webrtc_m130/logging/BUILD.gn

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# Copyright (c) 2016 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.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../webrtc.gni")
import("//third_party/protobuf/proto_library.gni")
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
group("logging") {
public_deps = [
":rtc_event_log_impl",
]
if (rtc_enable_protobuf) {
public_deps += [ ":rtc_event_log_parser" ]
}
}
rtc_source_set("rtc_event_log_api") {
sources = [
"rtc_event_log/events/rtc_event.h",
"rtc_event_log/events/rtc_event_audio_network_adaptation.cc",
"rtc_event_log/events/rtc_event_audio_network_adaptation.h",
"rtc_event_log/events/rtc_event_audio_playout.cc",
"rtc_event_log/events/rtc_event_audio_playout.h",
"rtc_event_log/events/rtc_event_audio_receive_stream_config.cc",
"rtc_event_log/events/rtc_event_audio_receive_stream_config.h",
"rtc_event_log/events/rtc_event_audio_send_stream_config.cc",
"rtc_event_log/events/rtc_event_audio_send_stream_config.h",
"rtc_event_log/events/rtc_event_bwe_update_delay_based.cc",
"rtc_event_log/events/rtc_event_bwe_update_delay_based.h",
"rtc_event_log/events/rtc_event_bwe_update_loss_based.cc",
"rtc_event_log/events/rtc_event_bwe_update_loss_based.h",
"rtc_event_log/events/rtc_event_logging_started.cc",
"rtc_event_log/events/rtc_event_logging_started.h",
"rtc_event_log/events/rtc_event_logging_stopped.cc",
"rtc_event_log/events/rtc_event_logging_stopped.h",
"rtc_event_log/events/rtc_event_probe_cluster_created.cc",
"rtc_event_log/events/rtc_event_probe_cluster_created.h",
"rtc_event_log/events/rtc_event_probe_result_failure.cc",
"rtc_event_log/events/rtc_event_probe_result_failure.h",
"rtc_event_log/events/rtc_event_probe_result_success.cc",
"rtc_event_log/events/rtc_event_probe_result_success.h",
"rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc",
"rtc_event_log/events/rtc_event_rtcp_packet_incoming.h",
"rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc",
"rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h",
"rtc_event_log/events/rtc_event_rtp_packet_incoming.cc",
"rtc_event_log/events/rtc_event_rtp_packet_incoming.h",
"rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc",
"rtc_event_log/events/rtc_event_rtp_packet_outgoing.h",
"rtc_event_log/events/rtc_event_video_receive_stream_config.cc",
"rtc_event_log/events/rtc_event_video_receive_stream_config.h",
"rtc_event_log/events/rtc_event_video_send_stream_config.cc",
"rtc_event_log/events/rtc_event_video_send_stream_config.h",
"rtc_event_log/rtc_event_log.h",
"rtc_event_log/rtc_event_log_factory_interface.h",
"rtc_event_log/rtc_stream_config.cc",
"rtc_event_log/rtc_stream_config.h",
]
deps = [
"..:webrtc_common",
"../api:array_view",
"../api:libjingle_peerconnection_api",
"../call:video_stream_api",
"../modules/audio_coding:audio_network_adaptor_config",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:rtc_base_approved",
]
}
rtc_static_library("rtc_event_log_impl") {
sources = [
"rtc_event_log/encoder/rtc_event_log_encoder.h",
"rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc",
"rtc_event_log/encoder/rtc_event_log_encoder_legacy.h",
"rtc_event_log/rtc_event_log.cc",
"rtc_event_log/rtc_event_log_factory.cc",
"rtc_event_log/rtc_event_log_factory.h",
]
defines = []
deps = [
":rtc_event_log_api",
"..:webrtc_common",
"../modules/audio_coding:audio_network_adaptor",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp",
"../rtc_base:protobuf_utils",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../rtc_base:sequenced_task_checker",
"../system_wrappers",
]
if (rtc_enable_protobuf) {
defines += [ "ENABLE_RTC_EVENT_LOG" ]
deps += [ ":rtc_event_log_proto" ]
}
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_enable_protobuf) {
proto_library("rtc_event_log_proto") {
sources = [
"rtc_event_log/rtc_event_log.proto",
]
proto_out_dir = "logging/rtc_event_log"
}
rtc_static_library("rtc_event_log_parser") {
sources = [
"rtc_event_log/rtc_event_log_parser.cc",
"rtc_event_log/rtc_event_log_parser.h",
]
public_deps = [
":rtc_event_log_api",
":rtc_event_log_proto",
"..:webrtc_common",
"../modules/audio_coding:audio_network_adaptor",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp:rtp_rtcp",
"../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" ]
}
deps = [
"../call:video_stream_api",
"../rtc_base:protobuf_utils",
"../rtc_base:rtc_base_approved",
]
}
if (rtc_include_tests) {
rtc_source_set("rtc_event_log_tests") {
testonly = true
assert(rtc_enable_protobuf)
defines = [ "ENABLE_RTC_EVENT_LOG" ]
sources = [
"rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc",
"rtc_event_log/rtc_event_log_unittest.cc",
"rtc_event_log/rtc_event_log_unittest_helper.cc",
"rtc_event_log/rtc_event_log_unittest_helper.h",
]
deps = [
":rtc_event_log_impl",
":rtc_event_log_parser",
"../call",
"../modules/audio_coding:audio_network_adaptor",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../system_wrappers:metrics_default",
"../test:test_support",
"//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_test("rtc_event_log2rtp_dump") {
testonly = true
sources = [
"rtc_event_log/rtc_event_log2rtp_dump.cc",
]
deps = [
":rtc_event_log_api",
":rtc_event_log_impl",
":rtc_event_log_parser",
"../modules/rtp_rtcp:rtp_rtcp",
"../rtc_base:rtc_base_approved",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
"../test:rtp_test_utils",
]
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_executable("rtc_event_log2text") {
testonly = true
sources = [
"rtc_event_log/rtc_event_log2text.cc",
]
deps = [
":rtc_event_log_api",
":rtc_event_log_impl",
":rtc_event_log_parser",
"../call:video_stream_api",
"../rtc_base:rtc_base_approved",
# TODO(kwiberg): Remove this dependency.
"../api/audio_codecs:audio_codecs_api",
"../modules/audio_coding:audio_network_adaptor_config",
"../modules/rtp_rtcp:rtp_rtcp",
]
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_executable("rtc_event_log2stats") {
testonly = true
sources = [
"rtc_event_log/rtc_event_log2stats.cc",
]
deps = [
":rtc_event_log_api",
":rtc_event_log_impl",
":rtc_event_log_proto",
"../rtc_base:rtc_base_approved",
]
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" ]
}
}
}
}