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 c7214e2bc8572226ec3477e63e9fb44be4402e82..3f0310853f667957b41ce37b46e70628676b7b0c 100644 |
--- a/webrtc/logging/rtc_event_log/rtc_event_log2text.cc |
+++ b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc |
@@ -8,13 +8,14 @@ |
* be found in the AUTHORS file in the root of the source tree. |
*/ |
+#include <string.h> |
+ |
#include <iostream> |
#include <map> |
#include <sstream> |
#include <string> |
#include <utility> // pair |
-#include "gflags/gflags.h" |
#include "webrtc/common_types.h" |
#include "webrtc/config.h" |
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h" |
@@ -35,6 +36,7 @@ |
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" |
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
#include "webrtc/rtc_base/checks.h" |
+#include "webrtc/rtc_base/flags.h" |
namespace { |
@@ -54,6 +56,7 @@ DEFINE_string(ssrc, |
"", |
"Print only packets with this SSRC (decimal or hex, the latter " |
"starting with 0x)."); |
+DEFINE_bool(help, false, "Prints this message."); |
using MediaType = webrtc::ParsedRtcEventLog::MediaType; |
@@ -81,17 +84,17 @@ bool ParseSsrc(std::string str) { |
bool ExcludePacket(webrtc::PacketDirection direction, |
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 == MediaType::AUDIO) |
+ if (FLAG_noaudio && media_type == MediaType::AUDIO) |
return true; |
- if (FLAGS_novideo && media_type == MediaType::VIDEO) |
+ if (FLAG_novideo && media_type == MediaType::VIDEO) |
return true; |
- if (FLAGS_nodata && media_type == MediaType::DATA) |
+ if (FLAG_nodata && media_type == MediaType::DATA) |
return true; |
- if (!FLAGS_ssrc.empty() && packet_ssrc != filtered_ssrc) |
+ if (strlen(FLAG_ssrc) > 0 && packet_ssrc != filtered_ssrc) |
return true; |
return false; |
} |
@@ -357,20 +360,22 @@ int main(int argc, char* argv[]) { |
"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; |
+ if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || |
+ FLAG_help || argc != 2) { |
+ std::cout << usage; |
+ if (FLAG_help) { |
+ rtc::FlagList::Print(nullptr, false); |
+ return 0; |
+ } |
+ return 1; |
} |
std::string input_file = argv[1]; |
- if (!FLAGS_ssrc.empty()) |
- RTC_CHECK(ParseSsrc(FLAGS_ssrc)) << "Flag verification has failed."; |
+ if (strlen(FLAG_ssrc) > 0) |
+ RTC_CHECK(ParseSsrc(FLAG_ssrc)) << "Flag verification has failed."; |
webrtc::RtpHeaderExtensionMap default_map = GetDefaultHeaderExtensionMap(); |
@@ -381,7 +386,7 @@ 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::rtclog::StreamConfig config = |
@@ -402,7 +407,7 @@ int main(int argc, char* argv[]) { |
} |
std::cout << "}" << 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) { |
std::vector<webrtc::rtclog::StreamConfig> configs = |
@@ -425,7 +430,7 @@ int main(int argc, char* argv[]) { |
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::rtclog::StreamConfig config = |
@@ -446,7 +451,7 @@ int main(int argc, char* argv[]) { |
} |
std::cout << "}" << 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::rtclog::StreamConfig config = parsed_stream.GetAudioSendConfig(i); |
@@ -465,7 +470,7 @@ int main(int argc, char* argv[]) { |
} |
std::cout << "}" << std::endl; |
} |
- if (!FLAGS_nortp && |
+ if (!FLAG_nortp && |
parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) { |
size_t header_length; |
size_t total_length; |
@@ -516,7 +521,7 @@ int main(int argc, char* argv[]) { |
} |
std::cout << std::endl; |
} |
- if (!FLAGS_nortcp && |
+ if (!FLAG_nortcp && |
parsed_stream.GetEventType(i) == |
webrtc::ParsedRtcEventLog::RTCP_EVENT) { |
size_t length; |