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

webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.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.
  */
 
@@ skipping 62 matching lines @@
   size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(),
                                            input_samples.size(), input_payload);
   RTC_CHECK_EQ(sizeof(input_payload), payload_len);
 
   // Main loop.
   webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
   int64_t start_time_ms = clock->TimeInMilliseconds();
   AudioFrame out_frame;
   while (time_now_ms < runtime_ms) {
     while (packet_input_time_ms <= time_now_ms) {
-      // Drop every N packets, where N = FLAGS_lossrate.
+      // Drop every N packets, where N = FLAG_lossrate.
       bool lost = false;
       if (lossrate > 0) {
         lost = ((rtp_header.sequenceNumber - 1) % lossrate) == 0;
       }
       if (!lost) {
         // Insert packet.
         int error =
             neteq->InsertPacket(rtp_header, input_payload,
                                 packet_input_time_ms * kSampRateHz / 1000);
         if (error != NetEq::kOK)
@@ skipping 30 matching lines @@
       drift_flipped = true;
     }
   }
   int64_t end_time_ms = clock->TimeInMilliseconds();
   delete neteq;
   return end_time_ms - start_time_ms;
 }
 
 }  // namespace test
 }  // namespace webrtc