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 2336caa284435afcc91df10bdddd952f8f8b9d9d..7b00c080cd7c0845b2d12e05cd5291011ed5d9e3 100644 |
--- a/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc |
+++ b/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc |
@@ -13,8 +13,8 @@ |
#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/logging/rtc_event_log/rtc_event_log.h" |
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h" |
@@ -42,6 +42,7 @@ 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, |
@@ -67,26 +68,28 @@ bool ParseSsrc(std::string str, uint32_t* ssrc) { |
// This utility will convert a stored event log to the rtpdump format. |
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 != 3) { |
+ std::cout << |
"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; |
} |
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 (FLAG_ssrc) |
+ RTC_CHECK(ParseSsrc(FLAG_ssrc, &ssrc_filter)) |
<< "Flag verification has failed."; |
webrtc::ParsedRtcEventLog parsed_stream; |
@@ -114,7 +117,7 @@ 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; |
@@ -128,13 +131,13 @@ int main(int argc, char* argv[]) { |
// TODO(terelius): Maybe add a flag to dump outgoing traffic instead? |
if (direction == webrtc::kOutgoingPacket) |
continue; |
- if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO) |
+ if (FLAG_noaudio && media_type == webrtc::MediaType::AUDIO) |
continue; |
- if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO) |
+ if (FLAG_novideo && media_type == webrtc::MediaType::VIDEO) |
continue; |
- if (FLAGS_nodata && media_type == webrtc::MediaType::DATA) |
+ if (FLAG_nodata && media_type == webrtc::MediaType::DATA) |
continue; |
- if (!FLAGS_ssrc.empty()) { |
+ if (FLAG_ssrc) { |
const uint32_t packet_ssrc = |
webrtc::ByteReader<uint32_t>::ReadBigEndian( |
reinterpret_cast<const uint8_t*>(packet.data + 8)); |
@@ -145,7 +148,7 @@ 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; |
@@ -161,13 +164,13 @@ int main(int argc, char* argv[]) { |
// TODO(terelius): Maybe add a flag to dump outgoing traffic instead? |
if (direction == webrtc::kOutgoingPacket) |
continue; |
- if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO) |
+ if (FLAG_noaudio && media_type == webrtc::MediaType::AUDIO) |
continue; |
- if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO) |
+ if (FLAG_novideo && media_type == webrtc::MediaType::VIDEO) |
continue; |
- if (FLAGS_nodata && media_type == webrtc::MediaType::DATA) |
+ if (FLAG_nodata && media_type == webrtc::MediaType::DATA) |
continue; |
- if (!FLAGS_ssrc.empty()) { |
+ if (FLAG_ssrc) { |
const uint32_t packet_ssrc = |
webrtc::ByteReader<uint32_t>::ReadBigEndian( |
reinterpret_cast<const uint8_t*>(packet.data + 4)); |