| 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));
|
|
|