| /* |
| * Copyright (c) 2017 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 <iomanip> // setfill, setw |
| #include <iostream> |
| #include <map> |
| #include <string> |
| #include <utility> // pair |
| |
| #include "common_types.h" // NOLINT(build/include) |
| #include "logging/rtc_event_log/rtc_event_log_parser_new.h" |
| #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/bye.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/fir.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/nack.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/pli.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/remb.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/sdes.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/tmmbn.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/tmmbr.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" |
| #include "modules/rtp_rtcp/source/rtp_header_extensions.h" |
| #include "modules/rtp_rtcp/source/rtp_utility.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/flags.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/strings/string_builder.h" |
| |
| namespace { |
| |
| WEBRTC_DEFINE_bool(unknown, true, "Use --nounknown to exclude unknown events."); |
| WEBRTC_DEFINE_bool(startstop, |
| true, |
| "Use --nostartstop to exclude start/stop events."); |
| WEBRTC_DEFINE_bool(config, |
| true, |
| "Use --noconfig to exclude stream configurations."); |
| WEBRTC_DEFINE_bool(bwe, true, "Use --nobwe to exclude BWE events."); |
| WEBRTC_DEFINE_bool(incoming, |
| true, |
| "Use --noincoming to exclude incoming packets."); |
| WEBRTC_DEFINE_bool(outgoing, true, "Use --nooutgoing to exclude packets."); |
| // TODO(terelius): Note that the media type doesn't work with outgoing packets. |
| WEBRTC_DEFINE_bool(audio, true, "Use --noaudio to exclude audio packets."); |
| // TODO(terelius): Note that the media type doesn't work with outgoing packets. |
| WEBRTC_DEFINE_bool(video, true, "Use --novideo to exclude video packets."); |
| // TODO(terelius): Note that the media type doesn't work with outgoing packets. |
| WEBRTC_DEFINE_bool(data, true, "Use --nodata to exclude data packets."); |
| WEBRTC_DEFINE_bool(rtp, true, "Use --nortp to exclude RTP packets."); |
| WEBRTC_DEFINE_bool(rtcp, true, "Use --nortcp to exclude RTCP packets."); |
| WEBRTC_DEFINE_bool(playout, |
| true, |
| "Use --noplayout to exclude audio playout events."); |
| WEBRTC_DEFINE_bool(ana, true, "Use --noana to exclude ANA events."); |
| WEBRTC_DEFINE_bool(probe, true, "Use --noprobe to exclude probe events."); |
| WEBRTC_DEFINE_bool(ice, true, "Use --noice to exclude ICE events."); |
| |
| WEBRTC_DEFINE_bool(print_full_packets, |
| false, |
| "Print the full RTP headers and RTCP packets in hex."); |
| |
| // TODO(terelius): Allow a list of SSRCs. |
| WEBRTC_DEFINE_string( |
| ssrc, |
| "", |
| "Print only packets with this SSRC (decimal or hex, the latter " |
| "starting with 0x)."); |
| WEBRTC_DEFINE_bool(help, false, "Prints this message."); |
| |
| using MediaType = webrtc::ParsedRtcEventLogNew::MediaType; |
| |
| static uint32_t filtered_ssrc = 0; |
| |
| // Parses the input string for a valid SSRC. If a valid SSRC is found, it is |
| // written to the static global variable |filtered_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) { |
| // 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 >> filtered_ssrc; |
| return str.empty() || (!ss.fail() && ss.eof()); |
| } |
| |
| bool ExcludePacket(webrtc::PacketDirection direction, |
| MediaType media_type, |
| uint32_t packet_ssrc) { |
| if (!FLAG_outgoing && direction == webrtc::kOutgoingPacket) |
| return true; |
| if (!FLAG_incoming && direction == webrtc::kIncomingPacket) |
| return true; |
| if (!FLAG_audio && media_type == MediaType::AUDIO) |
| return true; |
| if (!FLAG_video && media_type == MediaType::VIDEO) |
| return true; |
| if (!FLAG_data && media_type == MediaType::DATA) |
| return true; |
| if (strlen(FLAG_ssrc) > 0 && packet_ssrc != filtered_ssrc) |
| return true; |
| return false; |
| } |
| |
| const char* StreamInfo(webrtc::PacketDirection direction, |
| MediaType media_type) { |
| if (direction == webrtc::kOutgoingPacket) { |
| if (media_type == MediaType::AUDIO) |
| return "(out,audio)"; |
| else if (media_type == MediaType::VIDEO) |
| return "(out,video)"; |
| else if (media_type == MediaType::DATA) |
| return "(out,data)"; |
| else |
| return "(out)"; |
| } |
| if (direction == webrtc::kIncomingPacket) { |
| if (media_type == MediaType::AUDIO) |
| return "(in,audio)"; |
| else if (media_type == MediaType::VIDEO) |
| return "(in,video)"; |
| else if (media_type == MediaType::DATA) |
| return "(in,data)"; |
| else |
| return "(in)"; |
| } |
| return "(unknown)"; |
| } |
| |
| // Return default values for header extensions, to use on streams without stored |
| // mapping data. Currently this only applies to audio streams, since the mapping |
| // is not stored in the event log. |
| // TODO(ivoc): Remove this once this mapping is stored in the event log for |
| // audio streams. Tracking bug: webrtc:6399 |
| webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() { |
| webrtc::RtpHeaderExtensionMap default_map; |
| default_map.Register<webrtc::AudioLevel>( |
| webrtc::RtpExtension::kAudioLevelDefaultId); |
| default_map.Register<webrtc::TransmissionOffset>( |
| webrtc::RtpExtension::kTimestampOffsetDefaultId); |
| default_map.Register<webrtc::AbsoluteSendTime>( |
| webrtc::RtpExtension::kAbsSendTimeDefaultId); |
| default_map.Register<webrtc::VideoOrientation>( |
| webrtc::RtpExtension::kVideoRotationDefaultId); |
| default_map.Register<webrtc::VideoContentTypeExtension>( |
| webrtc::RtpExtension::kVideoContentTypeDefaultId); |
| default_map.Register<webrtc::VideoTimingExtension>( |
| webrtc::RtpExtension::kVideoTimingDefaultId); |
| default_map.Register<webrtc::FrameMarkingExtension>( |
| webrtc::RtpExtension::kFrameMarkingDefaultId); |
| default_map.Register<webrtc::TransportSequenceNumber>( |
| webrtc::RtpExtension::kTransportSequenceNumberDefaultId); |
| default_map.Register<webrtc::PlayoutDelayLimits>( |
| webrtc::RtpExtension::kPlayoutDelayDefaultId); |
| return default_map; |
| } |
| |
| void PrintSenderReport(const webrtc::ParsedRtcEventLogNew& parsed_stream, |
| const webrtc::rtcp::CommonHeader& rtcp_block, |
| uint64_t log_timestamp, |
| webrtc::PacketDirection direction) { |
| webrtc::rtcp::SenderReport sr; |
| if (!sr.Parse(rtcp_block)) |
| return; |
| MediaType media_type = |
| parsed_stream.GetMediaType(sr.sender_ssrc(), direction); |
| if (ExcludePacket(direction, media_type, sr.sender_ssrc())) |
| return; |
| std::cout << log_timestamp << "\t" |
| << "RTCP_SR" << StreamInfo(direction, media_type) |
| << "\tssrc=" << sr.sender_ssrc() |
| << "\ttimestamp=" << sr.rtp_timestamp() << std::endl; |
| } |
| |
| void PrintReceiverReport(const webrtc::ParsedRtcEventLogNew& parsed_stream, |
| const webrtc::rtcp::CommonHeader& rtcp_block, |
| uint64_t log_timestamp, |
| webrtc::PacketDirection direction) { |
| webrtc::rtcp::ReceiverReport rr; |
| if (!rr.Parse(rtcp_block)) |
| return; |
| MediaType media_type = |
| parsed_stream.GetMediaType(rr.sender_ssrc(), direction); |
| if (ExcludePacket(direction, media_type, rr.sender_ssrc())) |
| return; |
| std::cout << log_timestamp << "\t" |
| << "RTCP_RR" << StreamInfo(direction, media_type) |
| << "\tssrc=" << rr.sender_ssrc() << std::endl; |
| } |
| |
| void PrintXr(const webrtc::ParsedRtcEventLogNew& parsed_stream, |
| const webrtc::rtcp::CommonHeader& rtcp_block, |
| uint64_t log_timestamp, |
| webrtc::PacketDirection direction) { |
| webrtc::rtcp::ExtendedReports xr; |
| if (!xr.Parse(rtcp_block)) |
| return; |
| MediaType media_type = |
| parsed_stream.GetMediaType(xr.sender_ssrc(), direction); |
| if (ExcludePacket(direction, media_type, xr.sender_ssrc())) |
| return; |
| std::cout << log_timestamp << "\t" |
| << "RTCP_XR" << StreamInfo(direction, media_type) |
| << "\tssrc=" << xr.sender_ssrc() << std::endl; |
| } |
| |
| void PrintSdes(const webrtc::rtcp::CommonHeader& rtcp_block, |
| uint64_t log_timestamp, |
| webrtc::PacketDirection direction) { |
| std::cout << log_timestamp << "\t" |
| << "RTCP_SDES" << StreamInfo(direction, MediaType::ANY) |
| << std::endl; |
| RTC_NOTREACHED() << "SDES should have been redacted when writing the log"; |
| } |
| |
| void PrintBye(const webrtc::ParsedRtcEventLogNew& parsed_stream, |
| const webrtc::rtcp::CommonHeader& rtcp_block, |
| uint64_t log_timestamp, |
| webrtc::PacketDirection direction) { |
| webrtc::rtcp::Bye bye; |
| if (!bye.Parse(rtcp_block)) |
| return; |
| MediaType media_type = |
| parsed_stream.GetMediaType(bye.sender_ssrc(), direction); |
| if (ExcludePacket(direction, media_type, bye.sender_ssrc())) |
| return; |
| std::cout << log_timestamp << "\t" |
| << "RTCP_BYE" << StreamInfo(direction, media_type) |
| << "\tssrc=" << bye.sender_ssrc() << std::endl; |
| } |
| |
| void PrintRtpFeedback(const webrtc::ParsedRtcEventLogNew& parsed_stream, |
| const webrtc::rtcp::CommonHeader& rtcp_block, |
| uint64_t log_timestamp, |
| webrtc::PacketDirection direction) { |
| switch (rtcp_block.fmt()) { |
| case webrtc::rtcp::Nack::kFeedbackMessageType: { |
| webrtc::rtcp::Nack nack; |
| if (!nack.Parse(rtcp_block)) |
| return; |
| MediaType media_type = |
| parsed_stream.GetMediaType(nack.sender_ssrc(), direction); |
| if (ExcludePacket(direction, media_type, nack.sender_ssrc())) |
| return; |
| std::cout << log_timestamp << "\t" |
| << "RTCP_NACK" << StreamInfo(direction, media_type) |
| << "\tssrc=" << nack.sender_ssrc() << std::endl; |
| break; |
| } |
| case webrtc::rtcp::Tmmbr::kFeedbackMessageType: { |
| webrtc::rtcp::Tmmbr tmmbr; |
| if (!tmmbr.Parse(rtcp_block)) |
| return; |
| MediaType media_type = |
| parsed_stream.GetMediaType(tmmbr.sender_ssrc(), direction); |
| if (ExcludePacket(direction, media_type, tmmbr.sender_ssrc())) |
| return; |
| std::cout << log_timestamp << "\t" |
| << "RTCP_TMMBR" << StreamInfo(direction, media_type) |
| << "\tssrc=" << tmmbr.sender_ssrc() << std::endl; |
| break; |
| } |
| case webrtc::rtcp::Tmmbn::kFeedbackMessageType: { |
| webrtc::rtcp::Tmmbn tmmbn; |
| if (!tmmbn.Parse(rtcp_block)) |
| return; |
| MediaType media_type = |
| parsed_stream.GetMediaType(tmmbn.sender_ssrc(), direction); |
| if (ExcludePacket(direction, media_type, tmmbn.sender_ssrc())) |
| return; |
| std::cout << log_timestamp << "\t" |
| << "RTCP_TMMBN" << StreamInfo(direction, media_type) |
| << "\tssrc=" << tmmbn.sender_ssrc() << std::endl; |
| break; |
| } |
| case webrtc::rtcp::RapidResyncRequest::kFeedbackMessageType: { |
| webrtc::rtcp::RapidResyncRequest sr_req; |
| if (!sr_req.Parse(rtcp_block)) |
| return; |
| MediaType media_type = |
| parsed_stream.GetMediaType(sr_req.sender_ssrc(), direction); |
| if (ExcludePacket(direction, media_type, sr_req.sender_ssrc())) |
| return; |
| std::cout << log_timestamp << "\t" |
| << "RTCP_SRREQ" << StreamInfo(direction, media_type) |
| << "\tssrc=" << sr_req.sender_ssrc() << std::endl; |
| break; |
| } |
| case webrtc::rtcp::TransportFeedback::kFeedbackMessageType: { |
| webrtc::rtcp::TransportFeedback transport_feedback; |
| if (!transport_feedback.Parse(rtcp_block)) |
| return; |
| MediaType media_type = parsed_stream.GetMediaType( |
| transport_feedback.sender_ssrc(), direction); |
| if (ExcludePacket(direction, media_type, |
| transport_feedback.sender_ssrc())) |
| return; |
| std::cout << log_timestamp << "\t" |
| << "RTCP_NEWFB" << StreamInfo(direction, media_type) |
| << "\tsender_ssrc=" << transport_feedback.sender_ssrc() |
| << "\tmedia_ssrc=" << transport_feedback.media_ssrc() |
| << std::endl; |
| break; |
| } |
| default: |
| std::cout << log_timestamp << "\t" |
| << "RTCP_RTPFB(UNKNOWN)" << std::endl; |
| break; |
| } |
| } |
| |
| void PrintPsFeedback(const webrtc::ParsedRtcEventLogNew& parsed_stream, |
| const webrtc::rtcp::CommonHeader& rtcp_block, |
| uint64_t log_timestamp, |
| webrtc::PacketDirection direction) { |
| switch (rtcp_block.fmt()) { |
| case webrtc::rtcp::Pli::kFeedbackMessageType: { |
| webrtc::rtcp::Pli pli; |
| if (!pli.Parse(rtcp_block)) |
| return; |
| MediaType media_type = |
| parsed_stream.GetMediaType(pli.sender_ssrc(), direction); |
| if (ExcludePacket(direction, media_type, pli.sender_ssrc())) |
| return; |
| std::cout << log_timestamp << "\t" |
| << "RTCP_PLI" << StreamInfo(direction, media_type) |
| << "\tssrc=" << pli.sender_ssrc() << std::endl; |
| break; |
| } |
| case webrtc::rtcp::Fir::kFeedbackMessageType: { |
| webrtc::rtcp::Fir fir; |
| if (!fir.Parse(rtcp_block)) |
| return; |
| MediaType media_type = |
| parsed_stream.GetMediaType(fir.sender_ssrc(), direction); |
| if (ExcludePacket(direction, media_type, fir.sender_ssrc())) |
| return; |
| std::cout << log_timestamp << "\t" |
| << "RTCP_FIR" << StreamInfo(direction, media_type) |
| << "\tssrc=" << fir.sender_ssrc() << std::endl; |
| break; |
| } |
| case webrtc::rtcp::Remb::kFeedbackMessageType: { |
| webrtc::rtcp::Remb remb; |
| if (!remb.Parse(rtcp_block)) |
| return; |
| MediaType media_type = |
| parsed_stream.GetMediaType(remb.sender_ssrc(), direction); |
| if (ExcludePacket(direction, media_type, remb.sender_ssrc())) |
| return; |
| std::cout << log_timestamp << "\t" |
| << "RTCP_REMB" << StreamInfo(direction, media_type) |
| << "\tssrc=" << remb.sender_ssrc() << std::endl; |
| break; |
| } |
| default: |
| std::cout << log_timestamp << "\t" |
| << "RTCP_PSFB(UNKNOWN)" << std::endl; |
| break; |
| } |
| } |
| |
| enum class InputSource { |
| STDIN, |
| FILE, |
| }; |
| |
| void PrintUsageGuide(const std::string& program_name) { |
| std::cout |
| << "Tool for printing packet information from an RtcEventLog as text.\n" |
| << "* Run " + program_name + " --help for usage.\n" |
| << "* Example usage for parsing a file:\n" |
| << " " << program_name + " input.rel\n" |
| << "* Example usage for parsing the stdin:\n" |
| << " " << program_name + "\n"; |
| } |
| |
| // TODO(eladalon): Return a stream or file descriptor instead. |
| InputSource ParseCommandLineFlags(int argc, char* argv[]) { |
| if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) != 0) { |
| PrintUsageGuide(argv[0]); |
| exit(-1); |
| } |
| |
| if (FLAG_help) { |
| PrintUsageGuide(argv[0]); |
| std::cout << std::endl; |
| rtc::FlagList::Print(nullptr, false); |
| exit(0); |
| } |
| |
| switch (argc) { |
| case 1: |
| return InputSource::STDIN; |
| case 2: |
| return InputSource::FILE; |
| default: |
| PrintUsageGuide(argv[0]); |
| exit(-1); |
| } |
| } |
| |
| } // namespace |
| |
| // This utility will print basic information about each packet to stdout. |
| // Note that parser will assert if the protobuf event is missing some required |
| // fields and we attempt to access them. We don't handle this at the moment. |
| int main(int argc, char* argv[]) { |
| InputSource input_source = ParseCommandLineFlags(argc, argv); |
| |
| if (strlen(FLAG_ssrc) > 0) |
| RTC_CHECK(ParseSsrc(FLAG_ssrc)) << "Flag verification has failed."; |
| |
| webrtc::RtpHeaderExtensionMap default_map = GetDefaultHeaderExtensionMap(); |
| bool default_map_used = false; |
| |
| webrtc::ParsedRtcEventLogNew parsed_stream; |
| |
| switch (input_source) { |
| case InputSource::STDIN: { |
| if (!parsed_stream.ParseStream(std::cin)) { |
| std::cerr << "Error while parsing input stream." << std::endl; |
| return -1; |
| } |
| break; |
| } |
| case InputSource::FILE: { |
| if (!parsed_stream.ParseFile(argv[1])) { |
| std::cerr << "Error while parsing input file: " << argv[1] << std::endl; |
| return -1; |
| } |
| break; |
| } |
| default: { RTC_NOTREACHED() << "Unsupported input source."; } |
| } |
| |
| for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) { |
| bool event_recognized = false; |
| switch (parsed_stream.GetEventType(i)) { |
| case webrtc::ParsedRtcEventLogNew::EventType::UNKNOWN_EVENT: { |
| if (FLAG_unknown) { |
| std::cout << parsed_stream.GetTimestamp(i) << "\tUNKNOWN_EVENT" |
| << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::LOG_START: { |
| if (FLAG_startstop) { |
| std::cout << parsed_stream.GetTimestamp(i) << "\tLOG_START" |
| << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::LOG_END: { |
| if (FLAG_startstop) { |
| std::cout << parsed_stream.GetTimestamp(i) << "\tLOG_END" |
| << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::RTP_EVENT: { |
| if (FLAG_rtp) { |
| size_t header_length; |
| size_t total_length; |
| uint8_t header[IP_PACKET_SIZE]; |
| webrtc::PacketDirection direction; |
| const webrtc::RtpHeaderExtensionMap* extension_map = |
| parsed_stream.GetRtpHeader(i, &direction, header, &header_length, |
| &total_length, nullptr); |
| |
| if (extension_map == nullptr) { |
| extension_map = &default_map; |
| if (!default_map_used) |
| RTC_LOG(LS_WARNING) << "Using default header extension map"; |
| default_map_used = true; |
| } |
| |
| // Parse header to get SSRC and RTP time. |
| webrtc::RtpUtility::RtpHeaderParser rtp_parser(header, header_length); |
| webrtc::RTPHeader parsed_header; |
| rtp_parser.Parse(&parsed_header, extension_map); |
| MediaType media_type = |
| parsed_stream.GetMediaType(parsed_header.ssrc, direction); |
| |
| if (ExcludePacket(direction, media_type, parsed_header.ssrc)) { |
| event_recognized = true; |
| break; |
| } |
| |
| std::cout << parsed_stream.GetTimestamp(i) << "\tRTP" |
| << StreamInfo(direction, media_type) |
| << "\tssrc=" << parsed_header.ssrc |
| << "\ttimestamp=" << parsed_header.timestamp; |
| if (parsed_header.extension.hasAbsoluteSendTime) { |
| std::cout << "\tAbsSendTime=" |
| << parsed_header.extension.absoluteSendTime; |
| } |
| if (parsed_header.extension.hasVideoContentType) { |
| std::cout << "\tContentType=" |
| << static_cast<int>( |
| parsed_header.extension.videoContentType); |
| } |
| if (parsed_header.extension.hasVideoRotation) { |
| std::cout << "\tRotation=" |
| << static_cast<int>( |
| parsed_header.extension.videoRotation); |
| } |
| if (parsed_header.extension.hasTransportSequenceNumber) { |
| std::cout << "\tTransportSeq=" |
| << parsed_header.extension.transportSequenceNumber; |
| } |
| if (parsed_header.extension.hasTransmissionTimeOffset) { |
| std::cout << "\tTransmTimeOffset=" |
| << parsed_header.extension.transmissionTimeOffset; |
| } |
| if (parsed_header.extension.hasAudioLevel) { |
| std::cout << "\tAudioLevel=" |
| << static_cast<int>(parsed_header.extension.audioLevel); |
| } |
| std::cout << std::endl; |
| if (FLAG_print_full_packets) { |
| // TODO(terelius): Rewrite this file to use printf instead of cout. |
| std::cout << "\t\t" << std::hex; |
| char prev_fill = std::cout.fill('0'); |
| for (size_t i = 0; i < header_length; i++) { |
| std::cout << std::setw(2) << static_cast<unsigned>(header[i]); |
| if (i % 4 == 3) |
| std::cout << " "; // Separator between 32-bit words. |
| } |
| std::cout.fill(prev_fill); |
| std::cout << std::dec << std::endl; |
| } |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::RTCP_EVENT: { |
| if (FLAG_rtcp) { |
| size_t length; |
| uint8_t packet[IP_PACKET_SIZE]; |
| webrtc::PacketDirection direction; |
| parsed_stream.GetRtcpPacket(i, &direction, packet, &length); |
| |
| webrtc::rtcp::CommonHeader rtcp_block; |
| const uint8_t* packet_end = packet + length; |
| for (const uint8_t* next_block = packet; next_block != packet_end; |
| next_block = rtcp_block.NextPacket()) { |
| ptrdiff_t remaining_blocks_size = packet_end - next_block; |
| RTC_DCHECK_GT(remaining_blocks_size, 0); |
| if (!rtcp_block.Parse(next_block, remaining_blocks_size)) { |
| RTC_LOG(LS_WARNING) << "Failed to parse RTCP"; |
| break; |
| } |
| |
| uint64_t log_timestamp = parsed_stream.GetTimestamp(i); |
| switch (rtcp_block.type()) { |
| case webrtc::rtcp::SenderReport::kPacketType: |
| PrintSenderReport(parsed_stream, rtcp_block, log_timestamp, |
| direction); |
| break; |
| case webrtc::rtcp::ReceiverReport::kPacketType: |
| PrintReceiverReport(parsed_stream, rtcp_block, log_timestamp, |
| direction); |
| break; |
| case webrtc::rtcp::Sdes::kPacketType: |
| PrintSdes(rtcp_block, log_timestamp, direction); |
| break; |
| case webrtc::rtcp::ExtendedReports::kPacketType: |
| PrintXr(parsed_stream, rtcp_block, log_timestamp, direction); |
| break; |
| case webrtc::rtcp::Bye::kPacketType: |
| PrintBye(parsed_stream, rtcp_block, log_timestamp, direction); |
| break; |
| case webrtc::rtcp::Rtpfb::kPacketType: |
| PrintRtpFeedback(parsed_stream, rtcp_block, log_timestamp, |
| direction); |
| break; |
| case webrtc::rtcp::Psfb::kPacketType: |
| PrintPsFeedback(parsed_stream, rtcp_block, log_timestamp, |
| direction); |
| break; |
| default: |
| break; |
| } |
| if (FLAG_print_full_packets) { |
| std::cout << "\t\t" << std::hex; |
| char prev_fill = std::cout.fill('0'); |
| for (const uint8_t* p = next_block; p < rtcp_block.NextPacket(); |
| p++) { |
| std::cout << std::setw(2) << static_cast<unsigned>(*p); |
| ptrdiff_t chars_printed = p - next_block; |
| if (chars_printed % 4 == 3) |
| std::cout << " "; // Separator between 32-bit words. |
| } |
| std::cout.fill(prev_fill); |
| std::cout << std::dec << std::endl; |
| } |
| } |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::AUDIO_PLAYOUT_EVENT: { |
| if (FLAG_playout) { |
| auto audio_playout = parsed_stream.GetAudioPlayout(i); |
| std::cout << audio_playout.log_time_us() << "\tAUDIO_PLAYOUT" |
| << "\tssrc=" << audio_playout.ssrc << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::LOSS_BASED_BWE_UPDATE: { |
| if (FLAG_bwe) { |
| auto bwe_update = parsed_stream.GetLossBasedBweUpdate(i); |
| std::cout << bwe_update.log_time_us() << "\tBWE(LOSS_BASED)" |
| << "\tbitrate_bps=" << bwe_update.bitrate_bps |
| << "\tfraction_lost=" |
| << static_cast<unsigned>(bwe_update.fraction_lost) |
| << "\texpected_packets=" << bwe_update.expected_packets |
| << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::DELAY_BASED_BWE_UPDATE: { |
| if (FLAG_bwe) { |
| auto bwe_update = parsed_stream.GetDelayBasedBweUpdate(i); |
| std::cout << bwe_update.log_time_us() << "\tBWE(DELAY_BASED)" |
| << "\tbitrate_bps=" << bwe_update.bitrate_bps |
| << "\tdetector_state=" |
| << static_cast<int>(bwe_update.detector_state) << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType:: |
| VIDEO_RECEIVER_CONFIG_EVENT: { |
| if (FLAG_config && FLAG_video && FLAG_incoming) { |
| webrtc::rtclog::StreamConfig config = |
| parsed_stream.GetVideoReceiveConfig(i); |
| std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG" |
| << "\tssrc=" << config.remote_ssrc |
| << "\tfeedback_ssrc=" << config.local_ssrc; |
| std::cout << "\textensions={"; |
| for (const auto& extension : config.rtp_extensions) { |
| std::cout << extension.ToString() << ","; |
| } |
| std::cout << "}"; |
| std::cout << "\tcodecs={"; |
| for (const auto& codec : config.codecs) { |
| std::cout << "{name: " << codec.payload_name |
| << ", payload_type: " << codec.payload_type |
| << ", rtx_payload_type: " << codec.rtx_payload_type |
| << "}"; |
| } |
| std::cout << "}" << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::VIDEO_SENDER_CONFIG_EVENT: { |
| if (FLAG_config && FLAG_video && FLAG_outgoing) { |
| std::vector<webrtc::rtclog::StreamConfig> configs = |
| parsed_stream.GetVideoSendConfig(i); |
| for (const auto& config : configs) { |
| std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG"; |
| std::cout << "\tssrcs=" << config.local_ssrc; |
| std::cout << "\trtx_ssrcs=" << config.rtx_ssrc; |
| std::cout << "\textensions={"; |
| for (const auto& extension : config.rtp_extensions) { |
| std::cout << extension.ToString() << ","; |
| } |
| std::cout << "}"; |
| std::cout << "\tcodecs={"; |
| for (const auto& codec : config.codecs) { |
| std::cout << "{name: " << codec.payload_name |
| << ", payload_type: " << codec.payload_type |
| << ", rtx_payload_type: " << codec.rtx_payload_type |
| << "}"; |
| } |
| std::cout << "}" << std::endl; |
| } |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType:: |
| AUDIO_RECEIVER_CONFIG_EVENT: { |
| if (FLAG_config && FLAG_audio && FLAG_incoming) { |
| webrtc::rtclog::StreamConfig config = |
| parsed_stream.GetAudioReceiveConfig(i); |
| std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG" |
| << "\tssrc=" << config.remote_ssrc |
| << "\tfeedback_ssrc=" << config.local_ssrc; |
| std::cout << "\textensions={"; |
| for (const auto& extension : config.rtp_extensions) { |
| std::cout << extension.ToString() << ","; |
| } |
| std::cout << "}"; |
| std::cout << "\tcodecs={"; |
| for (const auto& codec : config.codecs) { |
| std::cout << "{name: " << codec.payload_name |
| << ", payload_type: " << codec.payload_type |
| << ", rtx_payload_type: " << codec.rtx_payload_type |
| << "}"; |
| } |
| std::cout << "}" << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::AUDIO_SENDER_CONFIG_EVENT: { |
| if (FLAG_config && FLAG_audio && FLAG_outgoing) { |
| webrtc::rtclog::StreamConfig config = |
| parsed_stream.GetAudioSendConfig(i); |
| std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG" |
| << "\tssrc=" << config.local_ssrc; |
| std::cout << "\textensions={"; |
| for (const auto& extension : config.rtp_extensions) { |
| std::cout << extension.ToString() << ","; |
| } |
| std::cout << "}"; |
| std::cout << "\tcodecs={"; |
| for (const auto& codec : config.codecs) { |
| std::cout << "{name: " << codec.payload_name |
| << ", payload_type: " << codec.payload_type |
| << ", rtx_payload_type: " << codec.rtx_payload_type |
| << "}"; |
| } |
| std::cout << "}" << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType:: |
| AUDIO_NETWORK_ADAPTATION_EVENT: { |
| if (FLAG_ana) { |
| auto ana_event = parsed_stream.GetAudioNetworkAdaptation(i); |
| char buffer[300]; |
| rtc::SimpleStringBuilder builder(buffer); |
| builder << parsed_stream.GetTimestamp(i) << "\tANA_UPDATE"; |
| if (ana_event.config.bitrate_bps) { |
| builder << "\tbitrate_bps=" << *ana_event.config.bitrate_bps; |
| } |
| if (ana_event.config.frame_length_ms) { |
| builder << "\tframe_length_ms=" |
| << *ana_event.config.frame_length_ms; |
| } |
| if (ana_event.config.uplink_packet_loss_fraction) { |
| builder << "\tuplink_packet_loss_fraction=" |
| << *ana_event.config.uplink_packet_loss_fraction; |
| } |
| if (ana_event.config.enable_fec) { |
| builder << "\tenable_fec=" << *ana_event.config.enable_fec; |
| } |
| if (ana_event.config.enable_dtx) { |
| builder << "\tenable_dtx=" << *ana_event.config.enable_dtx; |
| } |
| if (ana_event.config.num_channels) { |
| builder << "\tnum_channels=" << *ana_event.config.num_channels; |
| } |
| std::cout << builder.str() << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType:: |
| BWE_PROBE_CLUSTER_CREATED_EVENT: { |
| if (FLAG_probe) { |
| auto probe_event = parsed_stream.GetBweProbeClusterCreated(i); |
| std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_CREATED(" |
| << probe_event.id << ")" |
| << "\tbitrate_bps=" << probe_event.bitrate_bps |
| << "\tmin_packets=" << probe_event.min_packets |
| << "\tmin_bytes=" << probe_event.min_bytes << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::BWE_PROBE_FAILURE_EVENT: { |
| if (FLAG_probe) { |
| webrtc::LoggedBweProbeFailureEvent probe_result = |
| parsed_stream.GetBweProbeFailure(i); |
| std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_FAILURE(" |
| << probe_result.id << ")" |
| << "\tfailure_reason=" |
| << static_cast<int>(probe_result.failure_reason) |
| << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::BWE_PROBE_SUCCESS_EVENT: { |
| if (FLAG_probe) { |
| webrtc::LoggedBweProbeSuccessEvent probe_result = |
| parsed_stream.GetBweProbeSuccess(i); |
| std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_SUCCESS(" |
| << probe_result.id << ")" |
| << "\tbitrate_bps=" << probe_result.bitrate_bps |
| << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::ALR_STATE_EVENT: { |
| if (FLAG_bwe) { |
| webrtc::LoggedAlrStateEvent alr_state = parsed_stream.GetAlrState(i); |
| std::cout << parsed_stream.GetTimestamp(i) << "\tALR_STATE" |
| << "\tin_alr=" << alr_state.in_alr << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_CONFIG: { |
| if (FLAG_ice) { |
| webrtc::LoggedIceCandidatePairConfig ice_cp_config = |
| parsed_stream.GetIceCandidatePairConfig(i); |
| // TODO(qingsi): convert the numeric representation of states to text |
| std::cout << parsed_stream.GetTimestamp(i) |
| << "\tICE_CANDIDATE_PAIR_CONFIG" |
| << "\ttype=" << static_cast<int>(ice_cp_config.type) |
| << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| |
| case webrtc::ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_EVENT: { |
| if (FLAG_ice) { |
| webrtc::LoggedIceCandidatePairEvent ice_cp_event = |
| parsed_stream.GetIceCandidatePairEvent(i); |
| // TODO(qingsi): convert the numeric representation of states to text |
| std::cout << parsed_stream.GetTimestamp(i) |
| << "\tICE_CANDIDATE_PAIR_EVENT" |
| << "\ttype=" << static_cast<int>(ice_cp_event.type) |
| << std::endl; |
| } |
| event_recognized = true; |
| break; |
| } |
| } |
| |
| if (!event_recognized) { |
| std::cout << "Unrecognized event (" |
| << static_cast<int>(parsed_stream.GetEventType(i)) << ")" |
| << std::endl; |
| } |
| } |
| return 0; |
| } |
