blob: 0f9d674ecee549139762bece67f36ee16224bd45 [file] [log] [blame]
/*
* 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 <iostream>
#include <map>
#include <sstream>
#include <string>
#include <utility> // pair
#include "webrtc/call/video_config.h"
#include "webrtc/common_types.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/flags.h"
namespace {
DEFINE_bool(config, true, "Use --noconfig to exclude stream configurations.");
DEFINE_bool(incoming, true, "Use --noincoming to exclude incoming packets.");
DEFINE_bool(outgoing, true, "Use --nooutgoing to exclude packets.");
// TODO(terelius): Note that the media type doesn't work with outgoing packets.
DEFINE_bool(audio, true, "Use --noaudio to exclude audio packets.");
// TODO(terelius): Note that the media type doesn't work with outgoing packets.
DEFINE_bool(video, true, "Use --novideo to exclude video packets.");
// TODO(terelius): Note that the media type doesn't work with outgoing packets.
DEFINE_bool(data, true, "Use --nodata to exclude data packets.");
DEFINE_bool(rtp, true, "Use --nortp to exclude RTP packets.");
DEFINE_bool(rtcp, true, "Use --nortcp to exclude RTCP packets.");
// TODO(terelius): Allow a list of SSRCs.
DEFINE_string(ssrc,
"",
"Print only packets with this SSRC (decimal or hex, the latter "
"starting with 0x).");
DEFINE_bool(help, false, "Prints this message.");
using MediaType = webrtc::ParsedRtcEventLog::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::TransportSequenceNumber>(
webrtc::RtpExtension::kTransportSequenceNumberDefaultId);
default_map.Register<webrtc::PlayoutDelayLimits>(
webrtc::RtpExtension::kPlayoutDelayDefaultId);
return default_map;
}
void PrintSenderReport(const webrtc::ParsedRtcEventLog& 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::ParsedRtcEventLog& 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::ParsedRtcEventLog& 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::ParsedRtcEventLog& 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::ParsedRtcEventLog& 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)
<< "\tssrc=" << transport_feedback.sender_ssrc() << std::endl;
break;
}
default:
break;
}
}
void PrintPsFeedback(const webrtc::ParsedRtcEventLog& 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:
break;
}
}
} // 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[]) {
std::string program_name = argv[0];
std::string usage =
"Tool for printing packet information from an RtcEventLog as text.\n"
"Run " +
program_name +
" --help for usage.\n"
"Example usage:\n" +
program_name + " input.rel\n";
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
FLAG_help || argc != 2) {
std::cout << usage;
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
std::string input_file = argv[1];
if (strlen(FLAG_ssrc) > 0)
RTC_CHECK(ParseSsrc(FLAG_ssrc)) << "Flag verification has failed.";
webrtc::RtpHeaderExtensionMap default_map = GetDefaultHeaderExtensionMap();
webrtc::ParsedRtcEventLog parsed_stream;
if (!parsed_stream.ParseFile(input_file)) {
std::cerr << "Error while parsing input file: " << input_file << std::endl;
return -1;
}
for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
if (FLAG_config && FLAG_video && FLAG_incoming &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
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;
}
if (FLAG_config && FLAG_video && FLAG_outgoing &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
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;
}
}
if (FLAG_config && FLAG_audio && FLAG_incoming &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
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;
}
if (FLAG_config && FLAG_audio && FLAG_outgoing &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
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;
}
if (FLAG_rtp &&
parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
size_t header_length;
size_t total_length;
uint8_t header[IP_PACKET_SIZE];
webrtc::PacketDirection direction;
webrtc::RtpHeaderExtensionMap* extension_map = parsed_stream.GetRtpHeader(
i, &direction, header, &header_length, &total_length);
if (extension_map == nullptr)
extension_map = &default_map;
// 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))
continue;
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=" << parsed_header.extension.audioLevel;
}
std::cout << std::endl;
}
if (FLAG_rtcp && parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::RTCP_EVENT) {
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)) {
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;
}
}
}
}
return 0;
}