diff --git a/webrtc/tools/BUILD.gn b/webrtc/tools/BUILD.gn
index 2850552f1a..df9a85945c 100644
--- a/webrtc/tools/BUILD.gn
+++ b/webrtc/tools/BUILD.gn
@@ -244,8 +244,8 @@ if (rtc_include_tests) {
       defines = [ "ENABLE_RTC_EVENT_LOG" ]
       deps = [
         ":event_log_visualizer_utils",
+        "../base:rtc_base_approved",
         "../test:field_trial",
-        "//third_party/gflags",
       ]
     }
   }
@@ -262,13 +262,13 @@ if (rtc_include_tests) {
     }
 
     deps = [
+      "../base:rtc_base_approved",
       "../modules:module_api",
       "../modules/audio_processing",
       "../system_wrappers:metrics_default",
       "../test:test_support",
       "//build/win:default_exe_manifest",
       "//testing/gtest",
-      "//third_party/gflags",
     ]
   }
 
diff --git a/webrtc/tools/agc/activity_metric.cc b/webrtc/tools/agc/activity_metric.cc
index a98730a342..599777c3be 100644
--- a/webrtc/tools/agc/activity_metric.cc
+++ b/webrtc/tools/agc/activity_metric.cc
@@ -16,7 +16,7 @@
 #include <algorithm>
 #include <memory>
 
-#include "gflags/gflags.h"
+#include "webrtc/base/flags.h"
 #include "webrtc/modules/audio_processing/agc/agc.h"
 #include "webrtc/modules/audio_processing/agc/loudness_histogram.h"
 #include "webrtc/modules/audio_processing/agc/utility.h"
@@ -28,7 +28,7 @@
 #include "webrtc/test/gtest.h"
 
 static const int kAgcAnalWindowSamples = 100;
-static const double kDefaultActivityThreshold = 0.3;
+static const float kDefaultActivityThreshold = 0.3f;
 
 DEFINE_bool(standalone_vad, true, "enable stand-alone VAD");
 DEFINE_string(true_vad, "", "name of a file containing true VAD in 'int'"
@@ -44,8 +44,9 @@ DEFINE_string(result, "", "name of a file to write the results. The results"
               " will be appended to the end of the file. This is optional.");
 DEFINE_string(audio_content, "", "name of a file where audio content is written"
               " to, in double format.");
-DEFINE_double(activity_threshold, kDefaultActivityThreshold,
+DEFINE_float(activity_threshold, kDefaultActivityThreshold,
               "Activity threshold");
+DEFINE_bool(help, false, "prints this message");
 
 namespace webrtc {
 
@@ -105,13 +106,13 @@ class AgcStat {
     AudioFeatures features;
     audio_processing_->ExtractFeatures(
         frame.data_, frame.samples_per_channel_, &features);
-    if (FLAGS_standalone_vad) {
+    if (FLAG_standalone_vad) {
       standalone_vad_->AddAudio(frame.data_,
                                 frame.samples_per_channel_);
     }
     if (features.num_frames > 0) {
       double p[kMaxNumFrames] = {0.5, 0.5, 0.5, 0.5};
-      if (FLAGS_standalone_vad) {
+      if (FLAG_standalone_vad) {
         standalone_vad_->GetActivity(p, kMaxNumFrames);
       }
       // TODO(turajs) combining and limiting are used in the source files as
@@ -175,20 +176,20 @@ void void_main(int argc, char* argv[]) {
   }
 
   FILE* true_vad_fid = NULL;
-  ASSERT_GT(FLAGS_true_vad.size(), 0u) << "Specify the file containing true "
+  ASSERT_GT(strlen(FLAG_true_vad), 0u) << "Specify the file containing true "
       "VADs using --true_vad flag.";
-  true_vad_fid = fopen(FLAGS_true_vad.c_str(), "rb");
+  true_vad_fid = fopen(FLAG_true_vad, "rb");
   ASSERT_TRUE(true_vad_fid != NULL) << "Cannot open the active list " <<
-      FLAGS_true_vad;
+      FLAG_true_vad;
 
   FILE* results_fid = NULL;
-  if (FLAGS_result.size() > 0) {
+  if (strlen(FLAG_result) > 0) {
     // True if this is the first time writing to this function and we add a
     // header to the beginning of the file.
     bool write_header;
     // Open in the read mode. If it fails, the file doesn't exist and has to
     // write a header for it. Otherwise no need to write a header.
-    results_fid = fopen(FLAGS_result.c_str(), "r");
+    results_fid = fopen(FLAG_result, "r");
     if (results_fid == NULL) {
       write_header = true;
     } else {
@@ -196,9 +197,9 @@ void void_main(int argc, char* argv[]) {
       write_header = false;
     }
     // Open in append mode.
-    results_fid = fopen(FLAGS_result.c_str(), "a");
+    results_fid = fopen(FLAG_result, "a");
     ASSERT_TRUE(results_fid != NULL) << "Cannot open the file, " <<
-              FLAGS_result << ", to write the results.";
+              FLAG_result << ", to write the results.";
     // Write the header if required.
     if (write_header) {
       fprintf(results_fid, "%% Total Active,  Misdetection,  "
@@ -208,19 +209,19 @@ void void_main(int argc, char* argv[]) {
   }
 
   FILE* video_vad_fid = NULL;
-  if (FLAGS_video_vad.size() > 0) {
-    video_vad_fid = fopen(FLAGS_video_vad.c_str(), "rb");
+  if (strlen(FLAG_video_vad) > 0) {
+    video_vad_fid = fopen(FLAG_video_vad, "rb");
     ASSERT_TRUE(video_vad_fid != NULL) <<  "Cannot open the file, " <<
-              FLAGS_video_vad << " to read video-based VAD decisions.\n";
+              FLAG_video_vad << " to read video-based VAD decisions.\n";
   }
 
   // AgsStat will be the owner of this file and will close it at its
   // destructor.
   FILE* audio_content_fid = NULL;
-  if (FLAGS_audio_content.size() > 0) {
-    audio_content_fid = fopen(FLAGS_audio_content.c_str(), "wb");
+  if (strlen(FLAG_audio_content) > 0) {
+    audio_content_fid = fopen(FLAG_audio_content, "wb");
     ASSERT_TRUE(audio_content_fid != NULL) << "Cannot open file, " <<
-              FLAGS_audio_content << " to write audio-content.\n";
+              FLAG_audio_content << " to write audio-content.\n";
     agc_stat.set_audio_content_file(audio_content_fid);
   }
 
@@ -231,7 +232,7 @@ void void_main(int argc, char* argv[]) {
   const size_t kSamplesToRead = frame.num_channels_ *
       frame.samples_per_channel_;
 
-  agc_stat.SetActivityThreshold(FLAGS_activity_threshold);
+  agc_stat.SetActivityThreshold(FLAG_activity_threshold);
 
   int ret_val = 0;
   int num_frames = 0;
@@ -369,17 +370,25 @@ void void_main(int argc, char* argv[]) {
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
-  char kUsage[] =
+  if (argc == 1) {
+    // Print usage information.
+    std::cout <<
       "\nCompute the number of misdetected and false-positive frames. Not\n"
       " that for each frame of audio (10 ms) there should be one true\n"
       " activity. If any video-based activity is given, there should also be\n"
       " one probability per frame.\n"
+      "Run with --help for more details on available flags.\n"
       "\nUsage:\n\n"
       "activity_metric input_pcm [options]\n"
       "where 'input_pcm' is the input audio sampled at 16 kHz in 16 bits "
       "format.\n\n";
-  google::SetUsageMessage(kUsage);
-  google::ParseCommandLineFlags(&argc, &argv, true);
+    return 0;
+  }
+  rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
+  if (FLAG_help) {
+    rtc::FlagList::Print(nullptr, false);
+    return 0;
+  }
   webrtc::void_main(argc, argv);
   return 0;
 }
diff --git a/webrtc/tools/event_log_visualizer/main.cc b/webrtc/tools/event_log_visualizer/main.cc
index 2f5ecd63e4..b9edd99fa3 100644
--- a/webrtc/tools/event_log_visualizer/main.cc
+++ b/webrtc/tools/event_log_visualizer/main.cc
@@ -10,7 +10,7 @@
 
 #include <iostream>
 
-#include "gflags/gflags.h"
+#include "webrtc/base/flags.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 #include "webrtc/test/field_trial.h"
 #include "webrtc/tools/event_log_visualizer/analyzer.h"
@@ -84,6 +84,7 @@ DEFINE_string(
     "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enabled/"
     " will assign the group Enabled to field trial WebRTC-FooFeature. Multiple "
     "trials are separated by \"/\"");
+DEFINE_bool(help, false, "prints this message");
 
 int main(int argc, char* argv[]) {
   std::string program_name = argv[0];
@@ -92,16 +93,19 @@ int main(int argc, char* argv[]) {
       "Example usage:\n" +
       program_name + " <logfile> | python\n" + "Run " + program_name +
       " --help for a list of command line options\n";
-  google::SetUsageMessage(usage);
-  google::ParseCommandLineFlags(&argc, &argv, true);
-
-  if (argc != 2) {
-    // Print usage information.
-    std::cout << google::ProgramUsage();
+  rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
+  if (FLAG_help) {
+    rtc::FlagList::Print(nullptr, false);
     return 0;
   }
 
-  webrtc::test::InitFieldTrialsFromString(FLAGS_force_fieldtrials);
+  if (argc != 2) {
+    // Print usage information.
+    std::cout << usage;
+    return 0;
+  }
+
+  webrtc::test::InitFieldTrialsFromString(FLAG_force_fieldtrials);
 
   std::string filename = argv[1];
 
@@ -118,15 +122,15 @@ int main(int argc, char* argv[]) {
   std::unique_ptr<webrtc::plotting::PlotCollection> collection(
       new webrtc::plotting::PythonPlotCollection());
 
-  if (FLAGS_plot_all || FLAGS_plot_packets) {
-    if (FLAGS_incoming) {
+  if (FLAG_plot_all || FLAG_plot_packets) {
+    if (FLAG_incoming) {
       analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket,
                                  collection->AppendNewPlot());
       analyzer.CreateAccumulatedPacketsGraph(
           webrtc::PacketDirection::kIncomingPacket,
           collection->AppendNewPlot());
     }
-    if (FLAGS_outgoing) {
+    if (FLAG_outgoing) {
       analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket,
                                  collection->AppendNewPlot());
       analyzer.CreateAccumulatedPacketsGraph(
@@ -135,95 +139,95 @@ int main(int argc, char* argv[]) {
     }
   }
 
-  if (FLAGS_plot_all || FLAGS_plot_audio_playout) {
+  if (FLAG_plot_all || FLAG_plot_audio_playout) {
     analyzer.CreatePlayoutGraph(collection->AppendNewPlot());
   }
 
-  if (FLAGS_plot_all || FLAGS_plot_audio_level) {
+  if (FLAG_plot_all || FLAG_plot_audio_level) {
     analyzer.CreateAudioLevelGraph(collection->AppendNewPlot());
   }
 
-  if (FLAGS_plot_all || FLAGS_plot_sequence_number) {
-    if (FLAGS_incoming) {
+  if (FLAG_plot_all || FLAG_plot_sequence_number) {
+    if (FLAG_incoming) {
       analyzer.CreateSequenceNumberGraph(collection->AppendNewPlot());
     }
   }
 
-  if (FLAGS_plot_all || FLAGS_plot_delay_change) {
-    if (FLAGS_incoming) {
+  if (FLAG_plot_all || FLAG_plot_delay_change) {
+    if (FLAG_incoming) {
       analyzer.CreateDelayChangeGraph(collection->AppendNewPlot());
     }
   }
 
-  if (FLAGS_plot_all || FLAGS_plot_accumulated_delay_change) {
-    if (FLAGS_incoming) {
+  if (FLAG_plot_all || FLAG_plot_accumulated_delay_change) {
+    if (FLAG_incoming) {
       analyzer.CreateAccumulatedDelayChangeGraph(collection->AppendNewPlot());
     }
   }
 
-  if (FLAGS_plot_all || FLAGS_plot_fraction_loss) {
+  if (FLAG_plot_all || FLAG_plot_fraction_loss) {
     analyzer.CreateFractionLossGraph(collection->AppendNewPlot());
     analyzer.CreateIncomingPacketLossGraph(collection->AppendNewPlot());
   }
 
-  if (FLAGS_plot_all || FLAGS_plot_total_bitrate) {
-    if (FLAGS_incoming) {
+  if (FLAG_plot_all || FLAG_plot_total_bitrate) {
+    if (FLAG_incoming) {
       analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
                                        collection->AppendNewPlot());
     }
-    if (FLAGS_outgoing) {
+    if (FLAG_outgoing) {
       analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
                                        collection->AppendNewPlot());
     }
   }
 
-  if (FLAGS_plot_all || FLAGS_plot_stream_bitrate) {
-    if (FLAGS_incoming) {
+  if (FLAG_plot_all || FLAG_plot_stream_bitrate) {
+    if (FLAG_incoming) {
       analyzer.CreateStreamBitrateGraph(
           webrtc::PacketDirection::kIncomingPacket,
           collection->AppendNewPlot());
     }
-    if (FLAGS_outgoing) {
+    if (FLAG_outgoing) {
       analyzer.CreateStreamBitrateGraph(
           webrtc::PacketDirection::kOutgoingPacket,
           collection->AppendNewPlot());
     }
   }
 
-  if (FLAGS_plot_all || FLAGS_plot_bwe) {
+  if (FLAG_plot_all || FLAG_plot_bwe) {
     analyzer.CreateBweSimulationGraph(collection->AppendNewPlot());
   }
 
-  if (FLAGS_plot_all || FLAGS_plot_network_delay_feedback) {
+  if (FLAG_plot_all || FLAG_plot_network_delay_feedback) {
     analyzer.CreateNetworkDelayFeedbackGraph(collection->AppendNewPlot());
   }
 
-  if (FLAGS_plot_all || FLAGS_plot_timestamps) {
+  if (FLAG_plot_all || FLAG_plot_timestamps) {
     analyzer.CreateTimestampGraph(collection->AppendNewPlot());
   }
 
-  if (FLAGS_plot_all || FLAGS_audio_encoder_bitrate_bps) {
+  if (FLAG_plot_all || FLAG_audio_encoder_bitrate_bps) {
     analyzer.CreateAudioEncoderTargetBitrateGraph(collection->AppendNewPlot());
   }
 
-  if (FLAGS_plot_all || FLAGS_audio_encoder_frame_length_ms) {
+  if (FLAG_plot_all || FLAG_audio_encoder_frame_length_ms) {
     analyzer.CreateAudioEncoderFrameLengthGraph(collection->AppendNewPlot());
   }
 
-  if (FLAGS_plot_all || FLAGS_audio_encoder_uplink_packet_loss_fraction) {
+  if (FLAG_plot_all || FLAG_audio_encoder_uplink_packet_loss_fraction) {
     analyzer.CreateAudioEncoderUplinkPacketLossFractionGraph(
         collection->AppendNewPlot());
   }
 
-  if (FLAGS_plot_all || FLAGS_audio_encoder_fec) {
+  if (FLAG_plot_all || FLAG_audio_encoder_fec) {
     analyzer.CreateAudioEncoderEnableFecGraph(collection->AppendNewPlot());
   }
 
-  if (FLAGS_plot_all || FLAGS_audio_encoder_dtx) {
+  if (FLAG_plot_all || FLAG_audio_encoder_dtx) {
     analyzer.CreateAudioEncoderEnableDtxGraph(collection->AppendNewPlot());
   }
 
-  if (FLAGS_plot_all || FLAGS_audio_encoder_num_channels) {
+  if (FLAG_plot_all || FLAG_audio_encoder_num_channels) {
     analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot());
   }