Replace gflags usages with rtc_base/flags in all targets based on test_main

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 "gflags/gflags.h"
 #include "webrtc/audio/test/low_bandwidth_audio_test.h"
 #include "webrtc/common_audio/wav_file.h"
+#include "webrtc/rtc_base/flags.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/gtest.h"
-#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 
-DEFINE_int32(sample_rate_hz, 16000,
-             "Sample rate (Hz) of the produced audio files.");
+DEFINE_int(sample_rate_hz, 16000,
+           "Sample rate (Hz) of the produced audio files.");
 
 DEFINE_bool(quick, false,
             "Don't do the full audio recording. "
             "Used to quickly check that the test runs without crashing.");
 
 namespace {
 
 // Wait half a second between stopping sending and stopping receiving audio.
 constexpr int kExtraRecordTimeMs = 500;
 
 std::string FileSampleRateSuffix() {
-  return std::to_string(FLAGS_sample_rate_hz / 1000);
+  return std::to_string(FLAG_sample_rate_hz / 1000);
 }
 
 }  // namespace
 
 namespace webrtc {
 namespace test {
 
 AudioQualityTest::AudioQualityTest()
     : EndToEndTest(CallTest::kDefaultTimeoutMs) {}
 
@@ skipping 20 matching lines @@
 }
 
 std::unique_ptr<test::FakeAudioDevice::Capturer>
     AudioQualityTest::CreateCapturer() {
   return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile());
 }
 
 std::unique_ptr<test::FakeAudioDevice::Renderer>
     AudioQualityTest::CreateRenderer() {
   return test::FakeAudioDevice::CreateBoundedWavFileWriter(
-      AudioOutputFile(), FLAGS_sample_rate_hz);
+      AudioOutputFile(), FLAG_sample_rate_hz);
 }
 
 void AudioQualityTest::OnFakeAudioDevicesCreated(
     test::FakeAudioDevice* send_audio_device,
     test::FakeAudioDevice* recv_audio_device) {
   send_audio_device_ = send_audio_device;
 }
 
 FakeNetworkPipe::Config AudioQualityTest::GetNetworkPipeConfig() {
   return FakeNetworkPipe::Config();
@@ skipping 19 matching lines @@
   std::vector<AudioReceiveStream::Config>* receive_configs) {
   // Large bitrate by default.
   const webrtc::SdpAudioFormat kDefaultFormat("OPUS", 48000, 2,
                                               {{"stereo", "1"}});
   send_config->send_codec_spec =
       rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
           {test::CallTest::kAudioSendPayloadType, kDefaultFormat});
 }
 
 void AudioQualityTest::PerformTest() {
-  if (FLAGS_quick) {
+  if (FLAG_quick) {
     // Let the recording run for a small amount of time to check if it works.
     SleepMs(1000);
   } else {
     // Wait until the input audio file is done...
     send_audio_device_->WaitForRecordingEnd();
     // and some extra time to account for network delay.
     SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs);
   }
 }
 
@@ skipping 39 matching lines @@
   }
 };
 
 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
   Mobile2GNetworkTest test;
   RunBaseTest(&test);
 }
 
 }  // namespace test
 }  // namespace webrtc