Wire up audio packet loss to BWE.

webrtc/audio/BUILD.gn

 # Copyright (c) 2015 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")
 
@@ skipping 19 matching lines @@
   deps = [
     "..:webrtc_common",
     "../api:audio_mixer_api",
     "../api:call_api",
     "../base:rtc_base_approved",
     "../base:rtc_task_queue",
     "../call:call_interfaces",
     "../common_audio",
     "../modules/audio_device",
     "../modules/audio_processing",
+    "../modules/bitrate_controller:bitrate_controller",
     "../modules/congestion_controller:congestion_controller",
     "../modules/pacing:pacing",
     "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
     "../modules/rtp_rtcp:rtp_rtcp",
     "../system_wrappers",
     "../voice_engine",
   ]
 }
 if (rtc_include_tests) {
   rtc_source_set("audio_tests") {
@@ skipping 22 matching lines @@
       "utility:utility_tests",
       "//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" ]
     }
   }
 }