1/* 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11#include <iostream> 12#include <sstream> 13#include <string> 14 15#include "gflags/gflags.h" 16#include "webrtc/base/checks.h" 17#include "webrtc/base/scoped_ptr.h" 18#include "webrtc/call/rtc_event_log.h" 19#include "webrtc/modules/rtp_rtcp/source/byte_io.h" 20#include "webrtc/test/rtp_file_writer.h" 21 22// Files generated at build-time by the protobuf compiler. 23#ifdef WEBRTC_ANDROID_PLATFORM_BUILD 24#include "external/webrtc/webrtc/call/rtc_event_log.pb.h" 25#else 26#include "webrtc/call/rtc_event_log.pb.h" 27#endif 28 29namespace { 30 31DEFINE_bool(noaudio, 32 false, 33 "Excludes audio packets from the converted RTPdump file."); 34DEFINE_bool(novideo, 35 false, 36 "Excludes video packets from the converted RTPdump file."); 37DEFINE_bool(nodata, 38 false, 39 "Excludes data packets from the converted RTPdump file."); 40DEFINE_bool(nortp, 41 false, 42 "Excludes RTP packets from the converted RTPdump file."); 43DEFINE_bool(nortcp, 44 false, 45 "Excludes RTCP packets from the converted RTPdump file."); 46DEFINE_string(ssrc, 47 "", 48 "Store only packets with this SSRC (decimal or hex, the latter " 49 "starting with 0x)."); 50 51// Parses the input string for a valid SSRC. If a valid SSRC is found, it is 52// written to the output variable |ssrc|, and true is returned. Otherwise, 53// false is returned. 54// The empty string must be validated as true, because it is the default value 55// of the command-line flag. In this case, no value is written to the output 56// variable. 57bool ParseSsrc(std::string str, uint32_t* ssrc) { 58 // If the input string starts with 0x or 0X it indicates a hexadecimal number. 59 auto read_mode = std::dec; 60 if (str.size() > 2 && 61 (str.substr(0, 2) == "0x" || str.substr(0, 2) == "0X")) { 62 read_mode = std::hex; 63 str = str.substr(2); 64 } 65 std::stringstream ss(str); 66 ss >> read_mode >> *ssrc; 67 return str.empty() || (!ss.fail() && ss.eof()); 68} 69 70} // namespace 71 72// This utility will convert a stored event log to the rtpdump format. 73int main(int argc, char* argv[]) { 74 std::string program_name = argv[0]; 75 std::string usage = 76 "Tool for converting an RtcEventLog file to an RTP dump file.\n" 77 "Run " + 78 program_name + 79 " --helpshort for usage.\n" 80 "Example usage:\n" + 81 program_name + " input.rel output.rtp\n"; 82 google::SetUsageMessage(usage); 83 google::ParseCommandLineFlags(&argc, &argv, true); 84 85 if (argc != 3) { 86 std::cout << google::ProgramUsage(); 87 return 0; 88 } 89 std::string input_file = argv[1]; 90 std::string output_file = argv[2]; 91 92 uint32_t ssrc_filter = 0; 93 if (!FLAGS_ssrc.empty()) 94 RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc_filter)) 95 << "Flag verification has failed."; 96 97 webrtc::rtclog::EventStream event_stream; 98 if (!webrtc::RtcEventLog::ParseRtcEventLog(input_file, &event_stream)) { 99 std::cerr << "Error while parsing input file: " << input_file << std::endl; 100 return -1; 101 } 102 103 rtc::scoped_ptr<webrtc::test::RtpFileWriter> rtp_writer( 104 webrtc::test::RtpFileWriter::Create( 105 webrtc::test::RtpFileWriter::FileFormat::kRtpDump, output_file)); 106 107 if (!rtp_writer.get()) { 108 std::cerr << "Error while opening output file: " << output_file 109 << std::endl; 110 return -1; 111 } 112 113 std::cout << "Found " << event_stream.stream_size() 114 << " events in the input file." << std::endl; 115 int rtp_counter = 0, rtcp_counter = 0; 116 bool header_only = false; 117 // TODO(ivoc): This can be refactored once the packet interpretation 118 // functions are finished. 119 for (int i = 0; i < event_stream.stream_size(); i++) { 120 const webrtc::rtclog::Event& event = event_stream.stream(i); 121 if (!FLAGS_nortp && event.has_type() && event.type() == event.RTP_EVENT) { 122 if (event.has_timestamp_us() && event.has_rtp_packet() && 123 event.rtp_packet().has_header() && 124 event.rtp_packet().header().size() >= 12 && 125 event.rtp_packet().has_packet_length() && 126 event.rtp_packet().has_type()) { 127 const webrtc::rtclog::RtpPacket& rtp_packet = event.rtp_packet(); 128 if (FLAGS_noaudio && rtp_packet.type() == webrtc::rtclog::AUDIO) 129 continue; 130 if (FLAGS_novideo && rtp_packet.type() == webrtc::rtclog::VIDEO) 131 continue; 132 if (FLAGS_nodata && rtp_packet.type() == webrtc::rtclog::DATA) 133 continue; 134 if (!FLAGS_ssrc.empty()) { 135 const uint32_t packet_ssrc = 136 webrtc::ByteReader<uint32_t>::ReadBigEndian( 137 reinterpret_cast<const uint8_t*>(rtp_packet.header().data() + 138 8)); 139 if (packet_ssrc != ssrc_filter) 140 continue; 141 } 142 143 webrtc::test::RtpPacket packet; 144 packet.length = rtp_packet.header().size(); 145 if (packet.length > packet.kMaxPacketBufferSize) { 146 std::cout << "Skipping packet with size " << packet.length 147 << ", the maximum supported size is " 148 << packet.kMaxPacketBufferSize << std::endl; 149 continue; 150 } 151 packet.original_length = rtp_packet.packet_length(); 152 if (packet.original_length > packet.length) 153 header_only = true; 154 packet.time_ms = event.timestamp_us() / 1000; 155 memcpy(packet.data, rtp_packet.header().data(), packet.length); 156 rtp_writer->WritePacket(&packet); 157 rtp_counter++; 158 } else { 159 std::cout << "Skipping malformed event." << std::endl; 160 } 161 } 162 if (!FLAGS_nortcp && event.has_type() && event.type() == event.RTCP_EVENT) { 163 if (event.has_timestamp_us() && event.has_rtcp_packet() && 164 event.rtcp_packet().has_type() && 165 event.rtcp_packet().has_packet_data() && 166 event.rtcp_packet().packet_data().size() > 0) { 167 const webrtc::rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet(); 168 if (FLAGS_noaudio && rtcp_packet.type() == webrtc::rtclog::AUDIO) 169 continue; 170 if (FLAGS_novideo && rtcp_packet.type() == webrtc::rtclog::VIDEO) 171 continue; 172 if (FLAGS_nodata && rtcp_packet.type() == webrtc::rtclog::DATA) 173 continue; 174 if (!FLAGS_ssrc.empty()) { 175 const uint32_t packet_ssrc = 176 webrtc::ByteReader<uint32_t>::ReadBigEndian( 177 reinterpret_cast<const uint8_t*>( 178 rtcp_packet.packet_data().data() + 4)); 179 if (packet_ssrc != ssrc_filter) 180 continue; 181 } 182 183 webrtc::test::RtpPacket packet; 184 packet.length = rtcp_packet.packet_data().size(); 185 if (packet.length > packet.kMaxPacketBufferSize) { 186 std::cout << "Skipping packet with size " << packet.length 187 << ", the maximum supported size is " 188 << packet.kMaxPacketBufferSize << std::endl; 189 continue; 190 } 191 // For RTCP packets the original_length should be set to 0 in the 192 // RTPdump format. 193 packet.original_length = 0; 194 packet.time_ms = event.timestamp_us() / 1000; 195 memcpy(packet.data, rtcp_packet.packet_data().data(), packet.length); 196 rtp_writer->WritePacket(&packet); 197 rtcp_counter++; 198 } else { 199 std::cout << "Skipping malformed event." << std::endl; 200 } 201 } 202 } 203 std::cout << "Wrote " << rtp_counter << (header_only ? " header-only" : "") 204 << " RTP packets and " << rtcp_counter << " RTCP packets to the " 205 << "output file." << std::endl; 206 return 0; 207} 208