rhubarb-lip-sync/rhubarb/lib/webrtc-8d2248ff/webrtc/call/rtc_event_log2rtp_dump.cc

187 lines
6.6 KiB
C++

/*
* Copyright (c) 2015 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 <memory>
#include <sstream>
#include <string>
#include "gflags/gflags.h"
#include "webrtc/base/checks.h"
#include "webrtc/call.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/call/rtc_event_log_parser.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/test/rtp_file_writer.h"
namespace {
DEFINE_bool(noaudio,
false,
"Excludes audio packets from the converted RTPdump file.");
DEFINE_bool(novideo,
false,
"Excludes video packets from the converted RTPdump file.");
DEFINE_bool(nodata,
false,
"Excludes data packets from the converted RTPdump file.");
DEFINE_bool(nortp,
false,
"Excludes RTP packets from the converted RTPdump file.");
DEFINE_bool(nortcp,
false,
"Excludes RTCP packets from the converted RTPdump file.");
DEFINE_string(ssrc,
"",
"Store only packets with this SSRC (decimal or hex, the latter "
"starting with 0x).");
// 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,
// 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, uint32_t* ssrc) {
// 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 >> *ssrc;
return str.empty() || (!ss.fail() && ss.eof());
}
} // namespace
// 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 =
"Tool for converting an RtcEventLog file to an RTP dump file.\n"
"Run " +
program_name +
" --helpshort 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))
<< "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;
}
std::unique_ptr<webrtc::test::RtpFileWriter> rtp_writer(
webrtc::test::RtpFileWriter::Create(
webrtc::test::RtpFileWriter::FileFormat::kRtpDump, output_file));
if (!rtp_writer.get()) {
std::cerr << "Error while opening output file: " << output_file
<< std::endl;
return -1;
}
std::cout << "Found " << parsed_stream.GetNumberOfEvents()
<< " events in the input file." << std::endl;
int rtp_counter = 0, rtcp_counter = 0;
bool header_only = false;
for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
// The parsed_stream will assert if the protobuf event is missing
// 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 &&
parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
webrtc::test::RtpPacket packet;
webrtc::PacketDirection direction;
webrtc::MediaType media_type;
parsed_stream.GetRtpHeader(i, &direction, &media_type, packet.data,
&packet.length, &packet.original_length);
if (packet.original_length > packet.length)
header_only = true;
packet.time_ms = parsed_stream.GetTimestamp(i) / 1000;
// TODO(terelius): Maybe add a flag to dump outgoing traffic instead?
if (direction == webrtc::kOutgoingPacket)
continue;
if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO)
continue;
if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
continue;
if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
continue;
if (!FLAGS_ssrc.empty()) {
const uint32_t packet_ssrc =
webrtc::ByteReader<uint32_t>::ReadBigEndian(
reinterpret_cast<const uint8_t*>(packet.data + 8));
if (packet_ssrc != ssrc_filter)
continue;
}
rtp_writer->WritePacket(&packet);
rtp_counter++;
}
if (!FLAGS_nortcp &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::RTCP_EVENT) {
webrtc::test::RtpPacket packet;
webrtc::PacketDirection direction;
webrtc::MediaType media_type;
parsed_stream.GetRtcpPacket(i, &direction, &media_type, packet.data,
&packet.length);
// For RTCP packets the original_length should be set to 0 in the
// RTPdump format.
packet.original_length = 0;
packet.time_ms = parsed_stream.GetTimestamp(i) / 1000;
// TODO(terelius): Maybe add a flag to dump outgoing traffic instead?
if (direction == webrtc::kOutgoingPacket)
continue;
if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO)
continue;
if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
continue;
if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
continue;
if (!FLAGS_ssrc.empty()) {
const uint32_t packet_ssrc =
webrtc::ByteReader<uint32_t>::ReadBigEndian(
reinterpret_cast<const uint8_t*>(packet.data + 4));
if (packet_ssrc != ssrc_filter)
continue;
}
rtp_writer->WritePacket(&packet);
rtcp_counter++;
}
}
std::cout << "Wrote " << rtp_counter << (header_only ? " header-only" : "")
<< " RTP packets and " << rtcp_counter << " RTCP packets to the "
<< "output file." << std::endl;
return 0;
}