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| 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/modules/rtp_rtcp/source/byte_io.h" | |
| 19 #include "webrtc/test/rtp_file_writer.h" | |
| 20 #include "webrtc/video/rtc_event_log.h" | |
| 21 | |
| 22 // Files generated at build-time by the protobuf compiler. | |
| 23 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | |
| 24 #include "external/webrtc/webrtc/video/rtc_event_log.pb.h" | |
| 25 #else | |
| 26 #include "webrtc/video/rtc_event_log.pb.h" | |
| 27 #endif | |
| 28 | |
| 29 namespace { | |
| 30 | |
| 31 DEFINE_bool(noaudio, | |
| 32 false, | |
| 33 "Excludes audio packets from the converted RTPdump file."); | |
| 34 DEFINE_bool(novideo, | |
| 35 false, | |
| 36 "Excludes video packets from the converted RTPdump file."); | |
| 37 DEFINE_bool(nodata, | |
| 38 false, | |
| 39 "Excludes data packets from the converted RTPdump file."); | |
| 40 DEFINE_bool(nortp, | |
| 41 false, | |
| 42 "Excludes RTP packets from the converted RTPdump file."); | |
| 43 DEFINE_bool(nortcp, | |
| 44 false, | |
| 45 "Excludes RTCP packets from the converted RTPdump file."); | |
| 46 DEFINE_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. | |
| 57 bool 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. | |
| 73 int 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 } | |
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Chromium Code Reviews
Issue 1345983009:
Revert of Tool to convert RtcEventLog files to RtpDump format. (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master