Remove gflags dependency for rtc_event_log2text and rtc_event_log2rtp_dump

webrtc/logging/rtc_event_log/rtc_event_log2text.cc

Index: webrtc/logging/rtc_event_log/rtc_event_log2text.cc
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log2text.cc b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
index b2d537e3b3b0cb2a67158e89038a9126385e9825..1eb5a5c86ec0e5bc2ef1287b40756d266a09b51e 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
@@ -12,8 +12,8 @@
 #include <sstream>
 #include <string>
 
-#include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/flags.h"
 #include "webrtc/call/call.h"
 #include "webrtc/common_types.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
@@ -51,6 +51,7 @@ DEFINE_string(ssrc,
               "",
               "Print only packets with this SSRC (decimal or hex, the latter "
               "starting with 0x).");
+DEFINE_bool(help, false, "prints this message");
 
 static uint32_t filtered_ssrc = 0;
 
@@ -76,17 +77,17 @@ bool ParseSsrc(std::string str) {
 bool ExcludePacket(webrtc::PacketDirection direction,
                    webrtc::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 == webrtc::MediaType::AUDIO)
+  if (FLAG_noaudio && media_type == webrtc::MediaType::AUDIO)
     return true;
-  if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
+  if (FLAG_novideo && media_type == webrtc::MediaType::VIDEO)
     return true;
-  if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
+  if (FLAG_nodata && media_type == webrtc::MediaType::DATA)
     return true;
-  if (!FLAGS_ssrc.empty() && packet_ssrc != filtered_ssrc)
+  if (FLAG_ssrc && packet_ssrc != filtered_ssrc)
     return true;
   return false;
 }
@@ -298,24 +299,26 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
 // fields and we attempt to access them. We don't handle this at the moment.
 int main(int argc, char* argv[]) {
   std::string program_name = argv[0];
-  std::string usage =
+  rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
+  if (FLAG_help) {
+    rtc::FlagList::Print(nullptr, false);
+    return 0;
+  }
+
+  if (argc != 2) {
+    std::cout <<
       "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;
   }
   std::string input_file = argv[1];
 
-  if (!FLAGS_ssrc.empty())
-    RTC_CHECK(ParseSsrc(FLAGS_ssrc)) << "Flag verification has failed.";
+  if (FLAG_ssrc)
+    RTC_CHECK(ParseSsrc(FLAG_ssrc)) << "Flag verification has failed.";
 
   webrtc::ParsedRtcEventLog parsed_stream;
   if (!parsed_stream.ParseFile(input_file)) {
@@ -324,7 +327,7 @@ int main(int argc, char* argv[]) {
   }
 
   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::VideoReceiveStream::Config config(nullptr);
@@ -333,7 +336,7 @@ int main(int argc, char* argv[]) {
                 << "\tssrc=" << config.rtp.remote_ssrc
                 << "\tfeedback_ssrc=" << config.rtp.local_ssrc << 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) {
       webrtc::VideoSendStream::Config config(nullptr);
@@ -347,7 +350,7 @@ int main(int argc, char* argv[]) {
         std::cout << ssrc << ',';
       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::AudioReceiveStream::Config config;
@@ -356,7 +359,7 @@ int main(int argc, char* argv[]) {
                 << "\tssrc=" << config.rtp.remote_ssrc
                 << "\tfeedback_ssrc=" << config.rtp.local_ssrc << 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::AudioSendStream::Config config(nullptr);
@@ -364,7 +367,7 @@ int main(int argc, char* argv[]) {
       std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
                 << "\tssrc=" << config.rtp.ssrc << std::endl;
     }
-    if (!FLAGS_nortp &&
+    if (!FLAG_nortp &&
         parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
       size_t header_length;
       size_t total_length;
@@ -387,7 +390,7 @@ int main(int argc, char* argv[]) {
                 << "\tssrc=" << parsed_header.ssrc
                 << "\ttimestamp=" << parsed_header.timestamp << std::endl;
     }
-    if (!FLAGS_nortcp &&
+    if (!FLAG_nortcp &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::RTCP_EVENT) {
       size_t length;