diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index b8d1757..8f1e493 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -142,7 +142,6 @@
         "../system_wrappers:field_trial_default",
         "../system_wrappers:metrics_default",
         "../test:rtp_test_utils",
-        "//third_party/gflags",
       ]
       if (!build_with_chromium && is_clang) {
         # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
@@ -165,7 +164,6 @@
         # TODO(kwiberg): Remove this dependency.
         "../api/audio_codecs:audio_codecs_api",
         "../modules/rtp_rtcp:rtp_rtcp",
-        "//third_party/gflags",
       ]
       if (!build_with_chromium && is_clang) {
         # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
@@ -184,7 +182,6 @@
         ":rtc_event_log_impl",
         ":rtc_event_log_proto",
         "../rtc_base:rtc_base_approved",
-        "//third_party/gflags",
       ]
       if (!build_with_chromium && is_clang) {
         # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
diff --git a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
index 23d6941..4275e59 100644
--- a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
+++ b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
@@ -8,17 +8,19 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <string.h>
+
 #include <iostream>
 #include <memory>
 #include <sstream>
 #include <string>
 
-#include "gflags/gflags.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/test/rtp_file_writer.h"
 
 namespace {
@@ -44,6 +46,7 @@
               "",
               "Store only packets with this SSRC (decimal or hex, the latter "
               "starting with 0x).");
+DEFINE_bool(help, false, "Prints this message.");
 
 // Parses the input string for a valid SSRC. If a valid SSRC is found, it is
 // written to the output variable |ssrc|, and true is returned. Otherwise,
@@ -73,22 +76,25 @@
       "Tool for converting an RtcEventLog file to an RTP dump file.\n"
       "Run " +
       program_name +
-      " --helpshort for usage.\n"
+      " --help for usage.\n"
       "Example usage:\n" +
       program_name + " input.rel output.rtp\n";
-  google::SetUsageMessage(usage);
-  google::ParseCommandLineFlags(&argc, &argv, true);
-
-  if (argc != 3) {
-    std::cout << google::ProgramUsage();
-    return 0;
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
+      FLAG_help || argc != 3) {
+    std::cout << usage;
+    if (FLAG_help) {
+      rtc::FlagList::Print(nullptr, false);
+      return 0;
+    }
+    return 1;
   }
+
   std::string input_file = argv[1];
   std::string output_file = argv[2];
 
   uint32_t ssrc_filter = 0;
-  if (!FLAGS_ssrc.empty())
-    RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc_filter))
+  if (strlen(FLAG_ssrc) > 0)
+    RTC_CHECK(ParseSsrc(FLAG_ssrc, &ssrc_filter))
         << "Flag verification has failed.";
 
   webrtc::ParsedRtcEventLog parsed_stream;
@@ -116,7 +122,7 @@
     // some required fields and we attempt to access them. We could consider
     // a softer failure option, but it does not seem useful to generate
     // RTP dumps based on broken event logs.
-    if (!FLAGS_nortp &&
+    if (!FLAG_nortp &&
         parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
       webrtc::test::RtpPacket packet;
       webrtc::PacketDirection direction;
@@ -137,13 +143,13 @@
       rtp_parser.Parse(&parsed_header);
       MediaType media_type =
           parsed_stream.GetMediaType(parsed_header.ssrc, direction);
-      if (FLAGS_noaudio && media_type == MediaType::AUDIO)
+      if (FLAG_noaudio && media_type == MediaType::AUDIO)
         continue;
-      if (FLAGS_novideo && media_type == MediaType::VIDEO)
+      if (FLAG_novideo && media_type == MediaType::VIDEO)
         continue;
-      if (FLAGS_nodata && media_type == MediaType::DATA)
+      if (FLAG_nodata && media_type == MediaType::DATA)
         continue;
-      if (!FLAGS_ssrc.empty()) {
+      if (strlen(FLAG_ssrc) > 0) {
         const uint32_t packet_ssrc =
             webrtc::ByteReader<uint32_t>::ReadBigEndian(
                 reinterpret_cast<const uint8_t*>(packet.data + 8));
@@ -154,7 +160,7 @@
       rtp_writer->WritePacket(&packet);
       rtp_counter++;
     }
-    if (!FLAGS_nortcp &&
+    if (!FLAG_nortcp &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::RTCP_EVENT) {
       webrtc::test::RtpPacket packet;
@@ -175,13 +181,13 @@
       const uint32_t packet_ssrc = webrtc::ByteReader<uint32_t>::ReadBigEndian(
           reinterpret_cast<const uint8_t*>(packet.data + 4));
       MediaType media_type = parsed_stream.GetMediaType(packet_ssrc, direction);
-      if (FLAGS_noaudio && media_type == MediaType::AUDIO)
+      if (FLAG_noaudio && media_type == MediaType::AUDIO)
         continue;
-      if (FLAGS_novideo && media_type == MediaType::VIDEO)
+      if (FLAG_novideo && media_type == MediaType::VIDEO)
         continue;
-      if (FLAGS_nodata && media_type == MediaType::DATA)
+      if (FLAG_nodata && media_type == MediaType::DATA)
         continue;
-      if (!FLAGS_ssrc.empty()) {
+      if (strlen(FLAG_ssrc) > 0) {
         if (packet_ssrc != ssrc_filter)
           continue;
       }
diff --git a/logging/rtc_event_log/rtc_event_log2stats.cc b/logging/rtc_event_log/rtc_event_log2stats.cc
index 6b21cf1..36fa1e3 100644
--- a/logging/rtc_event_log/rtc_event_log2stats.cc
+++ b/logging/rtc_event_log/rtc_event_log2stats.cc
@@ -19,9 +19,9 @@
 #include <utility>
 #include <vector>
 
-#include "gflags/gflags.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/rtc_base/ignore_wundef.h"
 #include "webrtc/rtc_base/logging.h"
 
@@ -36,6 +36,8 @@
 
 namespace {
 
+DEFINE_bool(help, false, "Prints this message.");
+
 struct Stats {
   int count = 0;
   size_t total_size = 0;
@@ -176,15 +178,17 @@
       "Tool for file usage statistics from an RtcEventLog.\n"
       "Run " +
       program_name +
-      " --helpshort for usage.\n"
+      " --help for usage.\n"
       "Example usage:\n" +
       program_name + " input.rel\n";
-  google::SetUsageMessage(usage);
-  google::ParseCommandLineFlags(&argc, &argv, true);
-
-  if (argc != 2) {
-    std::cout << google::ProgramUsage();
-    return 0;
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
+      FLAG_help || argc != 2) {
+    std::cout << usage;
+    if (FLAG_help) {
+      rtc::FlagList::Print(nullptr, false);
+      return 0;
+    }
+    return 1;
   }
   std::string file_name = argv[1];
 
diff --git a/logging/rtc_event_log/rtc_event_log2text.cc b/logging/rtc_event_log/rtc_event_log2text.cc
index c7214e2..3f03108 100644
--- a/logging/rtc_event_log/rtc_event_log2text.cc
+++ b/logging/rtc_event_log/rtc_event_log2text.cc
@@ -8,13 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <string.h>
+
 #include <iostream>
 #include <map>
 #include <sstream>
 #include <string>
 #include <utility>  // pair
 
-#include "gflags/gflags.h"
 #include "webrtc/common_types.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
@@ -35,6 +36,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/flags.h"
 
 namespace {
 
@@ -54,6 +56,7 @@
               "",
               "Print only packets with this SSRC (decimal or hex, the latter "
               "starting with 0x).");
+DEFINE_bool(help, false, "Prints this message.");
 
 using MediaType = webrtc::ParsedRtcEventLog::MediaType;
 
@@ -81,17 +84,17 @@
 bool ExcludePacket(webrtc::PacketDirection direction,
                    MediaType media_type,
                    uint32_t packet_ssrc) {
-  if (FLAGS_nooutgoing && direction == webrtc::kOutgoingPacket)
+  if (FLAG_nooutgoing && direction == webrtc::kOutgoingPacket)
     return true;
-  if (FLAGS_noincoming && direction == webrtc::kIncomingPacket)
+  if (FLAG_noincoming && direction == webrtc::kIncomingPacket)
     return true;
-  if (FLAGS_noaudio && media_type == MediaType::AUDIO)
+  if (FLAG_noaudio && media_type == MediaType::AUDIO)
     return true;
-  if (FLAGS_novideo && media_type == MediaType::VIDEO)
+  if (FLAG_novideo && media_type == MediaType::VIDEO)
     return true;
-  if (FLAGS_nodata && media_type == MediaType::DATA)
+  if (FLAG_nodata && media_type == MediaType::DATA)
     return true;
-  if (!FLAGS_ssrc.empty() && packet_ssrc != filtered_ssrc)
+  if (strlen(FLAG_ssrc) > 0 && packet_ssrc != filtered_ssrc)
     return true;
   return false;
 }
@@ -357,20 +360,22 @@
       "Tool for printing packet information from an RtcEventLog as text.\n"
       "Run " +
       program_name +
-      " --helpshort for usage.\n"
+      " --help for usage.\n"
       "Example usage:\n" +
       program_name + " input.rel\n";
-  google::SetUsageMessage(usage);
-  google::ParseCommandLineFlags(&argc, &argv, true);
-
-  if (argc != 2) {
-    std::cout << google::ProgramUsage();
-    return 0;
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
+      FLAG_help || argc != 2) {
+    std::cout << usage;
+    if (FLAG_help) {
+      rtc::FlagList::Print(nullptr, false);
+      return 0;
+    }
+    return 1;
   }
   std::string input_file = argv[1];
 
-  if (!FLAGS_ssrc.empty())
-    RTC_CHECK(ParseSsrc(FLAGS_ssrc)) << "Flag verification has failed.";
+  if (strlen(FLAG_ssrc) > 0)
+    RTC_CHECK(ParseSsrc(FLAG_ssrc)) << "Flag verification has failed.";
 
   webrtc::RtpHeaderExtensionMap default_map = GetDefaultHeaderExtensionMap();
 
@@ -381,7 +386,7 @@
   }
 
   for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
-    if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming &&
+    if (!FLAG_noconfig && !FLAG_novideo && !FLAG_noincoming &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config =
@@ -402,7 +407,7 @@
       }
       std::cout << "}" << std::endl;
     }
-    if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing &&
+    if (!FLAG_noconfig && !FLAG_novideo && !FLAG_nooutgoing &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
       std::vector<webrtc::rtclog::StreamConfig> configs =
@@ -425,7 +430,7 @@
         std::cout << "}" << std::endl;
       }
     }
-    if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming &&
+    if (!FLAG_noconfig && !FLAG_noaudio && !FLAG_noincoming &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config =
@@ -446,7 +451,7 @@
       }
       std::cout << "}" << std::endl;
     }
-    if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing &&
+    if (!FLAG_noconfig && !FLAG_noaudio && !FLAG_nooutgoing &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config = parsed_stream.GetAudioSendConfig(i);
@@ -465,7 +470,7 @@
       }
       std::cout << "}" << std::endl;
     }
-    if (!FLAGS_nortp &&
+    if (!FLAG_nortp &&
         parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
       size_t header_length;
       size_t total_length;
@@ -516,7 +521,7 @@
       }
       std::cout << std::endl;
     }
-    if (!FLAGS_nortcp &&
+    if (!FLAG_nortcp &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::RTCP_EVENT) {
       size_t length;
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index abe064f..50819f1 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1395,7 +1395,6 @@
       "../../test:test_support",
       "../rtp_rtcp",
       "//testing/gtest",
-      "//third_party/gflags:gflags",
     ]
   }  # delay_test
 
@@ -1425,7 +1424,6 @@
       "../../test:test_support",
       "../rtp_rtcp",
       "//testing/gtest",
-      "//third_party/gflags:gflags",
     ]
   }  # insert_packet_with_timing
 
@@ -1520,7 +1518,6 @@
         "../../rtc_base:rtc_base_approved",
         "../../system_wrappers:system_wrappers_default",
         "../../test:test_support",
-        "//third_party/gflags",
       ]
     }
   }
@@ -1792,9 +1789,9 @@
       ":neteq",
       ":neteq_test_tools",
       ":pcm16b",
+      "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:system_wrappers_default",
       "//testing/gtest",
-      "//third_party/gflags:gflags",
     ]
 
     if (!build_with_chromium && is_clang) {
@@ -1832,9 +1829,9 @@
       ":neteq",
       ":neteq_test_support",
       "../..:webrtc_common",
+      "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support",
-      "//third_party/gflags",
     ]
   }
 
diff --git a/modules/audio_coding/neteq/test/neteq_speed_test.cc b/modules/audio_coding/neteq/test/neteq_speed_test.cc
index b835499..c58381f 100644
--- a/modules/audio_coding/neteq/test/neteq_speed_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_speed_test.cc
@@ -12,43 +12,18 @@
 
 #include <iostream>
 
-#include "gflags/gflags.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
-// Flag validators.
-static bool ValidateRuntime(const char* flagname, int value) {
-  if (value > 0)  // Value is ok.
-    return true;
-  printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
-  return false;
-}
-static bool ValidateLossrate(const char* flagname, int value) {
-  if (value >= 0)  // Value is ok.
-    return true;
-  printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
-  return false;
-}
-static bool ValidateDriftfactor(const char* flagname, double value) {
-  if (value >= 0.0 && value < 1.0)  // Value is ok.
-    return true;
-  printf("Invalid value for --%s: %f\n", flagname, value);
-  return false;
-}
-
 // Define command line flags.
-DEFINE_int32(runtime_ms, 10000, "Simulated runtime in ms.");
-static const bool runtime_ms_dummy =
-    google::RegisterFlagValidator(&FLAGS_runtime_ms, &ValidateRuntime);
-DEFINE_int32(lossrate, 10,
-             "Packet lossrate; drop every N packets.");
-static const bool lossrate_dummy =
-    google::RegisterFlagValidator(&FLAGS_lossrate, &ValidateLossrate);
-DEFINE_double(drift, 0.1,
+DEFINE_int(runtime_ms, 10000, "Simulated runtime in ms.");
+DEFINE_int(lossrate, 10,
+           "Packet lossrate; drop every N packets.");
+DEFINE_float(drift, 0.1f,
              "Clockdrift factor.");
-static const bool drift_dummy =
-    google::RegisterFlagValidator(&FLAGS_drift, &ValidateDriftfactor);
+DEFINE_bool(help, false, "Print this message.");
 
 int main(int argc, char* argv[]) {
   std::string program_name = argv[0];
@@ -58,19 +33,23 @@
       "  --lossrate=N           drop every N packets; default is 10\n"
       "  --drift=F              clockdrift factor between 0.0 and 1.0; "
       "default is 0.1\n";
-  google::SetUsageMessage(usage);
-  google::ParseCommandLineFlags(&argc, &argv, true);
   webrtc::test::SetExecutablePath(argv[0]);
-
-  if (argc != 1) {
-    // Print usage information.
-    std::cout << google::ProgramUsage();
-    return 0;
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
+      FLAG_help || argc != 1) {
+    printf("%s", usage.c_str());
+    if (FLAG_help) {
+      rtc::FlagList::Print(nullptr, false);
+      return 0;
+    }
+    return 1;
   }
+  RTC_CHECK_GT(FLAG_runtime_ms, 0);
+  RTC_CHECK_GE(FLAG_lossrate, 0);
+  RTC_CHECK(FLAG_drift >= 0.0 && FLAG_drift < 1.0);
 
   int64_t result =
-      webrtc::test::NetEqPerformanceTest::Run(FLAGS_runtime_ms, FLAGS_lossrate,
-                                              FLAGS_drift);
+      webrtc::test::NetEqPerformanceTest::Run(FLAG_runtime_ms, FLAG_lossrate,
+                                              FLAG_drift);
   if (result <= 0) {
     std::cout << "There was an error" << std::endl;
     return -1;
diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index c11394b..d6647e4 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -13,6 +13,7 @@
 #include <limits.h>  // For ULONG_MAX returned by strtoul.
 #include <stdio.h>
 #include <stdlib.h>  // For strtoul.
+#include <string.h>
 
 #include <algorithm>
 #include <ios>
@@ -21,7 +22,6 @@
 #include <numeric>
 #include <string>
 
-#include "gflags/gflags.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
@@ -34,6 +34,7 @@
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
@@ -65,86 +66,51 @@
 }
 
 // Flag validators.
-bool ValidatePayloadType(const char* flagname, int32_t value) {
+bool ValidatePayloadType(int value) {
   if (value >= 0 && value <= 127)  // Value is ok.
     return true;
-  printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
+  printf("Payload type must be between 0 and 127, not %d\n",
+         static_cast<int>(value));
   return false;
 }
 
-bool ValidateSsrcValue(const char* flagname, const std::string& str) {
+bool ValidateSsrcValue(const std::string& str) {
   uint32_t dummy_ssrc;
-  return ParseSsrc(str, &dummy_ssrc);
+  if (ParseSsrc(str, &dummy_ssrc)) // Value is ok.
+    return true;
+  printf("Invalid SSRC: %s\n", str.c_str());
+  return false;
 }
 
-static bool ValidateExtensionId(const char* flagname, int32_t value) {
+static bool ValidateExtensionId(int value) {
   if (value > 0 && value <= 255)  // Value is ok.
     return true;
-  printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
+  printf("Extension ID must be between 1 and 255, not %d\n",
+         static_cast<int>(value));
   return false;
 }
 
 // Define command line flags.
-DEFINE_int32(pcmu, 0, "RTP payload type for PCM-u");
-const bool pcmu_dummy =
-    google::RegisterFlagValidator(&FLAGS_pcmu, &ValidatePayloadType);
-DEFINE_int32(pcma, 8, "RTP payload type for PCM-a");
-const bool pcma_dummy =
-    google::RegisterFlagValidator(&FLAGS_pcma, &ValidatePayloadType);
-DEFINE_int32(ilbc, 102, "RTP payload type for iLBC");
-const bool ilbc_dummy =
-    google::RegisterFlagValidator(&FLAGS_ilbc, &ValidatePayloadType);
-DEFINE_int32(isac, 103, "RTP payload type for iSAC");
-const bool isac_dummy =
-    google::RegisterFlagValidator(&FLAGS_isac, &ValidatePayloadType);
-DEFINE_int32(isac_swb, 104, "RTP payload type for iSAC-swb (32 kHz)");
-const bool isac_swb_dummy =
-    google::RegisterFlagValidator(&FLAGS_isac_swb, &ValidatePayloadType);
-DEFINE_int32(opus, 111, "RTP payload type for Opus");
-const bool opus_dummy =
-    google::RegisterFlagValidator(&FLAGS_opus, &ValidatePayloadType);
-DEFINE_int32(pcm16b, 93, "RTP payload type for PCM16b-nb (8 kHz)");
-const bool pcm16b_dummy =
-    google::RegisterFlagValidator(&FLAGS_pcm16b, &ValidatePayloadType);
-DEFINE_int32(pcm16b_wb, 94, "RTP payload type for PCM16b-wb (16 kHz)");
-const bool pcm16b_wb_dummy =
-    google::RegisterFlagValidator(&FLAGS_pcm16b_wb, &ValidatePayloadType);
-DEFINE_int32(pcm16b_swb32, 95, "RTP payload type for PCM16b-swb32 (32 kHz)");
-const bool pcm16b_swb32_dummy =
-    google::RegisterFlagValidator(&FLAGS_pcm16b_swb32, &ValidatePayloadType);
-DEFINE_int32(pcm16b_swb48, 96, "RTP payload type for PCM16b-swb48 (48 kHz)");
-const bool pcm16b_swb48_dummy =
-    google::RegisterFlagValidator(&FLAGS_pcm16b_swb48, &ValidatePayloadType);
-DEFINE_int32(g722, 9, "RTP payload type for G.722");
-const bool g722_dummy =
-    google::RegisterFlagValidator(&FLAGS_g722, &ValidatePayloadType);
-DEFINE_int32(avt, 106, "RTP payload type for AVT/DTMF (8 kHz)");
-const bool avt_dummy =
-    google::RegisterFlagValidator(&FLAGS_avt, &ValidatePayloadType);
-DEFINE_int32(avt_16, 114, "RTP payload type for AVT/DTMF (16 kHz)");
-const bool avt_16_dummy =
-    google::RegisterFlagValidator(&FLAGS_avt_16, &ValidatePayloadType);
-DEFINE_int32(avt_32, 115, "RTP payload type for AVT/DTMF (32 kHz)");
-const bool avt_32_dummy =
-    google::RegisterFlagValidator(&FLAGS_avt_32, &ValidatePayloadType);
-DEFINE_int32(avt_48, 116, "RTP payload type for AVT/DTMF (48 kHz)");
-const bool avt_48_dummy =
-    google::RegisterFlagValidator(&FLAGS_avt_48, &ValidatePayloadType);
-DEFINE_int32(red, 117, "RTP payload type for redundant audio (RED)");
-const bool red_dummy =
-    google::RegisterFlagValidator(&FLAGS_red, &ValidatePayloadType);
-DEFINE_int32(cn_nb, 13, "RTP payload type for comfort noise (8 kHz)");
-const bool cn_nb_dummy =
-    google::RegisterFlagValidator(&FLAGS_cn_nb, &ValidatePayloadType);
-DEFINE_int32(cn_wb, 98, "RTP payload type for comfort noise (16 kHz)");
-const bool cn_wb_dummy =
-    google::RegisterFlagValidator(&FLAGS_cn_wb, &ValidatePayloadType);
-DEFINE_int32(cn_swb32, 99, "RTP payload type for comfort noise (32 kHz)");
-const bool cn_swb32_dummy =
-    google::RegisterFlagValidator(&FLAGS_cn_swb32, &ValidatePayloadType);
-DEFINE_int32(cn_swb48, 100, "RTP payload type for comfort noise (48 kHz)");
-const bool cn_swb48_dummy =
-    google::RegisterFlagValidator(&FLAGS_cn_swb48, &ValidatePayloadType);
+DEFINE_int(pcmu, 0, "RTP payload type for PCM-u");
+DEFINE_int(pcma, 8, "RTP payload type for PCM-a");
+DEFINE_int(ilbc, 102, "RTP payload type for iLBC");
+DEFINE_int(isac, 103, "RTP payload type for iSAC");
+DEFINE_int(isac_swb, 104, "RTP payload type for iSAC-swb (32 kHz)");
+DEFINE_int(opus, 111, "RTP payload type for Opus");
+DEFINE_int(pcm16b, 93, "RTP payload type for PCM16b-nb (8 kHz)");
+DEFINE_int(pcm16b_wb, 94, "RTP payload type for PCM16b-wb (16 kHz)");
+DEFINE_int(pcm16b_swb32, 95, "RTP payload type for PCM16b-swb32 (32 kHz)");
+DEFINE_int(pcm16b_swb48, 96, "RTP payload type for PCM16b-swb48 (48 kHz)");
+DEFINE_int(g722, 9, "RTP payload type for G.722");
+DEFINE_int(avt, 106, "RTP payload type for AVT/DTMF (8 kHz)");
+DEFINE_int(avt_16, 114, "RTP payload type for AVT/DTMF (16 kHz)");
+DEFINE_int(avt_32, 115, "RTP payload type for AVT/DTMF (32 kHz)");
+DEFINE_int(avt_48, 116, "RTP payload type for AVT/DTMF (48 kHz)");
+DEFINE_int(red, 117, "RTP payload type for redundant audio (RED)");
+DEFINE_int(cn_nb, 13, "RTP payload type for comfort noise (8 kHz)");
+DEFINE_int(cn_wb, 98, "RTP payload type for comfort noise (16 kHz)");
+DEFINE_int(cn_swb32, 99, "RTP payload type for comfort noise (32 kHz)");
+DEFINE_int(cn_swb48, 100, "RTP payload type for comfort noise (48 kHz)");
 DEFINE_bool(codec_map, false, "Prints the mapping between RTP payload type and "
     "codec");
 DEFINE_string(replacement_audio_file, "",
@@ -153,21 +119,13 @@
               "",
               "Only use packets with this SSRC (decimal or hex, the latter "
               "starting with 0x)");
-const bool hex_ssrc_dummy =
-    google::RegisterFlagValidator(&FLAGS_ssrc, &ValidateSsrcValue);
-DEFINE_int32(audio_level, 1, "Extension ID for audio level (RFC 6464)");
-const bool audio_level_dummy =
-    google::RegisterFlagValidator(&FLAGS_audio_level, &ValidateExtensionId);
-DEFINE_int32(abs_send_time, 3, "Extension ID for absolute sender time");
-const bool abs_send_time_dummy =
-    google::RegisterFlagValidator(&FLAGS_abs_send_time, &ValidateExtensionId);
-DEFINE_int32(transport_seq_no, 5, "Extension ID for transport sequence number");
-const bool transport_seq_no_dummy =
-    google::RegisterFlagValidator(&FLAGS_transport_seq_no,
-                                  &ValidateExtensionId);
+DEFINE_int(audio_level, 1, "Extension ID for audio level (RFC 6464)");
+DEFINE_int(abs_send_time, 3, "Extension ID for absolute sender time");
+DEFINE_int(transport_seq_no, 5, "Extension ID for transport sequence number");
 DEFINE_bool(matlabplot,
             false,
             "Generates a matlab script for plotting the delay profile");
+DEFINE_bool(help, false, "Prints this message");
 
 // Maps a codec type to a printable name string.
 std::string CodecName(NetEqDecoder codec) {
@@ -218,51 +176,51 @@
   }
 }
 
-void PrintCodecMappingEntry(NetEqDecoder codec, google::int32 flag) {
+void PrintCodecMappingEntry(NetEqDecoder codec, int flag) {
   std::cout << CodecName(codec) << ": " << flag << std::endl;
 }
 
 void PrintCodecMapping() {
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMu, FLAGS_pcmu);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMa, FLAGS_pcma);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderILBC, FLAGS_ilbc);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderISAC, FLAGS_isac);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderISACswb, FLAGS_isac_swb);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderOpus, FLAGS_opus);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16B, FLAGS_pcm16b);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bwb, FLAGS_pcm16b_wb);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMu, FLAG_pcmu);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMa, FLAG_pcma);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderILBC, FLAG_ilbc);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderISAC, FLAG_isac);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderISACswb, FLAG_isac_swb);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderOpus, FLAG_opus);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16B, FLAG_pcm16b);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bwb, FLAG_pcm16b_wb);
   PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bswb32kHz,
-                         FLAGS_pcm16b_swb32);
+                         FLAG_pcm16b_swb32);
   PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bswb48kHz,
-                         FLAGS_pcm16b_swb48);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderG722, FLAGS_g722);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT, FLAGS_avt);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT16kHz, FLAGS_avt_16);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT32kHz, FLAGS_avt_32);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT48kHz, FLAGS_avt_48);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderRED, FLAGS_red);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGnb, FLAGS_cn_nb);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGwb, FLAGS_cn_wb);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb32kHz, FLAGS_cn_swb32);
-  PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb48kHz, FLAGS_cn_swb48);
+                         FLAG_pcm16b_swb48);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderG722, FLAG_g722);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT, FLAG_avt);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT16kHz, FLAG_avt_16);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT32kHz, FLAG_avt_32);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT48kHz, FLAG_avt_48);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderRED, FLAG_red);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGnb, FLAG_cn_nb);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGwb, FLAG_cn_wb);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb32kHz, FLAG_cn_swb32);
+  PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb48kHz, FLAG_cn_swb48);
 }
 
 rtc::Optional<int> CodecSampleRate(uint8_t payload_type) {
-  if (payload_type == FLAGS_pcmu || payload_type == FLAGS_pcma ||
-      payload_type == FLAGS_ilbc || payload_type == FLAGS_pcm16b ||
-      payload_type == FLAGS_cn_nb || payload_type == FLAGS_avt)
+  if (payload_type == FLAG_pcmu || payload_type == FLAG_pcma ||
+      payload_type == FLAG_ilbc || payload_type == FLAG_pcm16b ||
+      payload_type == FLAG_cn_nb || payload_type == FLAG_avt)
     return rtc::Optional<int>(8000);
-  if (payload_type == FLAGS_isac || payload_type == FLAGS_pcm16b_wb ||
-      payload_type == FLAGS_g722 || payload_type == FLAGS_cn_wb ||
-      payload_type == FLAGS_avt_16)
+  if (payload_type == FLAG_isac || payload_type == FLAG_pcm16b_wb ||
+      payload_type == FLAG_g722 || payload_type == FLAG_cn_wb ||
+      payload_type == FLAG_avt_16)
     return rtc::Optional<int>(16000);
-  if (payload_type == FLAGS_isac_swb || payload_type == FLAGS_pcm16b_swb32 ||
-      payload_type == FLAGS_cn_swb32 || payload_type == FLAGS_avt_32)
+  if (payload_type == FLAG_isac_swb || payload_type == FLAG_pcm16b_swb32 ||
+      payload_type == FLAG_cn_swb32 || payload_type == FLAG_avt_32)
     return rtc::Optional<int>(32000);
-  if (payload_type == FLAGS_opus || payload_type == FLAGS_pcm16b_swb48 ||
-      payload_type == FLAGS_cn_swb48 || payload_type == FLAGS_avt_48)
+  if (payload_type == FLAG_opus || payload_type == FLAG_pcm16b_swb48 ||
+      payload_type == FLAG_cn_swb48 || payload_type == FLAG_avt_48)
     return rtc::Optional<int>(48000);
-  if (payload_type == FLAGS_red)
+  if (payload_type == FLAG_red)
     return rtc::Optional<int>(0);
   return rtc::Optional<int>();
 }
@@ -460,31 +418,61 @@
 int RunTest(int argc, char* argv[]) {
   std::string program_name = argv[0];
   std::string usage = "Tool for decoding an RTP dump file using NetEq.\n"
-      "Run " + program_name + " --helpshort for usage.\n"
+      "Run " + program_name + " --help for usage.\n"
       "Example usage:\n" + program_name +
       " input.rtp output.{pcm, wav}\n";
-  google::SetUsageMessage(usage);
-  google::ParseCommandLineFlags(&argc, &argv, true);
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
+    return 1;
+  }
+  if (FLAG_help) {
+    std::cout << usage;
+    rtc::FlagList::Print(nullptr, false);
+    return 0;
+  }
 
-  if (FLAGS_codec_map) {
+  if (FLAG_codec_map) {
     PrintCodecMapping();
   }
 
   if (argc != 3) {
-    if (FLAGS_codec_map) {
+    if (FLAG_codec_map) {
       // We have already printed the codec map. Just end the program.
       return 0;
     }
     // Print usage information.
-    std::cout << google::ProgramUsage();
+    std::cout << usage;
     return 0;
   }
+  RTC_CHECK(ValidatePayloadType(FLAG_pcmu));
+  RTC_CHECK(ValidatePayloadType(FLAG_pcma));
+  RTC_CHECK(ValidatePayloadType(FLAG_ilbc));
+  RTC_CHECK(ValidatePayloadType(FLAG_isac));
+  RTC_CHECK(ValidatePayloadType(FLAG_isac_swb));
+  RTC_CHECK(ValidatePayloadType(FLAG_opus));
+  RTC_CHECK(ValidatePayloadType(FLAG_pcm16b));
+  RTC_CHECK(ValidatePayloadType(FLAG_pcm16b_wb));
+  RTC_CHECK(ValidatePayloadType(FLAG_pcm16b_swb32));
+  RTC_CHECK(ValidatePayloadType(FLAG_pcm16b_swb48));
+  RTC_CHECK(ValidatePayloadType(FLAG_g722));
+  RTC_CHECK(ValidatePayloadType(FLAG_avt));
+  RTC_CHECK(ValidatePayloadType(FLAG_avt_16));
+  RTC_CHECK(ValidatePayloadType(FLAG_avt_32));
+  RTC_CHECK(ValidatePayloadType(FLAG_avt_48));
+  RTC_CHECK(ValidatePayloadType(FLAG_red));
+  RTC_CHECK(ValidatePayloadType(FLAG_cn_nb));
+  RTC_CHECK(ValidatePayloadType(FLAG_cn_wb));
+  RTC_CHECK(ValidatePayloadType(FLAG_cn_swb32));
+  RTC_CHECK(ValidatePayloadType(FLAG_cn_swb48));
+  RTC_CHECK(ValidateSsrcValue(FLAG_ssrc));
+  RTC_CHECK(ValidateExtensionId(FLAG_audio_level));
+  RTC_CHECK(ValidateExtensionId(FLAG_abs_send_time));
+  RTC_CHECK(ValidateExtensionId(FLAG_transport_seq_no));
 
   // Gather RTP header extensions in a map.
   NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
-      {FLAGS_audio_level, kRtpExtensionAudioLevel},
-      {FLAGS_abs_send_time, kRtpExtensionAbsoluteSendTime},
-      {FLAGS_transport_seq_no, kRtpExtensionTransportSequenceNumber}};
+      {FLAG_audio_level, kRtpExtensionAudioLevel},
+      {FLAG_abs_send_time, kRtpExtensionAbsoluteSendTime},
+      {FLAG_transport_seq_no, kRtpExtensionTransportSequenceNumber}};
 
   const std::string input_file_name = argv[1];
   std::unique_ptr<NetEqInput> input;
@@ -500,9 +488,9 @@
   RTC_CHECK(!input->ended()) << "Input file is empty";
 
   // Check if an SSRC value was provided.
-  if (!FLAGS_ssrc.empty()) {
+  if (strlen(FLAG_ssrc) > 0) {
     uint32_t ssrc;
-    RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc)) << "Flag verification has failed.";
+    RTC_CHECK(ParseSsrc(FLAG_ssrc, &ssrc)) << "Flag verification has failed.";
     input.reset(new FilterSsrcInput(std::move(input), ssrc));
   }
 
@@ -557,39 +545,39 @@
   std::cout << "Output file: " << output_file_name << std::endl;
 
   NetEqTest::DecoderMap codecs = {
-      {FLAGS_pcmu, std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu")},
-      {FLAGS_pcma, std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma")},
-      {FLAGS_ilbc, std::make_pair(NetEqDecoder::kDecoderILBC, "ilbc")},
-      {FLAGS_isac, std::make_pair(NetEqDecoder::kDecoderISAC, "isac")},
-      {FLAGS_isac_swb,
+      {FLAG_pcmu, std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu")},
+      {FLAG_pcma, std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma")},
+      {FLAG_ilbc, std::make_pair(NetEqDecoder::kDecoderILBC, "ilbc")},
+      {FLAG_isac, std::make_pair(NetEqDecoder::kDecoderISAC, "isac")},
+      {FLAG_isac_swb,
        std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb")},
-      {FLAGS_opus, std::make_pair(NetEqDecoder::kDecoderOpus, "opus")},
-      {FLAGS_pcm16b, std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb")},
-      {FLAGS_pcm16b_wb,
+      {FLAG_opus, std::make_pair(NetEqDecoder::kDecoderOpus, "opus")},
+      {FLAG_pcm16b, std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb")},
+      {FLAG_pcm16b_wb,
        std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb")},
-      {FLAGS_pcm16b_swb32,
+      {FLAG_pcm16b_swb32,
        std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32")},
-      {FLAGS_pcm16b_swb48,
+      {FLAG_pcm16b_swb48,
        std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48")},
-      {FLAGS_g722, std::make_pair(NetEqDecoder::kDecoderG722, "g722")},
-      {FLAGS_avt, std::make_pair(NetEqDecoder::kDecoderAVT, "avt")},
-      {FLAGS_avt_16, std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16")},
-      {FLAGS_avt_32,
+      {FLAG_g722, std::make_pair(NetEqDecoder::kDecoderG722, "g722")},
+      {FLAG_avt, std::make_pair(NetEqDecoder::kDecoderAVT, "avt")},
+      {FLAG_avt_16, std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16")},
+      {FLAG_avt_32,
        std::make_pair(NetEqDecoder::kDecoderAVT32kHz, "avt-32")},
-      {FLAGS_avt_48,
+      {FLAG_avt_48,
        std::make_pair(NetEqDecoder::kDecoderAVT48kHz, "avt-48")},
-      {FLAGS_red, std::make_pair(NetEqDecoder::kDecoderRED, "red")},
-      {FLAGS_cn_nb, std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb")},
-      {FLAGS_cn_wb, std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb")},
-      {FLAGS_cn_swb32,
+      {FLAG_red, std::make_pair(NetEqDecoder::kDecoderRED, "red")},
+      {FLAG_cn_nb, std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb")},
+      {FLAG_cn_wb, std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb")},
+      {FLAG_cn_swb32,
        std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32")},
-      {FLAGS_cn_swb48,
+      {FLAG_cn_swb48,
        std::make_pair(NetEqDecoder::kDecoderCNGswb48kHz, "cng-swb48")}};
 
   // Check if a replacement audio file was provided.
   std::unique_ptr<AudioDecoder> replacement_decoder;
   NetEqTest::ExtDecoderMap ext_codecs;
-  if (!FLAGS_replacement_audio_file.empty()) {
+  if (strlen(FLAG_replacement_audio_file) > 0) {
     // Find largest unused payload type.
     int replacement_pt = 127;
     while (!(codecs.find(replacement_pt) == codecs.end() &&
@@ -607,16 +595,16 @@
     };
 
     std::set<uint8_t> cn_types = std_set_int32_to_uint8(
-        {FLAGS_cn_nb, FLAGS_cn_wb, FLAGS_cn_swb32, FLAGS_cn_swb48});
+        {FLAG_cn_nb, FLAG_cn_wb, FLAG_cn_swb32, FLAG_cn_swb48});
     std::set<uint8_t> forbidden_types =
-        std_set_int32_to_uint8({FLAGS_g722, FLAGS_red, FLAGS_avt,
-                                FLAGS_avt_16, FLAGS_avt_32, FLAGS_avt_48});
+        std_set_int32_to_uint8({FLAG_g722, FLAG_red, FLAG_avt,
+                                FLAG_avt_16, FLAG_avt_32, FLAG_avt_48});
     input.reset(new NetEqReplacementInput(std::move(input), replacement_pt,
                                           cn_types, forbidden_types));
 
     replacement_decoder.reset(new FakeDecodeFromFile(
         std::unique_ptr<InputAudioFile>(
-            new InputAudioFile(FLAGS_replacement_audio_file)),
+            new InputAudioFile(FLAG_replacement_audio_file)),
         48000, false));
     NetEqTest::ExternalDecoderInfo ext_dec_info = {
         replacement_decoder.get(), NetEqDecoder::kDecoderArbitrary,
@@ -626,7 +614,7 @@
 
   NetEqTest::Callbacks callbacks;
   std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer;
-  if (FLAGS_matlabplot) {
+  if (FLAG_matlabplot) {
     delay_analyzer.reset(new NetEqDelayAnalyzer);
   }
 
@@ -641,7 +629,7 @@
 
   int64_t test_duration_ms = test.Run();
 
-  if (FLAGS_matlabplot) {
+  if (FLAG_matlabplot) {
     auto matlab_script_name = output_file_name;
     std::replace(matlab_script_name.begin(), matlab_script_name.end(), '.',
                  '_');
diff --git a/modules/audio_coding/neteq/tools/rtp_analyze.cc b/modules/audio_coding/neteq/tools/rtp_analyze.cc
index 74c64e0..23f96c5 100644
--- a/modules/audio_coding/neteq/tools/rtp_analyze.cc
+++ b/modules/audio_coding/neteq/tools/rtp_analyze.cc
@@ -14,34 +14,17 @@
 #include <memory>
 #include <vector>
 
-#include "gflags/gflags.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-
-// Flag validator.
-static bool ValidatePayloadType(const char* flagname, int32_t value) {
-  if (value >= 0 && value <= 127)  // Value is ok.
-    return true;
-  printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
-  return false;
-}
-static bool ValidateExtensionId(const char* flagname, int32_t value) {
-  if (value > 0 && value <= 255)  // Value is ok.
-    return true;
-  printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
-  return false;
-}
+#include "webrtc/rtc_base/flags.h"
 
 // Define command line flags.
-DEFINE_int32(red, 117, "RTP payload type for RED");
-static const bool red_dummy =
-    google::RegisterFlagValidator(&FLAGS_red, &ValidatePayloadType);
-DEFINE_int32(audio_level, 1, "Extension ID for audio level (RFC 6464)");
-static const bool audio_level_dummy =
-    google::RegisterFlagValidator(&FLAGS_audio_level, &ValidateExtensionId);
-DEFINE_int32(abs_send_time, 3, "Extension ID for absolute sender time");
-static const bool abs_send_time_dummy =
-    google::RegisterFlagValidator(&FLAGS_abs_send_time, &ValidateExtensionId);
+DEFINE_int(red, 117, "RTP payload type for RED");
+DEFINE_int(audio_level, -1, "Extension ID for audio level (RFC 6464); "
+                            "-1 not to print audio level");
+DEFINE_int(abs_send_time, -1, "Extension ID for absolute sender time; "
+                             "-1 not to print absolute send time");
+DEFINE_bool(help, false, "Print this message");
 
 int main(int argc, char* argv[]) {
   std::string program_name = argv[0];
@@ -49,36 +32,43 @@
       "Tool for parsing an RTP dump file to text output.\n"
       "Run " +
       program_name +
-      " --helpshort for usage.\n"
+      " --help for usage.\n"
       "Example usage:\n" +
       program_name + " input.rtp output.txt\n\n" +
-      "Output is sent to stdout if no output file is given." +
-      "Note that this tool can read files with our without payloads.";
-  google::SetUsageMessage(usage);
-  google::ParseCommandLineFlags(&argc, &argv, true);
-
-  if (argc != 2 && argc != 3) {
-    // Print usage information.
-    printf("%s", google::ProgramUsage());
-    return 0;
+      "Output is sent to stdout if no output file is given. " +
+      "Note that this tool can read files with or without payloads.\n";
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
+      FLAG_help || (argc != 2 && argc != 3)) {
+    printf("%s", usage.c_str());
+    if (FLAG_help) {
+      rtc::FlagList::Print(nullptr, false);
+      return 0;
+    }
+    return 1;
   }
 
+  RTC_CHECK(FLAG_red >= 0 && FLAG_red <= 127);  // Payload type
+  RTC_CHECK(FLAG_audio_level == -1 ||  // Default
+      (FLAG_audio_level > 0 && FLAG_audio_level <= 255));  // Extension ID
+  RTC_CHECK(FLAG_abs_send_time == -1 ||  // Default
+      (FLAG_abs_send_time > 0 && FLAG_abs_send_time <= 255));  // Extension ID
+
   printf("Input file: %s\n", argv[1]);
   std::unique_ptr<webrtc::test::RtpFileSource> file_source(
       webrtc::test::RtpFileSource::Create(argv[1]));
   assert(file_source.get());
   // Set RTP extension IDs.
   bool print_audio_level = false;
-  if (!google::GetCommandLineFlagInfoOrDie("audio_level").is_default) {
+  if (FLAG_audio_level != -1) {
     print_audio_level = true;
     file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
-                                            FLAGS_audio_level);
+                                            FLAG_audio_level);
   }
   bool print_abs_send_time = false;
-  if (!google::GetCommandLineFlagInfoOrDie("abs_send_time").is_default) {
+  if (FLAG_abs_send_time != -1) {
     print_abs_send_time = true;
     file_source->RegisterRtpHeaderExtension(
-        webrtc::kRtpExtensionAbsoluteSendTime, FLAGS_abs_send_time);
+        webrtc::kRtpExtensionAbsoluteSendTime, FLAG_abs_send_time);
   }
 
   FILE* out_file;
@@ -160,7 +150,7 @@
     }
     fprintf(out_file, "\n");
 
-    if (packet->header().payloadType == FLAGS_red) {
+    if (packet->header().payloadType == FLAG_red) {
       std::list<webrtc::RTPHeader*> red_headers;
       packet->ExtractRedHeaders(&red_headers);
       while (!red_headers.empty()) {
diff --git a/modules/audio_coding/test/delay_test.cc b/modules/audio_coding/test/delay_test.cc
index ce24493..0ce7fd2 100644
--- a/modules/audio_coding/test/delay_test.cc
+++ b/modules/audio_coding/test/delay_test.cc
@@ -10,11 +10,11 @@
 
 #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"
@@ -22,19 +22,21 @@
 #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 {
 
@@ -80,16 +82,16 @@
     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";
     }
 
@@ -166,8 +168,8 @@
 
   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");
@@ -240,26 +242,33 @@
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
-  google::ParseCommandLineFlags(&argc, &argv, true);
-  webrtc::TestSettings test_setting;
-  strcpy(test_setting.codec.name, FLAGS_codec.c_str());
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
+    return 1;
+  }
+  if (FLAG_help) {
+    rtc::FlagList::Print(nullptr, false);
+    return 0;
+  }
 
-  if (FLAGS_sample_rate_hz != 8000 &&
-      FLAGS_sample_rate_hz != 16000 &&
-      FLAGS_sample_rate_hz != 32000 &&
-      FLAGS_sample_rate_hz != 48000) {
+  webrtc::TestSettings test_setting;
+  strcpy(test_setting.codec.name, FLAG_codec);
+
+  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();
diff --git a/modules/audio_coding/test/insert_packet_with_timing.cc b/modules/audio_coding/test/insert_packet_with_timing.cc
index 4fa4e52..db58289 100644
--- a/modules/audio_coding/test/insert_packet_with_timing.cc
+++ b/modules/audio_coding/test/insert_packet_with_timing.cc
@@ -9,31 +9,32 @@
  */
 
 #include <stdio.h>
+#include <string.h>
 
 #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/test/Channel.h"
 #include "webrtc/modules/audio_coding/test/PCMFile.h"
 #include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 // Codec.
 DEFINE_string(codec, "opus", "Codec Name");
-DEFINE_int32(codec_sample_rate_hz, 48000, "Sampling rate in Hertz.");
-DEFINE_int32(codec_channels, 1, "Number of channels of the codec.");
+DEFINE_int(codec_sample_rate_hz, 48000, "Sampling rate in Hertz.");
+DEFINE_int(codec_channels, 1, "Number of channels of the codec.");
 
 // PCM input/output.
 DEFINE_string(input, "", "Input PCM file at 16 kHz.");
 DEFINE_bool(input_stereo, false, "Input is stereo.");
-DEFINE_int32(input_fs_hz, 32000, "Input sample rate Hz.");
+DEFINE_int(input_fs_hz, 32000, "Input sample rate Hz.");
 DEFINE_string(output, "insert_rtp_with_timing_out.pcm", "OutputFile");
-DEFINE_int32(output_fs_hz, 32000, "Output sample rate Hz");
+DEFINE_int(output_fs_hz, 32000, "Output sample rate Hz");
 
 // Timing files
 DEFINE_string(seq_num, "seq_num", "Sequence number file.");
@@ -45,7 +46,9 @@
 
 // Other setups
 DEFINE_bool(verbose, false, "Verbosity.");
-DEFINE_double(loss_rate, 0, "Rate of packet loss < 1");
+DEFINE_float(loss_rate, 0, "Rate of packet loss < 1");
+
+DEFINE_bool(help, false, "Prints this message.");
 
 const int32_t kAudioPlayedOut = 0x00000001;
 const int32_t kPacketPushedIn = 0x00000001 << 1;
@@ -61,10 +64,10 @@
         send_acm_(AudioCodingModule::Create(0, sender_clock_)),
         receive_acm_(AudioCodingModule::Create(0, receiver_clock_)),
         channel_(new Channel),
-        seq_num_fid_(fopen(FLAGS_seq_num.c_str(), "rt")),
-        send_ts_fid_(fopen(FLAGS_send_ts.c_str(), "rt")),
-        receive_ts_fid_(fopen(FLAGS_receive_ts.c_str(), "rt")),
-        pcm_out_fid_(fopen(FLAGS_output.c_str(), "wb")),
+        seq_num_fid_(fopen(FLAG_seq_num, "rt")),
+        send_ts_fid_(fopen(FLAG_send_ts, "rt")),
+        receive_ts_fid_(fopen(FLAG_receive_ts, "rt")),
+        pcm_out_fid_(fopen(FLAG_output, "wb")),
         samples_in_1ms_(48),
         num_10ms_in_codec_frame_(2),  // Typical 20 ms frames.
         time_to_insert_packet_ms_(3),  // An arbitrary offset on pushing packet.
@@ -90,9 +93,9 @@
     next_receive_ts_ = ReceiveTimestamp();
 
     CodecInst codec;
-    ASSERT_EQ(0, AudioCodingModule::Codec(FLAGS_codec.c_str(), &codec,
-                             FLAGS_codec_sample_rate_hz,
-                             FLAGS_codec_channels));
+    ASSERT_EQ(0, AudioCodingModule::Codec(FLAG_codec, &codec,
+                             FLAG_codec_sample_rate_hz,
+                             FLAG_codec_channels));
     ASSERT_EQ(0, receive_acm_->InitializeReceiver());
     ASSERT_EQ(0, send_acm_->RegisterSendCodec(codec));
     ASSERT_EQ(true, receive_acm_->RegisterReceiveCodec(codec.pltype,
@@ -105,27 +108,27 @@
     channel_->RegisterReceiverACM(receive_acm_.get());
     send_acm_->RegisterTransportCallback(channel_);
 
-    if (FLAGS_input.size() == 0) {
+    if (strlen(FLAG_input) == 0) {
       std::string file_name = test::ResourcePath("audio_coding/testfile32kHz",
                                                  "pcm");
       pcm_in_fid_.Open(file_name, 32000, "r", true);  // auto-rewind
       std::cout << "Input file " << file_name << " 32 kHz mono." << std::endl;
     } else {
-      pcm_in_fid_.Open(FLAGS_input, static_cast<uint16_t>(FLAGS_input_fs_hz),
+      pcm_in_fid_.Open(FLAG_input, static_cast<uint16_t>(FLAG_input_fs_hz),
                     "r", true);  // auto-rewind
-      std::cout << "Input file " << FLAGS_input << "at " << FLAGS_input_fs_hz
-          << " Hz in " << ((FLAGS_input_stereo) ? "stereo." : "mono.")
+      std::cout << "Input file " << FLAG_input << "at " << FLAG_input_fs_hz
+          << " Hz in " << ((FLAG_input_stereo) ? "stereo." : "mono.")
           << std::endl;
-      pcm_in_fid_.ReadStereo(FLAGS_input_stereo);
+      pcm_in_fid_.ReadStereo(FLAG_input_stereo);
     }
 
     ASSERT_TRUE(pcm_out_fid_ != NULL);
-    std::cout << "Output file " << FLAGS_output << " at " << FLAGS_output_fs_hz
+    std::cout << "Output file " << FLAG_output << " at " << FLAG_output_fs_hz
         << " Hz." << std::endl;
 
     // Other setups
-    if (FLAGS_loss_rate > 0)
-      loss_threshold_ = RAND_MAX * FLAGS_loss_rate;
+    if (FLAG_loss_rate > 0)
+      loss_threshold_ = RAND_MAX * FLAG_loss_rate;
     else
       loss_threshold_ = 0;
   }
@@ -144,7 +147,7 @@
     if (time_to_playout_audio_ms_ == 0) {
       time_to_playout_audio_ms_ = kPlayoutPeriodMs;
       bool muted;
-      receive_acm_->PlayoutData10Ms(static_cast<int>(FLAGS_output_fs_hz),
+      receive_acm_->PlayoutData10Ms(static_cast<int>(FLAG_output_fs_hz),
                                     &frame_, &muted);
       ASSERT_FALSE(muted);
       fwrite(frame_.data(), sizeof(*frame_.data()),
@@ -180,7 +183,7 @@
         lost = true;
       }
 
-      if (FLAGS_verbose) {
+      if (FLAG_verbose) {
         if (!lost) {
           std::cout << "\nInserting packet number " << seq_num
               << " timestamp " << ts << std::endl;
@@ -279,13 +282,20 @@
 }  // 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::InsertPacketWithTiming test;
   test.SetUp();
 
   FILE* delay_log = NULL;
-  if (FLAGS_delay.size() > 0) {
-    delay_log = fopen(FLAGS_delay.c_str(), "wt");
+  if (strlen(FLAG_delay) > 0) {
+    delay_log = fopen(FLAG_delay, "wt");
     if (delay_log == NULL) {
       std::cout << "Cannot open the file to log delay values." << std::endl;
       exit(1);
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index a569bd1..204a988 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -724,7 +724,6 @@
         "../../rtc_base:protobuf_utils",
         "../../rtc_base:rtc_base_approved",
         "../../system_wrappers:system_wrappers_default",
-        "//third_party/gflags:gflags",
       ]
     }  # unpack_aecdump
 
@@ -755,7 +754,6 @@
         "aec_dump",
         "aec_dump:aec_dump_impl",
         "//testing/gtest",
-        "//third_party/gflags:gflags",
       ]
     }  # audioproc_f
   }
@@ -794,11 +792,11 @@
       "..:module_api",
       "../..:webrtc_common",
       "../../common_audio:common_audio",
+      "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:metrics_default",
       "../../system_wrappers:system_wrappers",
       "../../test:test_support",
       "//testing/gtest",
-      "//third_party/gflags",
     ]
   }
 
@@ -828,7 +826,6 @@
       "../../common_audio:common_audio",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:metrics_default",
-      "//third_party/gflags",
     ]
   }
 
@@ -841,10 +838,10 @@
       deps = [
         ":audio_processing",
         ":audioproc_test_utils",
+        "../../rtc_base:rtc_base_approved",
         "../../system_wrappers:metrics_default",
         "../../test:test_support",
         "//testing/gtest",
-        "//third_party/gflags",
       ]
     }
   }
diff --git a/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc b/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
index 3ac68d4..11b172a 100644
--- a/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
+++ b/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
@@ -10,12 +10,12 @@
 
 #include <vector>
 
-#include "gflags/gflags.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/rtc_base/format_macros.h"
 
 DEFINE_string(i, "", "The name of the input file to read from.");
@@ -24,6 +24,7 @@
     "Space delimited cartesian coordinates of microphones in meters. "
     "The coordinates of each point are contiguous. "
     "For a two element array: \"x1 y1 z1 x2 y2 z2\"");
+DEFINE_bool(help, false, "Prints this message.");
 
 namespace webrtc {
 namespace {
@@ -34,29 +35,36 @@
 const char kUsage[] =
     "Command-line tool to run beamforming on WAV files. The signal is passed\n"
     "in as a single band, unlike the audio processing interface which splits\n"
-    "signals into multiple bands.";
+    "signals into multiple bands.\n";
 
 }  // namespace
 
 int main(int argc, char* argv[]) {
-  google::SetUsageMessage(kUsage);
-  google::ParseCommandLineFlags(&argc, &argv, true);
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
+      FLAG_help || argc != 1) {
+    printf("%s", kUsage);
+    if (FLAG_help) {
+      rtc::FlagList::Print(nullptr, false);
+      return 0;
+    }
+    return 1;
+  }
 
-  WavReader in_file(FLAGS_i);
-  WavWriter out_file(FLAGS_o, in_file.sample_rate(), in_file.num_channels());
+  WavReader in_file(FLAG_i);
+  WavWriter out_file(FLAG_o, in_file.sample_rate(), in_file.num_channels());
 
   const size_t num_mics = in_file.num_channels();
   const std::vector<Point> array_geometry =
-      ParseArrayGeometry(FLAGS_mic_positions, num_mics);
+      ParseArrayGeometry(FLAG_mic_positions, num_mics);
   RTC_CHECK_EQ(array_geometry.size(), num_mics);
 
   NonlinearBeamformer bf(array_geometry, array_geometry.size());
   bf.Initialize(kChunkSizeMs, in_file.sample_rate());
 
   printf("Input file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
-         FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate());
+         FLAG_i, in_file.num_channels(), in_file.sample_rate());
   printf("Output file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
-         FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate());
+         FLAG_o, out_file.num_channels(), out_file.sample_rate());
 
   ChannelBuffer<float> buf(
       rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond),
diff --git a/modules/audio_processing/intelligibility/test/intelligibility_proc.cc b/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
index cd76a95..6045e9f 100644
--- a/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
+++ b/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
@@ -8,7 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "gflags/gflags.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/wav_file.h"
@@ -16,7 +15,7 @@
 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
 #include "webrtc/modules/audio_processing/noise_suppression_impl.h"
 #include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/test/gtest.h"
+#include "webrtc/rtc_base/flags.h"
 
 using std::complex;
 
@@ -26,16 +25,24 @@
 DEFINE_string(clear_file, "speech.wav", "Input file with clear speech.");
 DEFINE_string(noise_file, "noise.wav", "Input file with noise data.");
 DEFINE_string(out_file, "proc_enhanced.wav", "Enhanced output file.");
+DEFINE_bool(help, false, "Print this message.");
 
-// void function for gtest
-void void_main(int argc, char* argv[]) {
-  google::SetUsageMessage(
-      "\n\nInput files must be little-endian 16-bit signed raw PCM.\n");
-  google::ParseCommandLineFlags(&argc, &argv, true);
+int int_main(int argc, char* argv[]) {
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
+    return 1;
+  }
+  if (FLAG_help) {
+    rtc::FlagList::Print(nullptr, false);
+    return 0;
+  }
+  if (argc != 1) {
+    printf("\n\nInput files must be little-endian 16-bit signed raw PCM.\n");
+    return 0;
+  }
 
-  WavReader in_file(FLAGS_clear_file);
-  WavReader noise_file(FLAGS_noise_file);
-  WavWriter out_file(FLAGS_out_file, in_file.sample_rate(),
+  WavReader in_file(FLAG_clear_file);
+  WavReader noise_file(FLAG_noise_file);
+  WavWriter out_file(FLAG_out_file, in_file.sample_rate(),
                      in_file.num_channels());
   rtc::CriticalSection crit;
   NoiseSuppressionImpl ns(&crit);
@@ -77,12 +84,13 @@
     FloatToFloatS16(in.data(), in.size(), in.data());
     out_file.WriteSamples(in.data(), in.size());
   }
+
+  return 0;
 }
 
 }  // namespace
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
-  webrtc::void_main(argc, argv);
-  return 0;
+  return webrtc::int_main(argc, argv);
 }
diff --git a/modules/audio_processing/test/audioproc_float.cc b/modules/audio_processing/test/audioproc_float.cc
index a11f41c..bdf49d7 100644
--- a/modules/audio_processing/test/audioproc_float.cc
+++ b/modules/audio_processing/test/audioproc_float.cc
@@ -13,11 +13,11 @@
 
 #include <string.h>
 
-#include "gflags/gflags.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/aec_dump_based_simulator.h"
 #include "webrtc/modules/audio_processing/test/audio_processing_simulator.h"
 #include "webrtc/modules/audio_processing/test/wav_based_simulator.h"
+#include "webrtc/rtc_base/flags.h"
 
 namespace webrtc {
 namespace test {
@@ -32,7 +32,7 @@
     "\n\n"
     "Command-line tool to simulate a call using the audio "
     "processing module, either based on wav files or "
-    "protobuf debug dump recordings.";
+    "protobuf debug dump recordings.\n";
 
 DEFINE_string(dump_input, "", "Aec dump input filename");
 DEFINE_string(dump_output, "", "Aec dump output filename");
@@ -41,126 +41,126 @@
 DEFINE_string(ri, "", "Reverse stream input wav filename");
 DEFINE_string(ro, "", "Reverse stream output wav filename");
 DEFINE_string(artificial_nearend, "", "Artificial nearend wav filename");
-DEFINE_int32(output_num_channels,
-             kParameterNotSpecifiedValue,
-             "Number of forward stream output channels");
-DEFINE_int32(reverse_output_num_channels,
-             kParameterNotSpecifiedValue,
-             "Number of Reverse stream output channels");
-DEFINE_int32(output_sample_rate_hz,
-             kParameterNotSpecifiedValue,
-             "Forward stream output sample rate in Hz");
-DEFINE_int32(reverse_output_sample_rate_hz,
-             kParameterNotSpecifiedValue,
-             "Reverse stream output sample rate in Hz");
+DEFINE_int(output_num_channels,
+           kParameterNotSpecifiedValue,
+           "Number of forward stream output channels");
+DEFINE_int(reverse_output_num_channels,
+           kParameterNotSpecifiedValue,
+           "Number of Reverse stream output channels");
+DEFINE_int(output_sample_rate_hz,
+           kParameterNotSpecifiedValue,
+           "Forward stream output sample rate in Hz");
+DEFINE_int(reverse_output_sample_rate_hz,
+           kParameterNotSpecifiedValue,
+           "Reverse stream output sample rate in Hz");
 DEFINE_string(mic_positions,
               "",
               "Space delimited cartesian coordinates of microphones in "
               "meters. The coordinates of each point are contiguous. For a "
               "two element array: \"x1 y1 z1 x2 y2 z2\"");
-DEFINE_int32(target_angle_degrees,
-             90,
-             "The azimuth of the target in degrees (0-359). Only applies to "
-             "beamforming.");
+DEFINE_int(target_angle_degrees,
+           90,
+           "The azimuth of the target in degrees (0-359). Only applies to "
+           "beamforming.");
 DEFINE_bool(fixed_interface,
             false,
             "Use the fixed interface when operating on wav files");
-DEFINE_int32(aec,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the echo canceller");
-DEFINE_int32(aecm,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the mobile echo controller");
-DEFINE_int32(ed,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate (0) the residual echo detector");
+DEFINE_int(aec,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the echo canceller");
+DEFINE_int(aecm,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the mobile echo controller");
+DEFINE_int(ed,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate (0) the residual echo detector");
 DEFINE_string(ed_graph, "", "Output filename for graph of echo likelihood");
-DEFINE_int32(agc,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the AGC");
-DEFINE_int32(agc2,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the AGC2");
-DEFINE_int32(hpf,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the high-pass filter");
-DEFINE_int32(ns,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the noise suppressor");
-DEFINE_int32(ts,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the transient suppressor");
-DEFINE_int32(bf,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the beamformer");
-DEFINE_int32(ie,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the intelligibility enhancer");
-DEFINE_int32(vad,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the voice activity detector");
-DEFINE_int32(le,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the level estimator");
+DEFINE_int(agc,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the AGC");
+DEFINE_int(agc2,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the AGC2");
+DEFINE_int(hpf,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the high-pass filter");
+DEFINE_int(ns,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the noise suppressor");
+DEFINE_int(ts,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the transient suppressor");
+DEFINE_int(bf,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the beamformer");
+DEFINE_int(ie,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the intelligibility enhancer");
+DEFINE_int(vad,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the voice activity detector");
+DEFINE_int(le,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the level estimator");
 DEFINE_bool(all_default,
             false,
             "Activate all of the default components (will be overridden by any "
             "other settings)");
-DEFINE_int32(aec_suppression_level,
-             kParameterNotSpecifiedValue,
-             "Set the aec suppression level (0-2)");
-DEFINE_int32(delay_agnostic,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the AEC delay agnostic mode");
-DEFINE_int32(extended_filter,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the AEC extended filter mode");
-DEFINE_int32(drift_compensation,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the drift compensation");
-DEFINE_int32(aec3,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the experimental AEC mode AEC3");
-DEFINE_int32(lc,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the level control");
-DEFINE_int32(experimental_agc,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the experimental AGC");
-DEFINE_int32(
+DEFINE_int(aec_suppression_level,
+           kParameterNotSpecifiedValue,
+           "Set the aec suppression level (0-2)");
+DEFINE_int(delay_agnostic,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the AEC delay agnostic mode");
+DEFINE_int(extended_filter,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the AEC extended filter mode");
+DEFINE_int(drift_compensation,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the drift compensation");
+DEFINE_int(aec3,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the experimental AEC mode AEC3");
+DEFINE_int(lc,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the level control");
+DEFINE_int(experimental_agc,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the experimental AGC");
+DEFINE_int(
     refined_adaptive_filter,
     kParameterNotSpecifiedValue,
     "Activate (1) or deactivate(0) the refined adaptive filter functionality");
-DEFINE_int32(aecm_routing_mode,
-             kParameterNotSpecifiedValue,
-             "Specify the AECM routing mode (0-4)");
-DEFINE_int32(aecm_comfort_noise,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the AECM comfort noise");
-DEFINE_int32(agc_mode,
-             kParameterNotSpecifiedValue,
-             "Specify the AGC mode (0-2)");
-DEFINE_int32(agc_target_level,
-             kParameterNotSpecifiedValue,
-             "Specify the AGC target level (0-31)");
-DEFINE_int32(agc_limiter,
-             kParameterNotSpecifiedValue,
-             "Activate (1) or deactivate(0) the level estimator");
-DEFINE_int32(agc_compression_gain,
-             kParameterNotSpecifiedValue,
-             "Specify the AGC compression gain (0-90)");
-DEFINE_int32(vad_likelihood,
-             kParameterNotSpecifiedValue,
-             "Specify the VAD likelihood (0-3)");
-DEFINE_int32(ns_level,
-             kParameterNotSpecifiedValue,
-             "Specify the NS level (0-3)");
-DEFINE_int32(stream_delay,
-             kParameterNotSpecifiedValue,
-             "Specify the stream delay in ms to use");
-DEFINE_int32(stream_drift_samples,
-             kParameterNotSpecifiedValue,
-             "Specify the number of stream drift samples to use");
+DEFINE_int(aecm_routing_mode,
+           kParameterNotSpecifiedValue,
+           "Specify the AECM routing mode (0-4)");
+DEFINE_int(aecm_comfort_noise,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the AECM comfort noise");
+DEFINE_int(agc_mode,
+           kParameterNotSpecifiedValue,
+           "Specify the AGC mode (0-2)");
+DEFINE_int(agc_target_level,
+           kParameterNotSpecifiedValue,
+           "Specify the AGC target level (0-31)");
+DEFINE_int(agc_limiter,
+           kParameterNotSpecifiedValue,
+           "Activate (1) or deactivate(0) the level estimator");
+DEFINE_int(agc_compression_gain,
+           kParameterNotSpecifiedValue,
+           "Specify the AGC compression gain (0-90)");
+DEFINE_int(vad_likelihood,
+           kParameterNotSpecifiedValue,
+           "Specify the VAD likelihood (0-3)");
+DEFINE_int(ns_level,
+           kParameterNotSpecifiedValue,
+           "Specify the NS level (0-3)");
+DEFINE_int(stream_delay,
+           kParameterNotSpecifiedValue,
+           "Specify the stream delay in ms to use");
+DEFINE_int(stream_drift_samples,
+           kParameterNotSpecifiedValue,
+           "Specify the number of stream drift samples to use");
 DEFINE_bool(performance_report, false, "Report the APM performance ");
 DEFINE_bool(verbose, false, "Produce verbose output");
 DEFINE_bool(bitexactness_report,
@@ -173,6 +173,7 @@
             false,
             "Creates new output files after each init");
 DEFINE_string(custom_call_order_file, "", "Custom process API call order file");
+DEFINE_bool(help, false, "Print this message");
 
 void SetSettingIfSpecified(const std::string value,
                            rtc::Optional<std::string>* parameter) {
@@ -197,7 +198,7 @@
 
 SimulationSettings CreateSettings() {
   SimulationSettings settings;
-  if (FLAGS_all_default) {
+  if (FLAG_all_default) {
     settings.use_le = rtc::Optional<bool>(true);
     settings.use_vad = rtc::Optional<bool>(true);
     settings.use_ie = rtc::Optional<bool>(false);
@@ -210,70 +211,70 @@
     settings.use_aecm = rtc::Optional<bool>(false);
     settings.use_ed = rtc::Optional<bool>(false);
   }
-  SetSettingIfSpecified(FLAGS_dump_input, &settings.aec_dump_input_filename);
-  SetSettingIfSpecified(FLAGS_dump_output, &settings.aec_dump_output_filename);
-  SetSettingIfSpecified(FLAGS_i, &settings.input_filename);
-  SetSettingIfSpecified(FLAGS_o, &settings.output_filename);
-  SetSettingIfSpecified(FLAGS_ri, &settings.reverse_input_filename);
-  SetSettingIfSpecified(FLAGS_ro, &settings.reverse_output_filename);
-  SetSettingIfSpecified(FLAGS_artificial_nearend,
+  SetSettingIfSpecified(FLAG_dump_input, &settings.aec_dump_input_filename);
+  SetSettingIfSpecified(FLAG_dump_output, &settings.aec_dump_output_filename);
+  SetSettingIfSpecified(FLAG_i, &settings.input_filename);
+  SetSettingIfSpecified(FLAG_o, &settings.output_filename);
+  SetSettingIfSpecified(FLAG_ri, &settings.reverse_input_filename);
+  SetSettingIfSpecified(FLAG_ro, &settings.reverse_output_filename);
+  SetSettingIfSpecified(FLAG_artificial_nearend,
                         &settings.artificial_nearend_filename);
-  SetSettingIfSpecified(FLAGS_output_num_channels,
+  SetSettingIfSpecified(FLAG_output_num_channels,
                         &settings.output_num_channels);
-  SetSettingIfSpecified(FLAGS_reverse_output_num_channels,
+  SetSettingIfSpecified(FLAG_reverse_output_num_channels,
                         &settings.reverse_output_num_channels);
-  SetSettingIfSpecified(FLAGS_output_sample_rate_hz,
+  SetSettingIfSpecified(FLAG_output_sample_rate_hz,
                         &settings.output_sample_rate_hz);
-  SetSettingIfSpecified(FLAGS_reverse_output_sample_rate_hz,
+  SetSettingIfSpecified(FLAG_reverse_output_sample_rate_hz,
                         &settings.reverse_output_sample_rate_hz);
-  SetSettingIfSpecified(FLAGS_mic_positions, &settings.microphone_positions);
-  settings.target_angle_degrees = FLAGS_target_angle_degrees;
-  SetSettingIfFlagSet(FLAGS_aec, &settings.use_aec);
-  SetSettingIfFlagSet(FLAGS_aecm, &settings.use_aecm);
-  SetSettingIfFlagSet(FLAGS_ed, &settings.use_ed);
-  SetSettingIfSpecified(FLAGS_ed_graph, &settings.ed_graph_output_filename);
-  SetSettingIfFlagSet(FLAGS_agc, &settings.use_agc);
-  SetSettingIfFlagSet(FLAGS_agc2, &settings.use_agc2);
-  SetSettingIfFlagSet(FLAGS_hpf, &settings.use_hpf);
-  SetSettingIfFlagSet(FLAGS_ns, &settings.use_ns);
-  SetSettingIfFlagSet(FLAGS_ts, &settings.use_ts);
-  SetSettingIfFlagSet(FLAGS_bf, &settings.use_bf);
-  SetSettingIfFlagSet(FLAGS_ie, &settings.use_ie);
-  SetSettingIfFlagSet(FLAGS_vad, &settings.use_vad);
-  SetSettingIfFlagSet(FLAGS_le, &settings.use_le);
-  SetSettingIfSpecified(FLAGS_aec_suppression_level,
+  SetSettingIfSpecified(FLAG_mic_positions, &settings.microphone_positions);
+  settings.target_angle_degrees = FLAG_target_angle_degrees;
+  SetSettingIfFlagSet(FLAG_aec, &settings.use_aec);
+  SetSettingIfFlagSet(FLAG_aecm, &settings.use_aecm);
+  SetSettingIfFlagSet(FLAG_ed, &settings.use_ed);
+  SetSettingIfSpecified(FLAG_ed_graph, &settings.ed_graph_output_filename);
+  SetSettingIfFlagSet(FLAG_agc, &settings.use_agc);
+  SetSettingIfFlagSet(FLAG_agc2, &settings.use_agc2);
+  SetSettingIfFlagSet(FLAG_hpf, &settings.use_hpf);
+  SetSettingIfFlagSet(FLAG_ns, &settings.use_ns);
+  SetSettingIfFlagSet(FLAG_ts, &settings.use_ts);
+  SetSettingIfFlagSet(FLAG_bf, &settings.use_bf);
+  SetSettingIfFlagSet(FLAG_ie, &settings.use_ie);
+  SetSettingIfFlagSet(FLAG_vad, &settings.use_vad);
+  SetSettingIfFlagSet(FLAG_le, &settings.use_le);
+  SetSettingIfSpecified(FLAG_aec_suppression_level,
                         &settings.aec_suppression_level);
-  SetSettingIfFlagSet(FLAGS_delay_agnostic, &settings.use_delay_agnostic);
-  SetSettingIfFlagSet(FLAGS_extended_filter, &settings.use_extended_filter);
-  SetSettingIfFlagSet(FLAGS_drift_compensation,
+  SetSettingIfFlagSet(FLAG_delay_agnostic, &settings.use_delay_agnostic);
+  SetSettingIfFlagSet(FLAG_extended_filter, &settings.use_extended_filter);
+  SetSettingIfFlagSet(FLAG_drift_compensation,
                       &settings.use_drift_compensation);
-  SetSettingIfFlagSet(FLAGS_refined_adaptive_filter,
+  SetSettingIfFlagSet(FLAG_refined_adaptive_filter,
                       &settings.use_refined_adaptive_filter);
 
-  SetSettingIfFlagSet(FLAGS_aec3, &settings.use_aec3);
-  SetSettingIfFlagSet(FLAGS_lc, &settings.use_lc);
-  SetSettingIfFlagSet(FLAGS_experimental_agc, &settings.use_experimental_agc);
-  SetSettingIfSpecified(FLAGS_aecm_routing_mode, &settings.aecm_routing_mode);
-  SetSettingIfFlagSet(FLAGS_aecm_comfort_noise,
+  SetSettingIfFlagSet(FLAG_aec3, &settings.use_aec3);
+  SetSettingIfFlagSet(FLAG_lc, &settings.use_lc);
+  SetSettingIfFlagSet(FLAG_experimental_agc, &settings.use_experimental_agc);
+  SetSettingIfSpecified(FLAG_aecm_routing_mode, &settings.aecm_routing_mode);
+  SetSettingIfFlagSet(FLAG_aecm_comfort_noise,
                       &settings.use_aecm_comfort_noise);
-  SetSettingIfSpecified(FLAGS_agc_mode, &settings.agc_mode);
-  SetSettingIfSpecified(FLAGS_agc_target_level, &settings.agc_target_level);
-  SetSettingIfFlagSet(FLAGS_agc_limiter, &settings.use_agc_limiter);
-  SetSettingIfSpecified(FLAGS_agc_compression_gain,
+  SetSettingIfSpecified(FLAG_agc_mode, &settings.agc_mode);
+  SetSettingIfSpecified(FLAG_agc_target_level, &settings.agc_target_level);
+  SetSettingIfFlagSet(FLAG_agc_limiter, &settings.use_agc_limiter);
+  SetSettingIfSpecified(FLAG_agc_compression_gain,
                         &settings.agc_compression_gain);
-  SetSettingIfSpecified(FLAGS_vad_likelihood, &settings.vad_likelihood);
-  SetSettingIfSpecified(FLAGS_ns_level, &settings.ns_level);
-  SetSettingIfSpecified(FLAGS_stream_delay, &settings.stream_delay);
-  SetSettingIfSpecified(FLAGS_stream_drift_samples,
+  SetSettingIfSpecified(FLAG_vad_likelihood, &settings.vad_likelihood);
+  SetSettingIfSpecified(FLAG_ns_level, &settings.ns_level);
+  SetSettingIfSpecified(FLAG_stream_delay, &settings.stream_delay);
+  SetSettingIfSpecified(FLAG_stream_drift_samples,
                         &settings.stream_drift_samples);
-  SetSettingIfSpecified(FLAGS_custom_call_order_file,
+  SetSettingIfSpecified(FLAG_custom_call_order_file,
                         &settings.custom_call_order_filename);
-  settings.report_performance = FLAGS_performance_report;
-  settings.use_verbose_logging = FLAGS_verbose;
-  settings.report_bitexactness = FLAGS_bitexactness_report;
-  settings.discard_all_settings_in_aecdump = FLAGS_discard_settings_in_aecdump;
-  settings.fixed_interface = FLAGS_fixed_interface;
-  settings.store_intermediate_output = FLAGS_store_intermediate_output;
+  settings.report_performance = FLAG_performance_report;
+  settings.use_verbose_logging = FLAG_verbose;
+  settings.report_bitexactness = FLAG_bitexactness_report;
+  settings.discard_all_settings_in_aecdump = FLAG_discard_settings_in_aecdump;
+  settings.fixed_interface = FLAG_fixed_interface;
+  settings.store_intermediate_output = FLAG_store_intermediate_output;
 
   return settings;
 }
@@ -422,8 +423,15 @@
 }  // namespace
 
 int main(int argc, char* argv[]) {
-  google::SetUsageMessage(kUsageDescription);
-  google::ParseCommandLineFlags(&argc, &argv, true);
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
+      FLAG_help || argc != 1) {
+    printf("%s", kUsageDescription);
+    if (FLAG_help) {
+      rtc::FlagList::Print(nullptr, false);
+      return 0;
+    }
+    return 1;
+  }
 
   SimulationSettings settings = CreateSettings();
   PerformBasicParameterSanityChecks(settings);
diff --git a/modules/audio_processing/test/conversational_speech/BUILD.gn b/modules/audio_processing/test/conversational_speech/BUILD.gn
index 587663b..5c681a0 100644
--- a/modules/audio_processing/test/conversational_speech/BUILD.gn
+++ b/modules/audio_processing/test/conversational_speech/BUILD.gn
@@ -24,7 +24,6 @@
     ":lib",
     "../../../../rtc_base:rtc_base_approved",
     "../../../../test:test_support",
-    "//third_party/gflags",
   ]
 }
 
diff --git a/modules/audio_processing/test/conversational_speech/generator.cc b/modules/audio_processing/test/conversational_speech/generator.cc
index c7f86de..53f7217 100644
--- a/modules/audio_processing/test/conversational_speech/generator.cc
+++ b/modules/audio_processing/test/conversational_speech/generator.cc
@@ -10,12 +10,12 @@
 
 #include <iostream>
 
-#include "gflags/gflags.h"
 #include "webrtc/modules/audio_processing/test/conversational_speech/config.h"
 #include "webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h"
 #include "webrtc/modules/audio_processing/test/conversational_speech/simulator.h"
 #include "webrtc/modules/audio_processing/test/conversational_speech/timing.h"
 #include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -23,14 +23,6 @@
 namespace test {
 namespace {
 
-// Adapting DirExists/FileExists interfaces to DEFINE_validator.
-auto dir_exists = [](const char* c, const std::string& dirpath) {
-  return DirExists(dirpath);
-};
-auto file_exists = [](const char* c, const std::string& filepath) {
-  return FileExists(filepath);
-};
-
 const char kUsageDescription[] =
     "Usage: conversational_speech_generator\n"
     "          -i <path/to/source/audiotracks>\n"
@@ -38,21 +30,30 @@
     "          -o <output/path>\n"
     "\n\n"
     "Command-line tool to generate multiple-end audio tracks to simulate "
-    "conversational speech with two or more participants.";
+    "conversational speech with two or more participants.\n";
 
 DEFINE_string(i, "", "Directory containing the speech turn wav files");
-DEFINE_validator(i, dir_exists);
 DEFINE_string(t, "", "Path to the timing text file");
-DEFINE_validator(t, file_exists);
 DEFINE_string(o, "", "Output wav files destination path");
-DEFINE_validator(o, dir_exists);
+DEFINE_bool(help, false, "Prints this message");
 
 }  // namespace
 
 int main(int argc, char* argv[]) {
-  google::SetUsageMessage(kUsageDescription);
-  google::ParseCommandLineFlags(&argc, &argv, true);
-  conversational_speech::Config config(FLAGS_i, FLAGS_t, FLAGS_o);
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
+      FLAG_help || argc != 1) {
+    printf("%s", kUsageDescription);
+    if (FLAG_help) {
+      rtc::FlagList::Print(nullptr, false);
+      return 0;
+    }
+    return 1;
+  }
+  RTC_CHECK(DirExists(FLAG_i));
+  RTC_CHECK(FileExists(FLAG_t));
+  RTC_CHECK(DirExists(FLAG_o));
+
+  conversational_speech::Config config(FLAG_i, FLAG_t, FLAG_o);
 
   // Load timing.
   std::vector<conversational_speech::Turn> timing =
diff --git a/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_gencfgs.py b/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_gencfgs.py
index d2764b0..338c38c 100755
--- a/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_gencfgs.py
+++ b/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_gencfgs.py
@@ -27,7 +27,7 @@
 
   The default settings are loaded via "-all_default".
   Check "src/webrtc/modules/audio_processing/test/audioproc_float.cc" and search
-  for "if (FLAGS_all_default) {".
+  for "if (FLAG_all_default) {".
 
   For instance, in 55eb6d621489730084927868fed195d3645a9ec9 the default is this:
   settings.use_aec = rtc::Optional<bool>(true);
diff --git a/modules/audio_processing/test/unpack.cc b/modules/audio_processing/test/unpack.cc
index 4c6c18d..13be279 100644
--- a/modules/audio_processing/test/unpack.cc
+++ b/modules/audio_processing/test/unpack.cc
@@ -17,9 +17,9 @@
 
 #include <memory>
 
-#include "gflags/gflags.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/rtc_base/format_macros.h"
 #include "webrtc/rtc_base/ignore_wundef.h"
 #include "webrtc/typedefs.h"
@@ -45,6 +45,7 @@
 DEFINE_bool(text,
             false,
             "Write non-audio files as text files instead of binary files.");
+DEFINE_bool(help, false, "Print this message.");
 
 #define PRINT_CONFIG(field_name) \
   if (msg.has_##field_name()) { \
@@ -70,11 +71,14 @@
   std::string program_name = argv[0];
   std::string usage = "Commandline tool to unpack audioproc debug files.\n"
     "Example usage:\n" + program_name + " debug_dump.pb\n";
-  google::SetUsageMessage(usage);
-  google::ParseCommandLineFlags(&argc, &argv, true);
 
-  if (argc < 2) {
-    printf("%s", google::ProgramUsage());
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
+      FLAG_help || argc < 2) {
+    printf("%s", usage.c_str());
+    if (FLAG_help) {
+      rtc::FlagList::Print(nullptr, false);
+      return 0;
+    }
     return 1;
   }
 
@@ -95,7 +99,7 @@
   std::unique_ptr<RawFile> input_raw_file;
   std::unique_ptr<RawFile> output_raw_file;
 
-  FILE* settings_file = OpenFile(FLAGS_settings_file, "wb");
+  FILE* settings_file = OpenFile(FLAG_settings_file, "wb");
 
   while (ReadMessageFromFile(debug_file, &event_msg)) {
     if (event_msg.type() == Event::REVERSE_STREAM) {
@@ -106,8 +110,9 @@
 
       const ReverseStream msg = event_msg.reverse_stream();
       if (msg.has_data()) {
-        if (FLAGS_raw && !reverse_raw_file) {
-          reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".pcm"));
+        if (FLAG_raw && !reverse_raw_file) {
+          reverse_raw_file.reset(new RawFile(std::string(FLAG_reverse_file) +
+                                             ".pcm"));
         }
         // TODO(aluebs): Replace "num_reverse_channels *
         // reverse_samples_per_channel" with "msg.data().size() /
@@ -118,8 +123,9 @@
                      reverse_wav_file.get(),
                      reverse_raw_file.get());
       } else if (msg.channel_size() > 0) {
-        if (FLAGS_raw && !reverse_raw_file) {
-          reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".float"));
+        if (FLAG_raw && !reverse_raw_file) {
+          reverse_raw_file.reset(new RawFile(std::string(FLAG_reverse_file) +
+                                             ".float"));
         }
         std::unique_ptr<const float* []> data(
             new const float* [num_reverse_channels]);
@@ -141,16 +147,18 @@
 
       const Stream msg = event_msg.stream();
       if (msg.has_input_data()) {
-        if (FLAGS_raw && !input_raw_file) {
-          input_raw_file.reset(new RawFile(FLAGS_input_file + ".pcm"));
+        if (FLAG_raw && !input_raw_file) {
+          input_raw_file.reset(new RawFile(std::string(FLAG_input_file) +
+                                           ".pcm"));
         }
         WriteIntData(reinterpret_cast<const int16_t*>(msg.input_data().data()),
                      num_input_channels * input_samples_per_channel,
                      input_wav_file.get(),
                      input_raw_file.get());
       } else if (msg.input_channel_size() > 0) {
-        if (FLAGS_raw && !input_raw_file) {
-          input_raw_file.reset(new RawFile(FLAGS_input_file + ".float"));
+        if (FLAG_raw && !input_raw_file) {
+          input_raw_file.reset(new RawFile(std::string(FLAG_input_file) +
+                                           ".float"));
         }
         std::unique_ptr<const float* []> data(
             new const float* [num_input_channels]);
@@ -165,16 +173,18 @@
       }
 
       if (msg.has_output_data()) {
-        if (FLAGS_raw && !output_raw_file) {
-          output_raw_file.reset(new RawFile(FLAGS_output_file + ".pcm"));
+        if (FLAG_raw && !output_raw_file) {
+          output_raw_file.reset(new RawFile(std::string(FLAG_output_file) +
+                                            ".pcm"));
         }
         WriteIntData(reinterpret_cast<const int16_t*>(msg.output_data().data()),
                      num_output_channels * output_samples_per_channel,
                      output_wav_file.get(),
                      output_raw_file.get());
       } else if (msg.output_channel_size() > 0) {
-        if (FLAGS_raw && !output_raw_file) {
-          output_raw_file.reset(new RawFile(FLAGS_output_file + ".float"));
+        if (FLAG_raw && !output_raw_file) {
+          output_raw_file.reset(new RawFile(std::string(FLAG_output_file) +
+                                            ".float"));
         }
         std::unique_ptr<const float* []> data(
             new const float* [num_output_channels]);
@@ -189,45 +199,45 @@
                        output_raw_file.get());
       }
 
-      if (FLAGS_full) {
+      if (FLAG_full) {
         if (msg.has_delay()) {
-          static FILE* delay_file = OpenFile(FLAGS_delay_file, "wb");
+          static FILE* delay_file = OpenFile(FLAG_delay_file, "wb");
           int32_t delay = msg.delay();
-          if (FLAGS_text) {
+          if (FLAG_text) {
             fprintf(delay_file, "%d\n", delay);
           } else {
-            WriteData(&delay, sizeof(delay), delay_file, FLAGS_delay_file);
+            WriteData(&delay, sizeof(delay), delay_file, FLAG_delay_file);
           }
         }
 
         if (msg.has_drift()) {
-          static FILE* drift_file = OpenFile(FLAGS_drift_file, "wb");
+          static FILE* drift_file = OpenFile(FLAG_drift_file, "wb");
           int32_t drift = msg.drift();
-          if (FLAGS_text) {
+          if (FLAG_text) {
             fprintf(drift_file, "%d\n", drift);
           } else {
-            WriteData(&drift, sizeof(drift), drift_file, FLAGS_drift_file);
+            WriteData(&drift, sizeof(drift), drift_file, FLAG_drift_file);
           }
         }
 
         if (msg.has_level()) {
-          static FILE* level_file = OpenFile(FLAGS_level_file, "wb");
+          static FILE* level_file = OpenFile(FLAG_level_file, "wb");
           int32_t level = msg.level();
-          if (FLAGS_text) {
+          if (FLAG_text) {
             fprintf(level_file, "%d\n", level);
           } else {
-            WriteData(&level, sizeof(level), level_file, FLAGS_level_file);
+            WriteData(&level, sizeof(level), level_file, FLAG_level_file);
           }
         }
 
         if (msg.has_keypress()) {
-          static FILE* keypress_file = OpenFile(FLAGS_keypress_file, "wb");
+          static FILE* keypress_file = OpenFile(FLAG_keypress_file, "wb");
           bool keypress = msg.keypress();
-          if (FLAGS_text) {
+          if (FLAG_text) {
             fprintf(keypress_file, "%d\n", keypress);
           } else {
             WriteData(&keypress, sizeof(keypress), keypress_file,
-                      FLAGS_keypress_file);
+                      FLAG_keypress_file);
           }
         }
       }
@@ -304,21 +314,21 @@
       output_samples_per_channel =
           static_cast<size_t>(output_sample_rate / 100);
 
-      if (!FLAGS_raw) {
+      if (!FLAG_raw) {
         // The WAV files need to be reset every time, because they cant change
         // their sample rate or number of channels.
         std::stringstream reverse_name;
-        reverse_name << FLAGS_reverse_file << frame_count << ".wav";
+        reverse_name << FLAG_reverse_file << frame_count << ".wav";
         reverse_wav_file.reset(new WavWriter(reverse_name.str(),
                                              reverse_sample_rate,
                                              num_reverse_channels));
         std::stringstream input_name;
-        input_name << FLAGS_input_file << frame_count << ".wav";
+        input_name << FLAG_input_file << frame_count << ".wav";
         input_wav_file.reset(new WavWriter(input_name.str(),
                                            input_sample_rate,
                                            num_input_channels));
         std::stringstream output_name;
-        output_name << FLAGS_output_file << frame_count << ".wav";
+        output_name << FLAG_output_file << frame_count << ".wav";
         output_wav_file.reset(new WavWriter(output_name.str(),
                                             output_sample_rate,
                                             num_output_channels));
diff --git a/modules/audio_processing/transient/transient_suppression_test.cc b/modules/audio_processing/transient/transient_suppression_test.cc
index 2158a81..ea894bc 100644
--- a/modules/audio_processing/transient/transient_suppression_test.cc
+++ b/modules/audio_processing/transient/transient_suppression_test.cc
@@ -12,14 +12,15 @@
 
 #include <stdlib.h>
 #include <stdio.h>
+#include <string.h>
 
 #include <memory>
 #include <string>
 
-#include "gflags/gflags.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/modules/audio_processing/agc/agc.h"
 #include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
@@ -32,31 +33,20 @@
               "",
               "PCM file that contains the reference signal.");
 
-static bool ValidatePositiveInt(const char* flagname, int32_t value) {
-  if (value <= 0) {
-    printf("%s must be a positive integer.\n", flagname);
-    return false;
-  }
-  return true;
-}
-DEFINE_int32(chunk_size_ms,
-             10,
-             "Time between each chunk of samples in milliseconds.");
-static const bool chunk_size_ms_dummy =
-    google::RegisterFlagValidator(&FLAGS_chunk_size_ms, &ValidatePositiveInt);
+DEFINE_int(chunk_size_ms,
+           10,
+           "Time between each chunk of samples in milliseconds.");
 
-DEFINE_int32(sample_rate_hz,
-             16000,
-             "Sampling frequency of the signal in Hertz.");
-static const bool sample_rate_hz_dummy =
-    google::RegisterFlagValidator(&FLAGS_sample_rate_hz, &ValidatePositiveInt);
-DEFINE_int32(detection_rate_hz,
-             0,
-             "Sampling frequency of the detection signal in Hertz.");
+DEFINE_int(sample_rate_hz,
+           16000,
+           "Sampling frequency of the signal in Hertz.");
+DEFINE_int(detection_rate_hz,
+           0,
+           "Sampling frequency of the detection signal in Hertz.");
 
-DEFINE_int32(num_channels, 1, "Number of channels.");
-static const bool num_channels_dummy =
-    google::RegisterFlagValidator(&FLAGS_num_channels, &ValidatePositiveInt);
+DEFINE_int(num_channels, 1, "Number of channels.");
+
+DEFINE_bool(help, false, "Print this message.");
 
 namespace webrtc {
 
@@ -146,19 +136,19 @@
 void void_main() {
   // TODO(aluebs): Remove all FileWrappers.
   // Prepare the input file.
-  FILE* in_file = fopen(FLAGS_in_file_name.c_str(), "rb");
+  FILE* in_file = fopen(FLAG_in_file_name, "rb");
   ASSERT_TRUE(in_file != NULL);
 
   // Prepare the detection file.
   FILE* detection_file = NULL;
-  if (!FLAGS_detection_file_name.empty()) {
-    detection_file = fopen(FLAGS_detection_file_name.c_str(), "rb");
+  if (strlen(FLAG_detection_file_name) > 0) {
+    detection_file = fopen(FLAG_detection_file_name, "rb");
   }
 
   // Prepare the reference file.
   FILE* reference_file = NULL;
-  if (!FLAGS_reference_file_name.empty()) {
-    reference_file = fopen(FLAGS_reference_file_name.c_str(), "rb");
+  if (strlen(FLAG_reference_file_name) > 0) {
+    reference_file = fopen(FLAG_reference_file_name, "rb");
   }
 
   // Prepare the output file.
@@ -166,27 +156,27 @@
   FILE* out_file = fopen(out_file_name.c_str(), "wb");
   ASSERT_TRUE(out_file != NULL);
 
-  int detection_rate_hz = FLAGS_detection_rate_hz;
+  int detection_rate_hz = FLAG_detection_rate_hz;
   if (detection_rate_hz == 0) {
-    detection_rate_hz = FLAGS_sample_rate_hz;
+    detection_rate_hz = FLAG_sample_rate_hz;
   }
 
   Agc agc;
 
   TransientSuppressor suppressor;
   suppressor.Initialize(
-      FLAGS_sample_rate_hz, detection_rate_hz, FLAGS_num_channels);
+      FLAG_sample_rate_hz, detection_rate_hz, FLAG_num_channels);
 
   const size_t audio_buffer_size =
-      FLAGS_chunk_size_ms * FLAGS_sample_rate_hz / 1000;
+      FLAG_chunk_size_ms * FLAG_sample_rate_hz / 1000;
   const size_t detection_buffer_size =
-      FLAGS_chunk_size_ms * detection_rate_hz / 1000;
+      FLAG_chunk_size_ms * detection_rate_hz / 1000;
 
   // int16 and float variants of the same data.
   std::unique_ptr<int16_t[]> audio_buffer_i(
-      new int16_t[FLAGS_num_channels * audio_buffer_size]);
+      new int16_t[FLAG_num_channels * audio_buffer_size]);
   std::unique_ptr<float[]> audio_buffer_f(
-      new float[FLAGS_num_channels * audio_buffer_size]);
+      new float[FLAG_num_channels * audio_buffer_size]);
 
   std::unique_ptr<float[]> detection_buffer, reference_buffer;
 
@@ -197,7 +187,7 @@
 
   while (ReadBuffers(in_file,
                      audio_buffer_size,
-                     FLAGS_num_channels,
+                     FLAG_num_channels,
                      audio_buffer_i.get(),
                      detection_file,
                      detection_buffer_size,
@@ -207,17 +197,17 @@
     ASSERT_EQ(0,
               agc.Process(audio_buffer_i.get(),
                           static_cast<int>(audio_buffer_size),
-                          FLAGS_sample_rate_hz))
+                          FLAG_sample_rate_hz))
         << "The AGC could not process the frame";
 
-    for (size_t i = 0; i < FLAGS_num_channels * audio_buffer_size; ++i) {
+    for (size_t i = 0; i < FLAG_num_channels * audio_buffer_size; ++i) {
       audio_buffer_f[i] = audio_buffer_i[i];
     }
 
     ASSERT_EQ(0,
               suppressor.Suppress(audio_buffer_f.get(),
                                   audio_buffer_size,
-                                  FLAGS_num_channels,
+                                  FLAG_num_channels,
                                   detection_buffer.get(),
                                   detection_buffer_size,
                                   reference_buffer.get(),
@@ -228,7 +218,7 @@
 
     // Write result to out file.
     WritePCM(
-        out_file, audio_buffer_size, FLAGS_num_channels, audio_buffer_f.get());
+        out_file, audio_buffer_size, FLAG_num_channels, audio_buffer_f.get());
   }
 
   fclose(in_file);
@@ -244,8 +234,19 @@
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
-  google::SetUsageMessage(webrtc::kUsage);
-  google::ParseCommandLineFlags(&argc, &argv, true);
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
+      FLAG_help || argc != 1) {
+    printf("%s", webrtc::kUsage);
+    if (FLAG_help) {
+      rtc::FlagList::Print(nullptr, false);
+      return 0;
+    }
+    return 1;
+  }
+  RTC_CHECK_GT(FLAG_chunk_size_ms, 0);
+  RTC_CHECK_GT(FLAG_sample_rate_hz, 0);
+  RTC_CHECK_GT(FLAG_num_channels, 0);
+
   webrtc::void_main();
   return 0;
 }
diff --git a/modules/remote_bitrate_estimator/BUILD.gn b/modules/remote_bitrate_estimator/BUILD.gn
index 2de7830..f13e8fc 100644
--- a/modules/remote_bitrate_estimator/BUILD.gn
+++ b/modules/remote_bitrate_estimator/BUILD.gn
@@ -225,7 +225,6 @@
       "../../test:test_main",
       "//testing/gmock",
       "//testing/gtest",
-      "//third_party/gflags",
     ]
 
     if (!build_with_chromium && is_clang) {
diff --git a/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
index 2b2e265..de0d5ea 100644
--- a/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
+++ b/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
@@ -16,11 +16,11 @@
 #include <sstream>
 #include <string>
 
-#include "gflags/gflags.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 namespace flags {
@@ -30,17 +30,17 @@
               "Extension type, either abs for absolute send time or tsoffset "
               "for timestamp offset.");
 std::string ExtensionType() {
-  return static_cast<std::string>(FLAGS_extension_type);
+  return static_cast<std::string>(FLAG_extension_type);
 }
 
-DEFINE_int32(extension_id, 3, "Extension id.");
+DEFINE_int(extension_id, 3, "Extension id.");
 int ExtensionId() {
-  return static_cast<int>(FLAGS_extension_id);
+  return static_cast<int>(FLAG_extension_id);
 }
 
 DEFINE_string(input_file, "", "Input file.");
 std::string InputFile() {
-  return static_cast<std::string>(FLAGS_input_file);
+  return static_cast<std::string>(FLAG_input_file);
 }
 
 DEFINE_string(ssrc_filter,
@@ -48,7 +48,7 @@
               "Comma-separated list of SSRCs in hexadecimal which are to be "
               "used as input to the BWE (only applicable to pcap files).");
 std::set<uint32_t> SsrcFilter() {
-  std::string ssrc_filter_string = static_cast<std::string>(FLAGS_ssrc_filter);
+  std::string ssrc_filter_string = static_cast<std::string>(FLAG_ssrc_filter);
   if (ssrc_filter_string.empty())
     return std::set<uint32_t>();
   std::stringstream ss;
@@ -64,6 +64,8 @@
   }
   return ssrcs;
 }
+
+DEFINE_bool(help, false, "Print this message.");
 }  // namespace flags
 
 bool ParseArgsAndSetupEstimator(int argc,
@@ -74,7 +76,13 @@
                                 webrtc::RtpHeaderParser** parser,
                                 webrtc::RemoteBitrateEstimator** estimator,
                                 std::string* estimator_used) {
-  google::ParseCommandLineFlags(&argc, &argv, true);
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
+    return 1;
+  }
+  if (FLAG_help) {
+    rtc::FlagList::Print(nullptr, false);
+    return 0;
+  }
   std::string filename = flags::InputFile();
 
   std::set<uint32_t> ssrc_filter = flags::SsrcFilter();
diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index 12f2f36..d4fb6e4 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -350,7 +350,6 @@
       "../../test:video_test_common",
       "../../test:video_test_support",
       "../video_capture",
-      "//third_party/gflags",
     ]
   }  # video_quality_measurement
 
diff --git a/modules/video_coding/codecs/tools/video_quality_measurement.cc b/modules/video_coding/codecs/tools/video_quality_measurement.cc
index 48929b9..f82ba42 100644
--- a/modules/video_coding/codecs/tools/video_quality_measurement.cc
+++ b/modules/video_coding/codecs/tools/video_quality_measurement.cc
@@ -11,6 +11,7 @@
 #include <assert.h>
 #include <stdio.h>
 #include <time.h>
+#include <string.h>
 
 #include <stdarg.h>
 #include <sys/stat.h>  // To check for directory existence.
@@ -19,13 +20,13 @@
 #define S_ISDIR(mode) (((mode)&S_IFMT) == S_IFDIR)
 #endif
 
-#include "gflags/gflags.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
 #include "webrtc/modules/video_coding/codecs/test/stats.h"
 #include "webrtc/modules/video_coding/codecs/test/videoprocessor.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/rtc_base/format_macros.h"
 #include "webrtc/test/testsupport/frame_reader.h"
 #include "webrtc/test/testsupport/frame_writer.h"
@@ -43,12 +44,12 @@
               "Input file. "
               "The source video file to be encoded and decoded. Must be in "
               ".yuv format");
-DEFINE_int32(width, -1, "Width in pixels of the frames in the input file.");
-DEFINE_int32(height, -1, "Height in pixels of the frames in the input file.");
-DEFINE_int32(framerate,
-             30,
-             "Frame rate of the input file, in FPS "
-             "(frames-per-second). ");
+DEFINE_int(width, -1, "Width in pixels of the frames in the input file.");
+DEFINE_int(height, -1, "Height in pixels of the frames in the input file.");
+DEFINE_int(framerate,
+           30,
+           "Frame rate of the input file, in FPS "
+           "(frames-per-second). ");
 DEFINE_string(output_dir,
               ".",
               "Output directory. "
@@ -76,40 +77,40 @@
               "The name of the output video file resulting of the processing "
               "of the source file. By default this is the same name as the "
               "input file with '_out' appended before the extension.");
-DEFINE_int32(bitrate, 500, "Bit rate in kilobits/second.");
-DEFINE_int32(keyframe_interval,
-             0,
-             "Forces a keyframe every Nth frame. "
-             "0 means the encoder decides when to insert keyframes.  Note that "
-             "the encoder may create a keyframe in other locations in addition "
-             "to the interval that is set using this parameter.");
-DEFINE_int32(temporal_layers,
-             0,
-             "The number of temporal layers to use "
-             "(VP8 specific codec setting). Must be 0-4.");
-DEFINE_int32(packet_size,
-             1500,
-             "Simulated network packet size in bytes (MTU). "
-             "Used for packet loss simulation.");
-DEFINE_int32(max_payload_size,
-             1440,
-             "Max payload size in bytes for the "
-             "encoder.");
+DEFINE_int(bitrate, 500, "Bit rate in kilobits/second.");
+DEFINE_int(keyframe_interval,
+           0,
+           "Forces a keyframe every Nth frame. "
+           "0 means the encoder decides when to insert keyframes.  Note that "
+           "the encoder may create a keyframe in other locations in addition "
+           "to the interval that is set using this parameter.");
+DEFINE_int(temporal_layers,
+           0,
+           "The number of temporal layers to use "
+           "(VP8 specific codec setting). Must be 0-4.");
+DEFINE_int(packet_size,
+           1500,
+           "Simulated network packet size in bytes (MTU). "
+           "Used for packet loss simulation.");
+DEFINE_int(max_payload_size,
+           1440,
+           "Max payload size in bytes for the "
+           "encoder.");
 DEFINE_string(packet_loss_mode,
               "uniform",
               "Packet loss mode. Two different "
               "packet loss models are supported: uniform or burst. This "
               "setting has no effect unless packet_loss_rate is >0. ");
-DEFINE_double(packet_loss_probability,
-              0.0,
-              "Packet loss probability. A value "
-              "between 0.0 and 1.0 that defines the probability of a packet "
-              "being lost. 0.1 means 10% and so on.");
-DEFINE_int32(packet_loss_burst_length,
-             1,
-             "Packet loss burst length. Defines "
-             "how many packets will be lost in a burst when a packet has been "
-             "decided to be lost. Must be >=1.");
+DEFINE_float(packet_loss_probability,
+             0.0f,
+             "Packet loss probability. A value "
+             "between 0.0 and 1.0 that defines the probability of a packet "
+             "being lost. 0.1 means 10% and so on.");
+DEFINE_int(packet_loss_burst_length,
+           1,
+           "Packet loss burst length. Defines "
+           "how many packets will be lost in a burst when a packet has been "
+           "decided to be lost. Must be >=1.");
 DEFINE_bool(csv,
             false,
             "CSV output. Enabling this will output all frame "
@@ -126,12 +127,13 @@
             "Verbose mode. Prints a lot of debugging info. "
             "Suitable for tracking progress but not for capturing output. "
             "Disable with --noverbose flag.");
+DEFINE_bool(help, false, "Prints this message.");
 
 // Custom log method that only prints if the verbose flag is given.
 // Supports all the standard printf parameters and formatting (just forwarded).
 int Log(const char* format, ...) {
   int result = 0;
-  if (FLAGS_verbose) {
+  if (FLAG_verbose) {
     va_list args;
     va_start(args, format);
     result = vprintf(format, args);
@@ -143,55 +145,58 @@
 // Validates the arguments given as command line flags and fills in the
 // TestConfig struct with all configurations needed for video processing.
 // Returns 0 if everything is OK, otherwise an exit code.
-int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
+int HandleCommandLineFlags(webrtc::test::TestConfig* config,
+                           const std::string& usage) {
   // Validate the mandatory flags:
-  if (FLAGS_input_filename.empty() || FLAGS_width == -1 || FLAGS_height == -1) {
-    printf("%s\n", google::ProgramUsage());
+  if (strlen(FLAG_input_filename) == 0 ||
+      FLAG_width == -1 || FLAG_height == -1) {
+    printf("%s\n", usage.c_str());
     return 1;
   }
-  config->name = FLAGS_test_name;
-  config->description = FLAGS_test_description;
+  config->name = FLAG_test_name;
+  config->description = FLAG_test_description;
 
   // Verify the input file exists and is readable.
   FILE* test_file;
-  test_file = fopen(FLAGS_input_filename.c_str(), "rb");
+  test_file = fopen(FLAG_input_filename, "rb");
   if (test_file == NULL) {
     fprintf(stderr, "Cannot read the specified input file: %s\n",
-            FLAGS_input_filename.c_str());
+            FLAG_input_filename);
     return 2;
   }
   fclose(test_file);
-  config->input_filename = FLAGS_input_filename;
+  config->input_filename = FLAG_input_filename;
 
   // Verify the output dir exists.
   struct stat dir_info;
-  if (!(stat(FLAGS_output_dir.c_str(), &dir_info) == 0 &&
+  if (!(stat(FLAG_output_dir, &dir_info) == 0 &&
         S_ISDIR(dir_info.st_mode))) {
     fprintf(stderr, "Cannot find output directory: %s\n",
-            FLAGS_output_dir.c_str());
+            FLAG_output_dir);
     return 3;
   }
-  config->output_dir = FLAGS_output_dir;
+  config->output_dir = FLAG_output_dir;
 
   // Manufacture an output filename if none was given.
-  if (FLAGS_output_filename.empty()) {
+  if (strlen(FLAG_output_filename) == 0) {
     // Cut out the filename without extension from the given input file
     // (which may include a path)
-    size_t startIndex = FLAGS_input_filename.find_last_of("/") + 1;
+    size_t startIndex = config->input_filename.find_last_of("/") + 1;
     if (startIndex == 0) {
       startIndex = 0;
     }
-    FLAGS_output_filename =
-        FLAGS_input_filename.substr(
-            startIndex, FLAGS_input_filename.find_last_of(".") - startIndex) +
+    config->output_filename =
+        config->input_filename.substr(
+            startIndex, config->input_filename.find_last_of(".") - startIndex) +
         "_out.yuv";
+  } else {
+    config->output_filename = FLAG_output_filename;
   }
 
   // Verify output file can be written.
-  if (FLAGS_output_dir == ".") {
-    config->output_filename = FLAGS_output_filename;
-  } else {
-    config->output_filename = FLAGS_output_dir + "/" + FLAGS_output_filename;
+  if (config->output_dir != ".") {
+    config->output_filename =
+        config->output_dir + "/" + config->output_filename;
   }
   test_file = fopen(config->output_filename.c_str(), "wb");
   if (test_file == NULL) {
@@ -202,99 +207,98 @@
   fclose(test_file);
 
   // Check single core flag.
-  config->use_single_core = FLAGS_use_single_core;
+  config->use_single_core = FLAG_use_single_core;
 
   // Get codec specific configuration.
   webrtc::test::CodecSettings(webrtc::kVideoCodecVP8, &config->codec_settings);
 
   // Check the temporal layers.
-  if (FLAGS_temporal_layers < 0 ||
-      FLAGS_temporal_layers > webrtc::kMaxTemporalStreams) {
+  if (FLAG_temporal_layers < 0 ||
+      FLAG_temporal_layers > webrtc::kMaxTemporalStreams) {
     fprintf(stderr, "Temporal layers number must be 0-4, was: %d\n",
-            FLAGS_temporal_layers);
+            FLAG_temporal_layers);
     return 13;
   }
-  config->codec_settings.VP8()->numberOfTemporalLayers = FLAGS_temporal_layers;
+  config->codec_settings.VP8()->numberOfTemporalLayers = FLAG_temporal_layers;
 
   // Check the bit rate.
-  if (FLAGS_bitrate <= 0) {
-    fprintf(stderr, "Bit rate must be >0 kbps, was: %d\n", FLAGS_bitrate);
+  if (FLAG_bitrate <= 0) {
+    fprintf(stderr, "Bit rate must be >0 kbps, was: %d\n", FLAG_bitrate);
     return 5;
   }
-  config->codec_settings.startBitrate = FLAGS_bitrate;
+  config->codec_settings.startBitrate = FLAG_bitrate;
 
   // Check the keyframe interval.
-  if (FLAGS_keyframe_interval < 0) {
+  if (FLAG_keyframe_interval < 0) {
     fprintf(stderr, "Keyframe interval must be >=0, was: %d\n",
-            FLAGS_keyframe_interval);
+            FLAG_keyframe_interval);
     return 6;
   }
-  config->keyframe_interval = FLAGS_keyframe_interval;
+  config->keyframe_interval = FLAG_keyframe_interval;
 
   // Check packet size and max payload size.
-  if (FLAGS_packet_size <= 0) {
+  if (FLAG_packet_size <= 0) {
     fprintf(stderr, "Packet size must be >0 bytes, was: %d\n",
-            FLAGS_packet_size);
+            FLAG_packet_size);
     return 7;
   }
   config->networking_config.packet_size_in_bytes =
-      static_cast<size_t>(FLAGS_packet_size);
+      static_cast<size_t>(FLAG_packet_size);
 
-  if (FLAGS_max_payload_size <= 0) {
+  if (FLAG_max_payload_size <= 0) {
     fprintf(stderr, "Max payload size must be >0 bytes, was: %d\n",
-            FLAGS_max_payload_size);
+            FLAG_max_payload_size);
     return 8;
   }
   config->networking_config.max_payload_size_in_bytes =
-      static_cast<size_t>(FLAGS_max_payload_size);
+      static_cast<size_t>(FLAG_max_payload_size);
 
   // Check the width and height
-  if (FLAGS_width <= 0 || FLAGS_height <= 0) {
+  if (FLAG_width <= 0 || FLAG_height <= 0) {
     fprintf(stderr, "Width and height must be >0.");
     return 9;
   }
-  config->codec_settings.width = FLAGS_width;
-  config->codec_settings.height = FLAGS_height;
-  config->codec_settings.maxFramerate = FLAGS_framerate;
+  config->codec_settings.width = FLAG_width;
+  config->codec_settings.height = FLAG_height;
+  config->codec_settings.maxFramerate = FLAG_framerate;
 
   // Calculate the size of each frame to read (according to YUV spec).
   config->frame_length_in_bytes =
       3 * config->codec_settings.width * config->codec_settings.height / 2;
 
   // Check packet loss settings
-  if (FLAGS_packet_loss_mode != "uniform" &&
-      FLAGS_packet_loss_mode != "burst") {
+  if (strcmp(FLAG_packet_loss_mode, "uniform") == 0) {
+    config->networking_config.packet_loss_mode = webrtc::test::kUniform;
+  } else if (strcmp(FLAG_packet_loss_mode, "burst") == 0) {
+    config->networking_config.packet_loss_mode = webrtc::test::kBurst;
+  } else {
     fprintf(stderr,
             "Unsupported packet loss mode, must be 'uniform' or "
             "'burst'\n.");
     return 10;
   }
-  config->networking_config.packet_loss_mode = webrtc::test::kUniform;
-  if (FLAGS_packet_loss_mode == "burst") {
-    config->networking_config.packet_loss_mode = webrtc::test::kBurst;
-  }
 
-  if (FLAGS_packet_loss_probability < 0.0 ||
-      FLAGS_packet_loss_probability > 1.0) {
+  if (FLAG_packet_loss_probability < 0.0 ||
+      FLAG_packet_loss_probability > 1.0) {
     fprintf(stderr,
             "Invalid packet loss probability. Must be 0.0 - 1.0, "
             "was: %f\n",
-            FLAGS_packet_loss_probability);
+            FLAG_packet_loss_probability);
     return 11;
   }
   config->networking_config.packet_loss_probability =
-      FLAGS_packet_loss_probability;
+      FLAG_packet_loss_probability;
 
-  if (FLAGS_packet_loss_burst_length < 1) {
+  if (FLAG_packet_loss_burst_length < 1) {
     fprintf(stderr,
             "Invalid packet loss burst length, must be >=1, "
             "was: %d\n",
-            FLAGS_packet_loss_burst_length);
+            FLAG_packet_loss_burst_length);
     return 12;
   }
   config->networking_config.packet_loss_burst_length =
-      FLAGS_packet_loss_burst_length;
-  config->verbose = FLAGS_verbose;
+      FLAG_packet_loss_burst_length;
+  config->verbose = FLAG_verbose;
   return 0;
 }
 
@@ -464,18 +468,21 @@
       "Quality test application for video comparisons.\n"
       "Run " +
       program_name +
-      " --helpshort for usage.\n"
+      " --help for usage.\n"
       "Example usage:\n" +
       program_name +
       " --input_filename=filename.yuv --width=352 --height=288\n";
-  google::SetUsageMessage(usage);
 
-  google::ParseCommandLineFlags(&argc, &argv, true);
+  rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
+  if (FLAG_help) {
+    rtc::FlagList::Print(nullptr, false);
+    return 0;
+  }
 
   // Create TestConfig.
   webrtc::test::TestConfig config;
 
-  int return_code = HandleCommandLineFlags(&config);
+  int return_code = HandleCommandLineFlags(&config, usage);
   // Exit if an invalid argument is supplied.
   if (return_code != 0) {
     return return_code;
@@ -500,7 +507,7 @@
       &packet_reader, config.networking_config, config.verbose);
   // By default the packet manipulator is seeded with a fixed random.
   // If disabled we must generate a new seed.
-  if (FLAGS_disable_fixed_random_seed) {
+  if (FLAG_disable_fixed_random_seed) {
     packet_manipulator.InitializeRandomSeed(time(NULL));
   }
   webrtc::test::VideoProcessor* processor = new webrtc::test::VideoProcessor(
@@ -539,10 +546,10 @@
   webrtc::test::QualityMetricsResult psnr_result;
   CalculatePsnrVideoMetrics(&config, &psnr_result);
 
-  if (FLAGS_csv) {
+  if (FLAG_csv) {
     PrintCsvOutput(stats, ssim_result, psnr_result);
   }
-  if (FLAGS_python) {
+  if (FLAG_python) {
     PrintPythonOutput(config, stats, ssim_result, psnr_result);
   }
   delete processor;
diff --git a/video/BUILD.gn b/video/BUILD.gn
index c855105..f3bba0e 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -223,7 +223,6 @@
       "../test:test_support",
       "../test:video_test_common",
       "../test:video_test_support",
-      "//third_party/gflags",
     ]
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
diff --git a/video/replay.cc b/video/replay.cc
index e23ff29..5a391d6 100644
--- a/video/replay.cc
+++ b/video/replay.cc
@@ -14,13 +14,14 @@
 #include <memory>
 #include <sstream>
 
-#include "gflags/gflags.h"
 #include "webrtc/api/video_codecs/video_decoder.h"
 #include "webrtc/call/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/flags.h"
+#include "webrtc/rtc_base/string_to_number.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/call_test.h"
@@ -36,121 +37,113 @@
 #include "webrtc/test/video_renderer.h"
 #include "webrtc/typedefs.h"
 
+namespace {
+
+static bool ValidatePayloadType(int32_t payload_type) {
+  return payload_type > 0 && payload_type <= 127;
+}
+
+static bool ValidateSsrc(const char* ssrc_string) {
+  return rtc::StringToNumber<uint32_t>(ssrc_string).has_value();
+}
+
+static bool ValidateOptionalPayloadType(int32_t payload_type) {
+  return payload_type == -1 || ValidatePayloadType(payload_type);
+}
+
+static bool ValidateRtpHeaderExtensionId(int32_t extension_id) {
+  return extension_id >= -1 && extension_id < 15;
+}
+
+bool ValidateInputFilenameNotEmpty(const std::string& string) {
+  return !string.empty();
+}
+
+}  // namespace
+
 namespace webrtc {
 namespace flags {
 
 // TODO(pbos): Multiple receivers.
 
 // Flag for payload type.
-static bool ValidatePayloadType(const char* flagname, int32_t payload_type) {
-  return payload_type > 0 && payload_type <= 127;
-}
-DEFINE_int32(payload_type, test::CallTest::kPayloadTypeVP8, "Payload type");
-static int PayloadType() { return static_cast<int>(FLAGS_payload_type); }
-static const bool payload_dummy =
-    google::RegisterFlagValidator(&FLAGS_payload_type, &ValidatePayloadType);
+DEFINE_int(payload_type, test::CallTest::kPayloadTypeVP8, "Payload type");
+static int PayloadType() { return static_cast<int>(FLAG_payload_type); }
 
-DEFINE_int32(payload_type_rtx,
-             test::CallTest::kSendRtxPayloadType,
-             "RTX payload type");
+DEFINE_int(payload_type_rtx,
+           test::CallTest::kSendRtxPayloadType,
+           "RTX payload type");
 static int PayloadTypeRtx() {
-  return static_cast<int>(FLAGS_payload_type_rtx);
+  return static_cast<int>(FLAG_payload_type_rtx);
 }
-static const bool payload_rtx_dummy =
-    google::RegisterFlagValidator(&FLAGS_payload_type_rtx,
-                                  &ValidatePayloadType);
 
 // Flag for SSRC.
-static bool ValidateSsrc(const char* flagname, uint64_t ssrc) {
-  return ssrc > 0 && ssrc <= 0xFFFFFFFFu;
+const std::string& DefaultSsrc() {
+  static const std::string ssrc = std::to_string(
+      test::CallTest::kVideoSendSsrcs[0]);
+  return ssrc;
+}
+DEFINE_string(ssrc, DefaultSsrc().c_str(), "Incoming SSRC");
+static uint32_t Ssrc() {
+  return rtc::StringToNumber<uint32_t>(FLAG_ssrc).value();
 }
 
-DEFINE_uint64(ssrc, test::CallTest::kVideoSendSsrcs[0], "Incoming SSRC");
-static uint32_t Ssrc() { return static_cast<uint32_t>(FLAGS_ssrc); }
-static const bool ssrc_dummy =
-    google::RegisterFlagValidator(&FLAGS_ssrc, &ValidateSsrc);
-
-DEFINE_uint64(ssrc_rtx, test::CallTest::kSendRtxSsrcs[0], "Incoming RTX SSRC");
+const std::string& DefaultSsrcRtx() {
+  static const std::string ssrc_rtx = std::to_string(
+      test::CallTest::kSendRtxSsrcs[0]);
+  return ssrc_rtx;
+}
+DEFINE_string(ssrc_rtx, DefaultSsrcRtx().c_str(), "Incoming RTX SSRC");
 static uint32_t SsrcRtx() {
-  return static_cast<uint32_t>(FLAGS_ssrc_rtx);
-}
-static const bool ssrc_rtx_dummy =
-    google::RegisterFlagValidator(&FLAGS_ssrc_rtx, &ValidateSsrc);
-
-static bool ValidateOptionalPayloadType(const char* flagname,
-                                        int32_t payload_type) {
-  return payload_type == -1 || ValidatePayloadType(flagname, payload_type);
+  return rtc::StringToNumber<uint32_t>(FLAG_ssrc_rtx).value();
 }
 
 // Flag for RED payload type.
-DEFINE_int32(red_payload_type, -1, "RED payload type");
+DEFINE_int(red_payload_type, -1, "RED payload type");
 static int RedPayloadType() {
-  return static_cast<int>(FLAGS_red_payload_type);
+  return static_cast<int>(FLAG_red_payload_type);
 }
-static const bool red_dummy =
-    google::RegisterFlagValidator(&FLAGS_red_payload_type,
-                                  &ValidateOptionalPayloadType);
 
 // Flag for ULPFEC payload type.
-DEFINE_int32(fec_payload_type, -1, "ULPFEC payload type");
+DEFINE_int(fec_payload_type, -1, "ULPFEC payload type");
 static int FecPayloadType() {
-  return static_cast<int>(FLAGS_fec_payload_type);
+  return static_cast<int>(FLAG_fec_payload_type);
 }
-static const bool fec_dummy =
-    google::RegisterFlagValidator(&FLAGS_fec_payload_type,
-                                  &ValidateOptionalPayloadType);
 
 // Flag for abs-send-time id.
-static bool ValidateRtpHeaderExtensionId(const char* flagname,
-                                         int32_t extension_id) {
-  return extension_id >= -1 || extension_id < 15;
-}
-DEFINE_int32(abs_send_time_id, -1, "RTP extension ID for abs-send-time");
-static int AbsSendTimeId() { return static_cast<int>(FLAGS_abs_send_time_id); }
-static const bool abs_send_time_dummy =
-    google::RegisterFlagValidator(&FLAGS_abs_send_time_id,
-                                  &ValidateRtpHeaderExtensionId);
+DEFINE_int(abs_send_time_id, -1, "RTP extension ID for abs-send-time");
+static int AbsSendTimeId() { return static_cast<int>(FLAG_abs_send_time_id); }
 
 // Flag for transmission-offset id.
-DEFINE_int32(transmission_offset_id,
-             -1,
-             "RTP extension ID for transmission-offset");
+DEFINE_int(transmission_offset_id,
+           -1,
+           "RTP extension ID for transmission-offset");
 static int TransmissionOffsetId() {
-  return static_cast<int>(FLAGS_transmission_offset_id);
+  return static_cast<int>(FLAG_transmission_offset_id);
 }
-static const bool timestamp_offset_dummy =
-    google::RegisterFlagValidator(&FLAGS_transmission_offset_id,
-                                  &ValidateRtpHeaderExtensionId);
 
 // Flag for rtpdump input file.
-bool ValidateInputFilenameNotEmpty(const char* flagname,
-                                   const std::string& string) {
-  return !string.empty();
-}
-
 DEFINE_string(input_file, "", "input file");
 static std::string InputFile() {
-  return static_cast<std::string>(FLAGS_input_file);
+  return static_cast<std::string>(FLAG_input_file);
 }
-static const bool input_file_dummy =
-    google::RegisterFlagValidator(&FLAGS_input_file,
-                                  &ValidateInputFilenameNotEmpty);
 
 // Flag for raw output files.
 DEFINE_string(out_base, "", "Basename (excluding .jpg) for raw output");
 static std::string OutBase() {
-  return static_cast<std::string>(FLAGS_out_base);
+  return static_cast<std::string>(FLAG_out_base);
 }
 
 DEFINE_string(decoder_bitstream_filename, "", "Decoder bitstream output file");
 static std::string DecoderBitstreamFilename() {
-  return static_cast<std::string>(FLAGS_decoder_bitstream_filename);
+  return static_cast<std::string>(FLAG_decoder_bitstream_filename);
 }
 
 // Flag for video codec.
 DEFINE_string(codec, "VP8", "Video codec");
-static std::string Codec() { return static_cast<std::string>(FLAGS_codec); }
+static std::string Codec() { return static_cast<std::string>(FLAG_codec); }
 
+DEFINE_bool(help, false, "Print this message.");
 }  // namespace flags
 
 static const uint32_t kReceiverLocalSsrc = 0x123456;
@@ -330,7 +323,24 @@
 
 int main(int argc, char* argv[]) {
   ::testing::InitGoogleTest(&argc, argv);
-  google::ParseCommandLineFlags(&argc, &argv, true);
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
+    return 1;
+  }
+  if (webrtc::flags::FLAG_help) {
+    rtc::FlagList::Print(nullptr, false);
+    return 0;
+  }
+
+  RTC_CHECK(ValidatePayloadType(webrtc::flags::FLAG_payload_type));
+  RTC_CHECK(ValidatePayloadType(webrtc::flags::FLAG_payload_type_rtx));
+  RTC_CHECK(ValidateSsrc(webrtc::flags::FLAG_ssrc));
+  RTC_CHECK(ValidateSsrc(webrtc::flags::FLAG_ssrc_rtx));
+  RTC_CHECK(ValidateOptionalPayloadType(webrtc::flags::FLAG_red_payload_type));
+  RTC_CHECK(ValidateOptionalPayloadType(webrtc::flags::FLAG_fec_payload_type));
+  RTC_CHECK(ValidateRtpHeaderExtensionId(webrtc::flags::FLAG_abs_send_time_id));
+  RTC_CHECK(ValidateRtpHeaderExtensionId(
+      webrtc::flags::FLAG_transmission_offset_id));
+  RTC_CHECK(ValidateInputFilenameNotEmpty(webrtc::flags::FLAG_input_file));
 
   webrtc::test::RunTest(webrtc::RtpReplay);
   return 0;
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index e458fee..984d149 100644
--- a/voice_engine/BUILD.gn
+++ b/voice_engine/BUILD.gn
@@ -260,7 +260,6 @@
         "../test/:video_test_common",
         "//testing/gmock",
         "//testing/gtest",
-        "//third_party/gflags",
       ]
 
       sources = [
diff --git a/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc b/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
index 9af7f86..24bb0a7 100644
--- a/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
+++ b/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
@@ -11,11 +11,14 @@
 #include <memory>
 
 #include "webrtc/rtc_base/criticalsection.h"
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
 #include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
 
+DECLARE_bool(include_timing_dependent_tests);
+
 class TestRtpObserver : public webrtc::VoERTPObserver {
  public:
   TestRtpObserver() : changed_ssrc_event_(webrtc::EventWrapper::Create()) {}
@@ -85,7 +88,7 @@
 };
 
 TEST_F(RtpRtcpTest, RemoteRtcpCnameHasPropagatedToRemoteSide) {
-  if (!FLAGS_include_timing_dependent_tests) {
+  if (!FLAG_include_timing_dependent_tests) {
     TEST_LOG("Skipping test - running in slow execution environment...\n");
     return;
   }
diff --git a/voice_engine/test/auto_test/voe_standard_test.cc b/voice_engine/test/auto_test/voe_standard_test.cc
index 5449a2f..545be71 100644
--- a/voice_engine/test/auto_test/voe_standard_test.cc
+++ b/voice_engine/test/auto_test/voe_standard_test.cc
@@ -14,6 +14,7 @@
 #include <stdio.h>
 #include <string.h>
 
+#include "webrtc/rtc_base/flags.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/voice_engine/test/auto_test/automated_mode.h"
@@ -26,6 +27,7 @@
 DEFINE_bool(automated, false,
             "If true, we'll run the automated tests we have in noninteractive "
             "mode.");
+DEFINE_bool(help, false, "Print this message.");
 
 namespace webrtc {
 namespace voetest {
@@ -101,9 +103,15 @@
   // This function and RunInAutomatedMode is defined in automated_mode.cc
   // to avoid macro clashes with googletest (for instance ASSERT_TRUE).
   webrtc::voetest::InitializeGoogleTest(&argc, 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;
+  }
 
-  if (FLAGS_automated) {
+  if (FLAG_automated) {
     return webrtc::voetest::RunInAutomatedMode();
   }
 
diff --git a/voice_engine/test/auto_test/voe_standard_test.h b/voice_engine/test/auto_test/voe_standard_test.h
index 1fd2706..9eaed1c 100644
--- a/voice_engine/test/auto_test/voe_standard_test.h
+++ b/voice_engine/test/auto_test/voe_standard_test.h
@@ -14,12 +14,9 @@
 #include <stdio.h>
 #include <string>
 
-#include "gflags/gflags.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/test/auto_test/voe_test_common.h"
 
-DECLARE_bool(include_timing_dependent_tests);
-
 namespace webrtc {
 namespace voetest {