webrtc_m130/webrtc/logging/BUILD.gn

# 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",
    "rtc_event_log/rtc_event_log_helper_thread.cc",
    "rtc_event_log/rtc_event_log_helper_thread.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",
    "../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" ]
      }
    }
  }
}