blob: 144987fec73b7c1542e3f90d6c3017e43d452b2b [file] [log] [blame]
# Copyright (c) 2014 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_static_library("video") {
sources = [
"call_stats.cc",
"call_stats.h",
"encoder_rtcp_feedback.cc",
"encoder_rtcp_feedback.h",
"overuse_frame_detector.cc",
"overuse_frame_detector.h",
"payload_router.cc",
"payload_router.h",
"quality_threshold.cc",
"quality_threshold.h",
"receive_statistics_proxy.cc",
"receive_statistics_proxy.h",
"report_block_stats.cc",
"report_block_stats.h",
"rtp_streams_synchronizer.cc",
"rtp_streams_synchronizer.h",
"rtp_video_stream_receiver.cc",
"rtp_video_stream_receiver.h",
"send_delay_stats.cc",
"send_delay_stats.h",
"send_statistics_proxy.cc",
"send_statistics_proxy.h",
"stats_counter.cc",
"stats_counter.h",
"stream_synchronization.cc",
"stream_synchronization.h",
"transport_adapter.cc",
"transport_adapter.h",
"video_receive_stream.cc",
"video_receive_stream.h",
"video_send_stream.cc",
"video_send_stream.h",
"video_stream_decoder.cc",
"video_stream_decoder.h",
"video_stream_encoder.cc",
"video_stream_encoder.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 = [
"..:webrtc_common",
"../:typedefs",
"../api:fec_controller_api",
"../api:libjingle_peerconnection_api",
"../api:optional",
"../api:transport_api",
"../api:video_frame_api",
"../api:video_frame_api_i420",
"../api/video_codecs:video_codecs_api",
"../call:bitrate_allocator",
"../call:call_interfaces",
"../call:rtp_interfaces",
"../call:video_stream_api",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/video_coding:video_codec_interface",
"../rtc_base:checks",
"../rtc_base/experiments:alr_experiment",
"../rtc_base/system:fallthrough",
"../system_wrappers:field_trial_api",
"../system_wrappers:metrics_api",
# For RtxReceiveStream.
"../call:rtp_receiver",
"../common_video",
"../logging:rtc_event_log_api",
"../modules:module_api",
"../modules/bitrate_controller",
"../modules/congestion_controller",
"../modules/pacing",
"../modules/remote_bitrate_estimator",
"../modules/rtp_rtcp",
"../modules/utility",
"../modules/video_coding",
"../modules/video_coding:video_coding_utility",
"../modules/video_processing",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_numerics",
"../rtc_base:rtc_task_queue",
"../rtc_base:sequenced_task_checker",
"../rtc_base:weak_ptr",
"../system_wrappers",
]
if (!build_with_mozilla) {
deps += [ "../media:rtc_media_base" ]
}
}
if (rtc_include_tests) {
rtc_source_set("video_quality_test") {
testonly = true
visibility = [ ":*" ] # Only targets in this file can depend on this.
sources = [
"video_quality_test.cc",
"video_quality_test.h",
]
deps = [
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_output",
"../media:rtc_audio_video",
"../media:rtc_internal_video_codecs",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/rtp_rtcp",
"../modules/video_coding:webrtc_h264",
"../modules/video_coding:webrtc_vp8",
"../modules/video_coding:webrtc_vp9",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../system_wrappers",
"../test:perf_test",
"../test:rtp_test_utils",
"../test:test_common",
"../test:test_renderer",
"../test:test_support",
"../test:test_support_test_artifacts",
"../test:video_test_common",
"../test:video_test_support",
]
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("video_full_stack_tests") {
testonly = true
sources = [
"full_stack_tests.cc",
]
deps = [
":video_quality_test",
"../modules/pacing:pacing",
"../rtc_base/experiments:alr_experiment",
"../test:field_trial",
"../test:test_common",
"../test:test_support",
"//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" ]
}
if (rtc_use_h264) {
defines = [ "WEBRTC_USE_H264" ]
}
}
rtc_executable("video_loopback") {
testonly = true
sources = [
"video_loopback.cc",
]
deps = [
":video_quality_test",
"../rtc_base:rtc_base_approved",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
"../system_wrappers:runtime_enabled_features_default",
"../test:field_trial",
"../test:run_test",
"../test:run_test_interface",
"../test:test_common",
"../test:test_renderer",
"../test:test_support",
"//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_executable("screenshare_loopback") {
testonly = true
sources = [
"screenshare_loopback.cc",
]
deps = [
":video_quality_test",
"../rtc_base:rtc_base_approved",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
"../system_wrappers:runtime_enabled_features_default",
"../test:field_trial",
"../test:run_test",
"../test:run_test_interface",
"../test:test_common",
"../test:test_renderer",
"../test:test_support",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from Chrome's Clang plugins.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
rtc_executable("sv_loopback") {
testonly = true
sources = [
"sv_loopback.cc",
]
deps = [
":video_quality_test",
"../rtc_base:rtc_base_approved",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
"../system_wrappers:runtime_enabled_features_default",
"../test:field_trial",
"../test:run_test",
"../test:run_test_interface",
"../test:test_common",
"../test:test_renderer",
"../test:test_support",
"//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_executable("video_replay") {
testonly = true
sources = [
"replay.cc",
]
deps = [
"..:webrtc_common",
"../:typedefs",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../common_video",
"../logging:rtc_event_log_api",
"../modules/rtp_rtcp",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"../system_wrappers:metrics_default",
"../system_wrappers:runtime_enabled_features_default",
"../test:field_trial",
"../test:rtp_test_utils",
"../test:run_test",
"../test:run_test_interface",
"../test:test_common",
"../test:test_renderer",
"../test:test_support",
"../test:video_test_common",
"../test:video_test_support",
]
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(pbos): Rename test suite.
rtc_source_set("video_tests") {
testonly = true
defines = []
sources = [
"call_stats_unittest.cc",
"encoder_rtcp_feedback_unittest.cc",
"end_to_end_tests/bandwidth_tests.cc",
"end_to_end_tests/call_operation_tests.cc",
"end_to_end_tests/codec_tests.cc",
"end_to_end_tests/config_tests.cc",
"end_to_end_tests/extended_reports_tests.cc",
"end_to_end_tests/fec_tests.cc",
"end_to_end_tests/histogram_tests.cc",
"end_to_end_tests/log_tests.cc",
"end_to_end_tests/multi_stream_tester.cc",
"end_to_end_tests/multi_stream_tester.h",
"end_to_end_tests/multi_stream_tests.cc",
"end_to_end_tests/network_state_tests.cc",
"end_to_end_tests/probing_tests.cc",
"end_to_end_tests/retransmission_tests.cc",
"end_to_end_tests/rtp_rtcp_tests.cc",
"end_to_end_tests/ssrc_tests.cc",
"end_to_end_tests/stats_tests.cc",
"end_to_end_tests/transport_feedback_tests.cc",
"overuse_frame_detector_unittest.cc",
"payload_router_unittest.cc",
"picture_id_tests.cc",
"quality_threshold_unittest.cc",
"receive_statistics_proxy_unittest.cc",
"report_block_stats_unittest.cc",
"rtp_video_stream_receiver_unittest.cc",
"send_delay_stats_unittest.cc",
"send_statistics_proxy_unittest.cc",
"stats_counter_unittest.cc",
"stream_synchronization_unittest.cc",
"video_receive_stream_unittest.cc",
"video_send_stream_tests.cc",
"video_stream_encoder_unittest.cc",
]
deps = [
":video",
"../api:optional",
"../api:video_frame_api",
"../api:video_frame_api_i420",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../call:mock_rtp_interfaces",
"../call:rtp_receiver",
"../call:rtp_sender",
"../call:video_stream_api",
"../common_video",
"../logging:rtc_event_log_api",
"../media:rtc_audio_video",
"../media:rtc_internal_video_codecs",
"../media:rtc_media",
"../media:rtc_media_base",
"../media:rtc_media_tests_utils",
"../modules:module_api",
"../modules/pacing",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:mock_rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/utility",
"../modules/video_coding",
"../modules/video_coding:video_codec_interface",
"../modules/video_coding:video_coding_utility",
"../modules/video_coding:webrtc_h264",
"../modules/video_coding:webrtc_vp8_helpers",
"../modules/video_coding:webrtc_vp9",
"../rtc_base:checks",
"../rtc_base:rate_limiter",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:rtc_numerics",
"../rtc_base/experiments:alr_experiment",
"../system_wrappers",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_api",
"../system_wrappers:metrics_default",
"../test:direct_transport",
"../test:field_trial",
"../test:perf_test",
"../test:rtp_test_utils",
"../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" ]
}
if (rtc_use_h264) {
defines += [ "WEBRTC_USE_H264" ]
}
if (!build_with_mozilla) {
deps += [ "../media:rtc_media_base" ]
}
}
}