Replace remaining gflags usages with rtc_base/flags

webrtc/modules/audio_coding/test/delay_test.cc

 /*
  *  Copyright (c) 2013 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 <assert.h>
 #include <math.h>
+#include <string.h>
 
 #include <iostream>
 #include <memory>
 
-#include "gflags/gflags.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/test/Channel.h"
 #include "webrtc/modules/audio_coding/test/PCMFile.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
 DEFINE_string(codec, "isac", "Codec Name");
-DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
-DEFINE_int32(num_channels, 1, "Number of Channels.");
+DEFINE_int(sample_rate_hz, 16000, "Sampling rate in Hertz.");
+DEFINE_int(num_channels, 1, "Number of Channels.");
 DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
-DEFINE_int32(delay, 0, "Delay in millisecond.");
+DEFINE_int(delay, 0, "Delay in millisecond.");
 DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
 DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
 DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
+DEFINE_bool(help, false, "Print this message.");
 
 namespace webrtc {
 
 namespace {
 
 struct CodecSettings {
   char name[50];
   int sample_rate_hz;
   int num_channels;
 };
@@ skipping 25 matching lines @@
       delete channel_a2b_;
       channel_a2b_ = NULL;
     }
     in_file_a_.Close();
   }
 
   void Initialize() {
     test_cntr_ = 0;
     std::string file_name = webrtc::test::ResourcePath(
         "audio_coding/testfile32kHz", "pcm");
-    if (FLAGS_input_file.size() > 0)
-      file_name = FLAGS_input_file;
+    if (strlen(FLAG_input_file) > 0)
+      file_name = FLAG_input_file;
     in_file_a_.Open(file_name, 32000, "rb");
     ASSERT_EQ(0, acm_a_->InitializeReceiver()) <<
         "Couldn't initialize receiver.\n";
     ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
         "Couldn't initialize receiver.\n";
 
-    if (FLAGS_delay > 0) {
-      ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
+    if (FLAG_delay > 0) {
+      ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAG_delay)) <<
           "Failed to set minimum delay.\n";
     }
 
     int num_encoders = acm_a_->NumberOfCodecs();
     CodecInst my_codec_param;
     for (int n = 0; n < num_encoders; n++) {
       EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) <<
           "Failed to get codec.";
       if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
         my_codec_param.channels = 1;
@@ skipping 56 matching lines @@
     ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) <<
         "Failed to set RED.\n";
   }
 
   void ConfigChannel(bool packet_loss) {
     channel_a2b_->SetFECTestWithPacketLoss(packet_loss);
   }
 
   void OpenOutFile(const char* output_id) {
     std::stringstream file_stream;
-    file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
-        << "Hz" << "_" << FLAGS_delay << "ms.pcm";
+    file_stream << "delay_test_" << FLAG_codec << "_" << FLAG_sample_rate_hz
+        << "Hz" << "_" << FLAG_delay << "ms.pcm";
     std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
     std::string file_name = webrtc::test::OutputPath() + file_stream.str();
     out_file_b_.Open(file_name.c_str(), 32000, "wb");
   }
 
   void Run(int duration_sec, const char* output_prefix) {
     OpenOutFile(output_prefix);
     AudioFrame audio_frame;
     uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency();
 
@@ skipping 52 matching lines @@
 
   PCMFile in_file_a_;
   PCMFile out_file_b_;
   int test_cntr_;
   int encoding_sample_rate_hz_;
 };
 
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
-  google::ParseCommandLineFlags(&argc, &argv, true);
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
+    return 1;
+  }
+  if (FLAG_help) {
+    rtc::FlagList::Print(nullptr, false);
+    return 0;
+  }
+
   webrtc::TestSettings test_setting;
-  strcpy(test_setting.codec.name, FLAGS_codec.c_str());
+  strcpy(test_setting.codec.name, FLAG_codec);
 
-  if (FLAGS_sample_rate_hz != 8000 &&
-      FLAGS_sample_rate_hz != 16000 &&
-      FLAGS_sample_rate_hz != 32000 &&
-      FLAGS_sample_rate_hz != 48000) {
+  if (FLAG_sample_rate_hz != 8000 &&
+      FLAG_sample_rate_hz != 16000 &&
+      FLAG_sample_rate_hz != 32000 &&
+      FLAG_sample_rate_hz != 48000) {
     std::cout << "Invalid sampling rate.\n";
     return 1;
   }
-  test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
-  if (FLAGS_num_channels < 1 || FLAGS_num_channels > 2) {
+  test_setting.codec.sample_rate_hz = FLAG_sample_rate_hz;
+  if (FLAG_num_channels < 1 || FLAG_num_channels > 2) {
     std::cout << "Only mono and stereo are supported.\n";
     return 1;
   }
-  test_setting.codec.num_channels = FLAGS_num_channels;
-  test_setting.acm.dtx = FLAGS_dtx;
-  test_setting.acm.fec = FLAGS_fec;
-  test_setting.packet_loss = FLAGS_packet_loss;
+  test_setting.codec.num_channels = FLAG_num_channels;
+  test_setting.acm.dtx = FLAG_dtx;
+  test_setting.acm.fec = FLAG_fec;
+  test_setting.packet_loss = FLAG_packet_loss;
 
   webrtc::DelayTest delay_test;
   delay_test.Initialize();
   delay_test.Perform(&test_setting, 1, 240, "delay_test");
   return 0;
 }