Resolve cyclic dependency between audio network adaptor and event log api

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") {
-  # TODO(kjellander): Remove (bugs.webrtc.org/7257)
-  # Enabling GN check triggers cyclic dependency error:
-  #  //webrtc/logging:rtc_event_log_api ->
-  #  //webrtc/modules/audio_coding:audio_network_adaptor ->
-  #  //webrtc/logging:rtc_event_log_api
-  check_includes = false
   sources = [
     "rtc_event_log/rtc_event_log.h",
   ]
   deps = [
     "../base:rtc_base_approved",
     "../call:call_interfaces",
+    "../modules/remote_bitrate_estimator:remote_bitrate_estimator",

[terelius, 2017/04/06 09:39:37]

I don't like this. The event log API should not link with
remote_bitrate_estimator. Maybe change BandwidthUsage to an enum class and
forward declare it in the event log?

   ]
 }
 
 rtc_static_library("rtc_event_log_impl") {
   sources = [
     "rtc_event_log/ringbuffer.h",
     "rtc_event_log/rtc_event_log.cc",
     "rtc_event_log/rtc_event_log_helper_thread.cc",
     "rtc_event_log/rtc_event_log_helper_thread.h",
   ]
@@ skipping 32 matching lines @@
     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",
       "../call:call_interfaces",
+      "../modules/audio_coding:audio_network_adaptor",
       "../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 = [
       "../base:rtc_base_approved",
     ]
   }
 
   if (rtc_include_tests) {
     rtc_source_set("rtc_event_log_tests") {
       testonly = true
       sources = [
         "rtc_event_log/ringbuffer_unittest.cc",
         "rtc_event_log/rtc_event_log_unittest.cc",
         "rtc_event_log/rtc_event_log_unittest_helper.cc",
       ]
       deps = [
         ":rtc_event_log_impl",
         ":rtc_event_log_parser",
         "../base:rtc_base_approved",
         "../base:rtc_base_tests_utils",
         "../call",
+        "../modules/audio_coding:audio_network_adaptor",
         "../modules/rtp_rtcp",
         "../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" ]
       }
@@ skipping 39 matching lines @@
         "../modules/rtp_rtcp:rtp_rtcp",
         "//third_party/gflags",
       ]
       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" ]
       }
     }
   }
 }