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webrtc/logging/BUILD.gn
Per Kjellander a7f2d84ad1 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" 
This reverts commit c73e1f4378.

Reason for revert: 
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660

Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
> 
> This reverts commit 588c548657.
> 
> Reason for revert: 
> 
> Breaks Chrome FYI:
> 
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
>   -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
>     static_library(target_name) {
>     ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
>   //third_party/webrtc/*
>   //third_party/webrtc_overrides/*
> ]
> 
>  https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
> 
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> > 
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> > 
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> > 
> > BUG=webrtc:8254
> > 
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
> 
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org

Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:55:04 +00:00

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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.
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")
}
group("logging") {
deps = [
":rtc_event_log_impl",
]
if (rtc_enable_protobuf) {
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_alr_state.cc",
"rtc_event_log/events/rtc_event_alr_state.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_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/output/rtc_event_log_output_file.cc",
"rtc_event_log/output/rtc_event_log_output_file.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",
"../:typedefs",
"../api:array_view",
"../api:libjingle_logging_api",
"../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:checks",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
]
# TODO(eladalon): Remove this.
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("rtc_event_log_impl") {
visibility = [ "*" ]
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:rtp_rtcp_format",
"../rtc_base:checks",
"../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" ]
}
# TODO(eladalon): Remove this.
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"
}
proto_library("rtc_event_log2_proto") {
sources = [
"rtc_event_log/rtc_event_log2.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",
]
deps = [
":rtc_event_log2_proto",
":rtc_event_log_api",
":rtc_event_log_proto",
"..:webrtc_common",
"../call:video_stream_api",
"../modules/audio_coding:audio_network_adaptor",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:protobuf_utils",
"../rtc_base:rtc_base_approved",
"../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("rtc_event_log_tests") {
testonly = true
assert(rtc_enable_protobuf)
defines = [ "ENABLE_RTC_EVENT_LOG" ]
if (rtc_use_memcheck) {
defines += [ "WEBRTC_USE_MEMCHECK" ]
}
sources = [
"rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc",
"rtc_event_log/output/rtc_event_log_output_file_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_api",
":rtc_event_log_impl",
":rtc_event_log_parser",
":rtc_event_log_proto",
"../api:libjingle_peerconnection_api",
"../call",
"../call:call_interfaces",
"../modules/audio_coding:audio_network_adaptor",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../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",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:protobuf_utils",
"../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",
"../:webrtc_common",
"../call:video_stream_api",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:protobuf_utils",
"../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",
]
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:checks",
"../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" ]
}
}
}
}
if (rtc_include_tests) {
rtc_source_set("mocks") {
testonly = true
sources = [
"rtc_event_log/mock/mock_rtc_event_log.h",
]
deps = [
":rtc_event_log_api",
"../../test:test_support",
]
}
}
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