blob: fac7e3d9b451364a40718fbf625d1b1bceda7674 [file] [log] [blame]
# 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") {
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/rtc_event_log.h",
"rtc_event_log/rtc_event_log_factory_interface.h",
]
deps = [
"..:webrtc_common",
"../api:libjingle_peerconnection_api",
"../call:video_stream_api",
"../rtc_base:rtc_base_approved",
]
}
rtc_static_library("rtc_event_log_impl") {
sources = [
"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 = "webrtc/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
sources = [
"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/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" ]
}
}
}
}