| 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;
|
|
|