GN: Add audio_conference_mixer_unittests to modules_unittests.

webrtc/modules/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("../build/webrtc.gni")
 import("audio_coding/audio_coding.gni")
@@ skipping 242 matching lines @@
     ]
 
     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 += [
       "../test:test_main",
       "audio_coding:audio_coding_unittests",
+      "audio_conference_mixer:audio_conference_mixer_unittests",
       "audio_device:audio_device_unittests",
       "audio_mixer:audio_mixer_unittests",
       "audio_processing:audio_processing_unittests",
       "bitrate_controller:bitrate_controller_unittests",
       "congestion_controller:congestion_controller_unittests",
       "desktop_capture:desktop_capture_unittests",
       "media_file:media_file_unittests",
       "pacing:pacing_unittests",
       "remote_bitrate_estimator:remote_bitrate_estimator_unittests",
       "rtp_rtcp:rtp_rtcp_unittests",
@@ skipping 45 matching lines @@
       "../test:test_common",
       "../test:test_main",
       "remote_bitrate_estimator:bwe_simulator_lib",
       "video_capture",
       "//testing/gmock",
       "//testing/gtest",
       "//third_party/gflags",
     ]
   }
 }