Reimplemeted "Test and fix for huge bwe drop after alr state"

webrtc/BUILD.gn

 # Copyright (c) 2014 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.
 
 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
 
@@ skipping 478 matching lines @@
         "{{bundle_resources_dir}}/{{source_file_part}}",
       ]
     }
   }
 
   rtc_test("webrtc_perf_tests") {
     testonly = true
     configs += [ ":rtc_unittests_config" ]
 
     deps = [
+      "audio:audio_perf_tests",
       "call:call_perf_tests",
       "modules/audio_coding:audio_coding_perf_tests",
       "modules/audio_processing:audio_processing_perf_tests",
       "modules/remote_bitrate_estimator:remote_bitrate_estimator_perf_tests",
       "test:test_main",
       "video:video_full_stack_tests",
     ]
 
     data = webrtc_perf_tests_resources
     if (is_android) {
@@ skipping 26 matching lines @@
       ]
 
       deps = [
         "examples:AppRTCMobile_javalib",
         "sdk/android:libjingle_peerconnection_java",
         "//base:base_java_test_support",
       ]
     }
   }
 }