Replace remaining gflags usages with rtc_base/flags

webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc

Index: webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc b/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
index 23d69416e72b63718946976cd0fc3ed1e2bfb745..4275e5933ffe5e17af6b7705e0e84da6c13791d3 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
+++ b/webrtc/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 @@ DEFINE_string(ssrc,
               "",
               "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 @@ int main(int argc, char* argv[]) {
       "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 @@ int main(int argc, char* argv[]) {
     // 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 @@ int main(int argc, char* argv[]) {
       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 @@ int main(int argc, char* argv[]) {
       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 @@ int main(int argc, char* argv[]) {
       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;
       }