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

webrtc/voice_engine/file_player_unittests.cc

 /*
  *  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.
  */
 
 // Unit tests for FilePlayer.
 
 #include "webrtc/voice_engine/file_player.h"
 
 #include <stdio.h>
 
 #include <memory>
 #include <string>
 
-#include "gflags/gflags.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/rtc_base/md5digest.h"
 #include "webrtc/rtc_base/stringencode.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 DEFINE_bool(file_player_output, false, "Generate reference files.");
 
 namespace webrtc {
 
 class FilePlayerTest : public ::testing::Test {
  protected:
   static const uint32_t kId = 0;
   static const FileFormats kFileFormat = kFileFormatWavFile;
   static const int kSampleRateHz = 8000;
 
   FilePlayerTest()
       : player_(FilePlayer::CreateFilePlayer(kId, kFileFormat)),
         output_file_(NULL) {}
 
   void SetUp() override {
-    if (FLAGS_file_player_output) {
+    if (FLAG_file_player_output) {
       std::string output_file =
           webrtc::test::OutputPath() + "file_player_unittest_out.pcm";
       output_file_ = fopen(output_file.c_str(), "wb");
       ASSERT_TRUE(output_file_ != NULL);
     }
   }
 
   void TearDown() override {
     if (output_file_)
       fclose(output_file_);
   }
 
   void PlayFileAndCheck(const std::string& input_file,
                         const std::string& ref_checksum,
                         int output_length_ms) {
     const float kScaling = 1;
     ASSERT_EQ(0, player_->StartPlayingFile(input_file.c_str(), false, 0,
                                            kScaling, 0, 0, NULL));
     rtc::Md5Digest checksum;
     for (int i = 0; i < output_length_ms / 10; ++i) {
       int16_t out[10 * kSampleRateHz / 1000] = {0};
       size_t num_samples;
       EXPECT_EQ(
           0, player_->Get10msAudioFromFile(out, &num_samples, kSampleRateHz));
       checksum.Update(out, num_samples * sizeof(out[0]));
-      if (FLAGS_file_player_output) {
+      if (FLAG_file_player_output) {
         ASSERT_EQ(num_samples,
                   fwrite(out, sizeof(out[0]), num_samples, output_file_));
       }
     }
     char checksum_result[rtc::Md5Digest::kSize];
     EXPECT_EQ(rtc::Md5Digest::kSize,
               checksum.Finish(checksum_result, rtc::Md5Digest::kSize));
     EXPECT_EQ(ref_checksum,
               rtc::hex_encode(checksum_result, sizeof(checksum_result)));
   }
@@ skipping 28 matching lines @@
       test::ResourcePath("utility/encapsulated_pcm16b_8khz", "wav");
   // The file is longer than this, but keeping the output shorter limits the
   // runtime for the test.
   const int kOutputLengthMs = 10000;
   const std::string kRefChecksum = "e41d7e1dac8aeae9f21e8e03cd7ecd71";
 
   PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs);
 }
 
 }  // namespace webrtc