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

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 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 "webrtc/modules/audio_coding/neteq/include/neteq.h"
 
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>  // memset
 
 #include <algorithm>
 #include <memory>
 #include <set>
 #include <string>
 #include <vector>
 
-#include "gflags/gflags.h"
 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
 #include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/rtc_base/ignore_wundef.h"
 #include "webrtc/rtc_base/protobuf_utils.h"
 #include "webrtc/rtc_base/sha1digest.h"
 #include "webrtc/rtc_base/stringencode.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
 RTC_PUSH_IGNORING_WUNDEF()
@@ skipping 413 matching lines @@
   const std::string rtcp_stats_checksum = PlatformChecksum(
       "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
       "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4",
       "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
       "b8880bf9fed2487efbddcb8d94b9937a29ae521d");
 
   DecodeAndCompare(input_rtp_file,
                    output_checksum,
                    network_stats_checksum,
                    rtcp_stats_checksum,
-                   FLAGS_gen_ref);
+                   FLAG_gen_ref);
 }
 
 #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) &&             \
     defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
     defined(WEBRTC_CODEC_OPUS)
 #define MAYBE_TestOpusBitExactness TestOpusBitExactness
 #else
 #define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
 #endif
 TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
@@ skipping 15 matching lines @@
   const std::string rtcp_stats_checksum = PlatformChecksum(
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
 
   DecodeAndCompare(input_rtp_file,
                    output_checksum,
                    network_stats_checksum,
                    rtcp_stats_checksum,
-                   FLAGS_gen_ref);
+                   FLAG_gen_ref);
 }
 
 // Use fax mode to avoid time-scaling. This is to simplify the testing of
 // packet waiting times in the packet buffer.
 class NetEqDecodingTestFaxMode : public NetEqDecodingTest {
  protected:
   NetEqDecodingTestFaxMode() : NetEqDecodingTest() {
     config_.playout_mode = kPlayoutFax;
   }
 };
@@ skipping 1118 matching lines @@
   // Pull out data once.
   AudioFrame output;
   bool muted;
   ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
 
   EXPECT_EQ(std::vector<uint32_t>({kRtpTimestamp1, kRtpTimestamp2}),
             neteq_->LastDecodedTimestamps());
 }
 
 }  // namespace webrtc