Low-bandwidth audio testing

webrtc/test/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.
 
 import("../webrtc.gni")
 import("//build/config/ui.gni")
@@ skipping 236 matching lines @@
     sources = test_support_unittests_resources
     outputs = [
       "{{bundle_resources_dir}}/{{source_file_part}}",
     ]
   }
 }
 
 rtc_test("test_support_unittests") {
   deps = []
   sources = [
+    "fake_audio_device_unittest.cc",
     "fake_network_pipe_unittest.cc",
     "frame_generator_unittest.cc",
     "rtp_file_reader_unittest.cc",
     "rtp_file_writer_unittest.cc",
     "testsupport/always_passing_unittest.cc",
     "testsupport/isolated_output_unittest.cc",
     "testsupport/metrics/video_metrics_unittest.cc",
     "testsupport/packet_reader_unittest.cc",
     "testsupport/perf_test_unittest.cc",
     "testsupport/y4m_frame_writer_unittest.cc",
@@ skipping 207 matching lines @@
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 
   deps = [
     ":test_support",
     "../modules/media_file",
     "//testing/gtest",
   ]
 }