Add Windows, Mac, Android support to low bandwidth audio test

webrtc/audio/test/low_bandwidth_audio_test.cc

 /*
  *  Copyright (c) 2017 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.
  */
 
 #include <algorithm>
 
 #include "webrtc/audio/test/low_bandwidth_audio_test.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/test/gtest.h"
-#include "webrtc/test/run_test.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace {
 // Wait half a second between stopping sending and stopping receiving audio.
 constexpr int kExtraRecordTimeMs = 500;
 
 // Large bitrate by default.
 const webrtc::CodecInst kDefaultCodec{120, "OPUS", 48000, 960, 2, 64000};
 
@@ skipping 73 matching lines @@
   // and some extra time to account for network delay.
   SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs);
 }
 
 void AudioQualityTest::OnTestFinished() {
   const ::testing::TestInfo* const test_info =
       ::testing::UnitTest::GetInstance()->current_test_info();
 
   // Output information about the input and output audio files so that further
   // processing can be done by an external process.
-  printf("TEST %s %s:%s\n", test_info->name(),
+  printf("TEST %s %s %s\n", test_info->name(),
          AudioInputFile().c_str(), AudioOutputFile().c_str());
 }
 
 
 using LowBandwidthAudioTest = CallTest;
 
 TEST_F(LowBandwidthAudioTest, GoodNetworkHighBitrate) {
   AudioQualityTest test;
   RunBaseTest(&test);
 }
@@ skipping 21 matching lines @@
   }
 };
 
 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
   Mobile2GNetworkTest test;
   RunBaseTest(&test);
 }
 
 }  // namespace test
 }  // namespace webrtc