Remove gflags dependency for rtc_event_log2text and rtc_event_log2rtp_dump

webrtc/logging/rtc_event_log/rtc_event_log2text.cc

 /*
  *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include <iostream>
 #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"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
@@ skipping 17 matching lines @@
 DEFINE_bool(novideo, false, "Excludes video packets.");
 // TODO(terelius): Note that the media type doesn't work with outgoing packets.
 DEFINE_bool(nodata, false, "Excludes data packets.");
 DEFINE_bool(nortp, false, "Excludes RTP packets.");
 DEFINE_bool(nortcp, false, "Excludes RTCP packets.");
 // TODO(terelius): Allow a list of SSRCs.
 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;
 
 // Parses the input string for a valid SSRC. If a valid SSRC is found, it is
 // written to the static global variable |filtered_ssrc|, and true is returned.
 // Otherwise, false is returned.
 // The empty string must be validated as true, because it is the default value
 // of the command-line flag. In this case, no value is written to the output
 // variable.
 bool ParseSsrc(std::string str) {
   // If the input string starts with 0x or 0X it indicates a hexadecimal number.
   auto read_mode = std::dec;
   if (str.size() > 2 &&
       (str.substr(0, 2) == "0x" || str.substr(0, 2) == "0X")) {
     read_mode = std::hex;
     str = str.substr(2);
   }
   std::stringstream ss(str);
   ss >> read_mode >> filtered_ssrc;
   return str.empty() || (!ss.fail() && ss.eof());
 }
 
 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;
 }
 
 const char* StreamInfo(webrtc::PacketDirection direction,
                        webrtc::MediaType media_type) {
   if (direction == webrtc::kOutgoingPacket) {
     if (media_type == webrtc::MediaType::AUDIO)
       return "(out,audio)";
     else if (media_type == webrtc::MediaType::VIDEO)
@@ skipping 191 matching lines @@
   }
 }
 
 }  // namespace
 
 // This utility will print basic information about each packet to stdout.
 // Note that parser will assert if the protobuf event is missing some required
 // 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)) {
     std::cerr << "Error while parsing input file: " << input_file << std::endl;
     return -1;
   }
 
   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);
       parsed_stream.GetVideoReceiveConfig(i, &config);
       std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
                 << "\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);
       parsed_stream.GetVideoSendConfig(i, &config);
       std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
       std::cout << "\tssrcs=";
       for (const auto& ssrc : config.rtp.ssrcs)
         std::cout << ssrc << ',';
       std::cout << "\trtx_ssrcs=";
       for (const auto& ssrc : config.rtp.rtx.ssrcs)
         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;
       parsed_stream.GetAudioReceiveConfig(i, &config);
       std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
                 << "\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);
       parsed_stream.GetAudioSendConfig(i, &config);
       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;
       uint8_t header[IP_PACKET_SIZE];
       webrtc::PacketDirection direction;
       webrtc::MediaType media_type;
       parsed_stream.GetRtpHeader(i, &direction, &media_type, header,
                                  &header_length, &total_length);
 
       // Parse header to get SSRC and RTP time.
       webrtc::RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
       webrtc::RTPHeader parsed_header;
       rtp_parser.Parse(&parsed_header);
 
       if (ExcludePacket(direction, media_type, parsed_header.ssrc))
         continue;
 
       std::cout << parsed_stream.GetTimestamp(i) << "\tRTP"
                 << StreamInfo(direction, media_type)
                 << "\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;
       uint8_t packet[IP_PACKET_SIZE];
       webrtc::PacketDirection direction;
       webrtc::MediaType media_type;
       parsed_stream.GetRtcpPacket(i, &direction, &media_type, packet, &length);
 
       webrtc::rtcp::CommonHeader rtcp_block;
       const uint8_t* packet_end = packet + length;
@@ skipping 30 matching lines @@
             PrintPsFeedback(rtcp_block, log_timestamp, direction, media_type);
             break;
           default:
             break;
         }
       }
     }
   }
   return 0;
 }