blob: 39903631635dd16f027285d143f596473bf7ded0 [file] [log] [blame]
/*
* 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 <string.h>
#include <iostream>
#include <memory>
#include <sstream>
#include <string>
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
#include "test/rtp_file_writer.h"
namespace {
using MediaType = webrtc::ParsedRtcEventLogNew::MediaType;
DEFINE_bool(
audio,
true,
"Use --noaudio to exclude audio packets from the converted RTPdump file.");
DEFINE_bool(
video,
true,
"Use --novideo to exclude video packets from the converted RTPdump file.");
DEFINE_bool(
data,
true,
"Use --nodata to exclude data packets from the converted RTPdump file.");
DEFINE_bool(
rtp,
true,
"Use --nortp to exclude RTP packets from the converted RTPdump file.");
DEFINE_bool(
rtcp,
true,
"Use --nortcp to exclude RTCP packets from the converted RTPdump file.");
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,
// 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 +
" --help for usage.\n"
"Example usage:\n" +
program_name + " input.rel output.rtp\n";
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
FLAG_help || argc != 3) {
std::cout << usage;
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
std::string input_file = argv[1];
std::string output_file = argv[2];
uint32_t ssrc_filter = 0;
if (strlen(FLAG_ssrc) > 0)
RTC_CHECK(ParseSsrc(FLAG_ssrc, &ssrc_filter))
<< "Flag verification has failed.";
webrtc::ParsedRtcEventLogNew 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 (FLAG_rtp && parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLogNew::RTP_EVENT) {
webrtc::test::RtpPacket packet;
webrtc::PacketDirection direction;
parsed_stream.GetRtpHeader(i, &direction, packet.data, &packet.length,
&packet.original_length, nullptr);
if (packet.original_length > packet.length)
header_only = true;
packet.time_ms = parsed_stream.GetTimestamp(i) / 1000;
webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet.data,
packet.length);
// TODO(terelius): Maybe add a flag to dump outgoing traffic instead?
if (direction == webrtc::kOutgoingPacket)
continue;
webrtc::RTPHeader parsed_header;
rtp_parser.Parse(&parsed_header);
MediaType media_type =
parsed_stream.GetMediaType(parsed_header.ssrc, direction);
if (!FLAG_audio && media_type == MediaType::AUDIO)
continue;
if (!FLAG_video && media_type == MediaType::VIDEO)
continue;
if (!FLAG_data && media_type == MediaType::DATA)
continue;
if (strlen(FLAG_ssrc) > 0) {
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 (FLAG_rtcp && parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLogNew::RTCP_EVENT) {
webrtc::test::RtpPacket packet;
webrtc::PacketDirection direction;
parsed_stream.GetRtcpPacket(i, &direction, 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;
// Note that |packet_ssrc| is the sender SSRC. An RTCP message may contain
// report blocks for many streams, thus several SSRCs and they doen't
// necessarily have to be of the same media type.
const uint32_t packet_ssrc = webrtc::ByteReader<uint32_t>::ReadBigEndian(
reinterpret_cast<const uint8_t*>(packet.data + 4));
MediaType media_type = parsed_stream.GetMediaType(packet_ssrc, direction);
if (!FLAG_audio && media_type == MediaType::AUDIO)
continue;
if (!FLAG_video && media_type == MediaType::VIDEO)
continue;
if (!FLAG_data && media_type == MediaType::DATA)
continue;
if (strlen(FLAG_ssrc) > 0) {
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;
}