",
|
|
"../system_wrappers",
|
|
"../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",
|
|
"../:webrtc_common",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:optional",
|
|
"../api:transport_api",
|
|
"../common_video:common_video",
|
|
"../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",
|
|
":mock_rtp_interfaces",
|
|
":rtp_interfaces",
|
|
":rtp_receiver",
|
|
":rtp_sender",
|
|
"..:webrtc_common",
|
|
"../api:array_view",
|
|
"../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/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/utility:mock_process_thread",
|
|
"../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: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",
|
|
]
|
|
}
|
|
}
|