| /* |
| * 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 <sstream> |
| #include <string> |
| |
| #include "gflags/gflags.h" |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/scoped_ptr.h" |
| #include "webrtc/call/rtc_event_log.h" |
| #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
| #include "webrtc/test/rtp_file_writer.h" |
| |
| // Files generated at build-time by the protobuf compiler. |
| #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| #include "external/webrtc/webrtc/call/rtc_event_log.pb.h" |
| #else |
| #include "webrtc/call/rtc_event_log.pb.h" |
| #endif |
| |
| 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::rtclog::EventStream event_stream; |
| if (!webrtc::RtcEventLog::ParseRtcEventLog(input_file, &event_stream)) { |
| std::cerr << "Error while parsing input file: " << input_file << std::endl; |
| return -1; |
| } |
| |
| rtc::scoped_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 " << event_stream.stream_size() |
| << " events in the input file." << std::endl; |
| int rtp_counter = 0, rtcp_counter = 0; |
| bool header_only = false; |
| // TODO(ivoc): This can be refactored once the packet interpretation |
| // functions are finished. |
| for (int i = 0; i < event_stream.stream_size(); i++) { |
| const webrtc::rtclog::Event& event = event_stream.stream(i); |
| if (!FLAGS_nortp && event.has_type() && event.type() == event.RTP_EVENT) { |
| if (event.has_timestamp_us() && event.has_rtp_packet() && |
| event.rtp_packet().has_header() && |
| event.rtp_packet().header().size() >= 12 && |
| event.rtp_packet().has_packet_length() && |
| event.rtp_packet().has_type()) { |
| const webrtc::rtclog::RtpPacket& rtp_packet = event.rtp_packet(); |
| if (FLAGS_noaudio && rtp_packet.type() == webrtc::rtclog::AUDIO) |
| continue; |
| if (FLAGS_novideo && rtp_packet.type() == webrtc::rtclog::VIDEO) |
| continue; |
| if (FLAGS_nodata && rtp_packet.type() == webrtc::rtclog::DATA) |
| continue; |
| if (!FLAGS_ssrc.empty()) { |
| const uint32_t packet_ssrc = |
| webrtc::ByteReader<uint32_t>::ReadBigEndian( |
| reinterpret_cast<const uint8_t*>(rtp_packet.header().data() + |
| 8)); |
| if (packet_ssrc != ssrc_filter) |
| continue; |
| } |
| |
| webrtc::test::RtpPacket packet; |
| packet.length = rtp_packet.header().size(); |
| if (packet.length > packet.kMaxPacketBufferSize) { |
| std::cout << "Skipping packet with size " << packet.length |
| << ", the maximum supported size is " |
| << packet.kMaxPacketBufferSize << std::endl; |
| continue; |
| } |
| packet.original_length = rtp_packet.packet_length(); |
| if (packet.original_length > packet.length) |
| header_only = true; |
| packet.time_ms = event.timestamp_us() / 1000; |
| memcpy(packet.data, rtp_packet.header().data(), packet.length); |
| rtp_writer->WritePacket(&packet); |
| rtp_counter++; |
| } else { |
| std::cout << "Skipping malformed event." << std::endl; |
| } |
| } |
| if (!FLAGS_nortcp && event.has_type() && event.type() == event.RTCP_EVENT) { |
| if (event.has_timestamp_us() && event.has_rtcp_packet() && |
| event.rtcp_packet().has_type() && |
| event.rtcp_packet().has_packet_data() && |
| event.rtcp_packet().packet_data().size() > 0) { |
| const webrtc::rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet(); |
| if (FLAGS_noaudio && rtcp_packet.type() == webrtc::rtclog::AUDIO) |
| continue; |
| if (FLAGS_novideo && rtcp_packet.type() == webrtc::rtclog::VIDEO) |
| continue; |
| if (FLAGS_nodata && rtcp_packet.type() == webrtc::rtclog::DATA) |
| continue; |
| if (!FLAGS_ssrc.empty()) { |
| const uint32_t packet_ssrc = |
| webrtc::ByteReader<uint32_t>::ReadBigEndian( |
| reinterpret_cast<const uint8_t*>( |
| rtcp_packet.packet_data().data() + 4)); |
| if (packet_ssrc != ssrc_filter) |
| continue; |
| } |
| |
| webrtc::test::RtpPacket packet; |
| packet.length = rtcp_packet.packet_data().size(); |
| if (packet.length > packet.kMaxPacketBufferSize) { |
| std::cout << "Skipping packet with size " << packet.length |
| << ", the maximum supported size is " |
| << packet.kMaxPacketBufferSize << std::endl; |
| continue; |
| } |
| // For RTCP packets the original_length should be set to 0 in the |
| // RTPdump format. |
| packet.original_length = 0; |
| packet.time_ms = event.timestamp_us() / 1000; |
| memcpy(packet.data, rtcp_packet.packet_data().data(), packet.length); |
| rtp_writer->WritePacket(&packet); |
| rtcp_counter++; |
| } else { |
| std::cout << "Skipping malformed event." << std::endl; |
| } |
| } |
| } |
| std::cout << "Wrote " << rtp_counter << (header_only ? " header-only" : "") |
| << " RTP packets and " << rtcp_counter << " RTCP packets to the " |
| << "output file." << std::endl; |
| return 0; |
| } |
