blob: 47b4c1e76b6ff356f119fb3d50d4e4488b22beca [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 "logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h"
#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
#include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
#include "logging/rtc_event_log/events/rtc_event_logging_started.h"
#include "logging/rtc_event_log/events/rtc_event_logging_stopped.h"
#include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
#include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h"
#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
#include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
#include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet/app.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_jitter_report.h"
#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.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/rtp_packet.h"
#include "rtc_base/checks.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/logging.h"
#ifdef ENABLE_RTC_EVENT_LOG
// *.pb.h files are generated at build-time by the protobuf compiler.
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
#else
#include "logging/rtc_event_log/rtc_event_log.pb.h"
#endif
RTC_POP_IGNORING_WUNDEF()
namespace webrtc {
namespace {
rtclog::DelayBasedBweUpdate::DetectorState ConvertDetectorState(
BandwidthUsage state) {
switch (state) {
case BandwidthUsage::kBwNormal:
return rtclog::DelayBasedBweUpdate::BWE_NORMAL;
case BandwidthUsage::kBwUnderusing:
return rtclog::DelayBasedBweUpdate::BWE_UNDERUSING;
case BandwidthUsage::kBwOverusing:
return rtclog::DelayBasedBweUpdate::BWE_OVERUSING;
case BandwidthUsage::kLast:
RTC_NOTREACHED();
}
RTC_NOTREACHED();
return rtclog::DelayBasedBweUpdate::BWE_NORMAL;
}
rtclog::BweProbeResult::ResultType ConvertProbeResultType(
ProbeFailureReason failure_reason) {
switch (failure_reason) {
case ProbeFailureReason::kInvalidSendReceiveInterval:
return rtclog::BweProbeResult::INVALID_SEND_RECEIVE_INTERVAL;
case ProbeFailureReason::kInvalidSendReceiveRatio:
return rtclog::BweProbeResult::INVALID_SEND_RECEIVE_RATIO;
case ProbeFailureReason::kTimeout:
return rtclog::BweProbeResult::TIMEOUT;
case ProbeFailureReason::kLast:
RTC_NOTREACHED();
}
RTC_NOTREACHED();
return rtclog::BweProbeResult::SUCCESS;
}
rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(RtcpMode rtcp_mode) {
switch (rtcp_mode) {
case RtcpMode::kCompound:
return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
case RtcpMode::kReducedSize:
return rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE;
case RtcpMode::kOff:
RTC_NOTREACHED();
}
RTC_NOTREACHED();
return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
}
} // namespace
std::string RtcEventLogEncoderLegacy::Encode(const RtcEvent& event) {
switch (event.GetType()) {
case RtcEvent::Type::AudioNetworkAdaptation: {
auto& rtc_event =
static_cast<const RtcEventAudioNetworkAdaptation&>(event);
return EncodeAudioNetworkAdaptation(rtc_event);
}
case RtcEvent::Type::AudioPlayout: {
auto& rtc_event = static_cast<const RtcEventAudioPlayout&>(event);
return EncodeAudioPlayout(rtc_event);
}
case RtcEvent::Type::AudioReceiveStreamConfig: {
auto& rtc_event =
static_cast<const RtcEventAudioReceiveStreamConfig&>(event);
return EncodeAudioReceiveStreamConfig(rtc_event);
}
case RtcEvent::Type::AudioSendStreamConfig: {
auto& rtc_event =
static_cast<const RtcEventAudioSendStreamConfig&>(event);
return EncodeAudioSendStreamConfig(rtc_event);
}
case RtcEvent::Type::BweUpdateDelayBased: {
auto& rtc_event = static_cast<const RtcEventBweUpdateDelayBased&>(event);
return EncodeBweUpdateDelayBased(rtc_event);
}
case RtcEvent::Type::BweUpdateLossBased: {
auto& rtc_event = static_cast<const RtcEventBweUpdateLossBased&>(event);
return EncodeBweUpdateLossBased(rtc_event);
}
case RtcEvent::Type::LoggingStarted: {
auto& rtc_event = static_cast<const RtcEventLoggingStarted&>(event);
return EncodeLoggingStarted(rtc_event);
}
case RtcEvent::Type::LoggingStopped: {
auto& rtc_event = static_cast<const RtcEventLoggingStopped&>(event);
return EncodeLoggingStopped(rtc_event);
}
case RtcEvent::Type::ProbeClusterCreated: {
auto& rtc_event = static_cast<const RtcEventProbeClusterCreated&>(event);
return EncodeProbeClusterCreated(rtc_event);
}
case RtcEvent::Type::ProbeResultFailure: {
auto& rtc_event = static_cast<const RtcEventProbeResultFailure&>(event);
return EncodeProbeResultFailure(rtc_event);
}
case RtcEvent::Type::ProbeResultSuccess: {
auto& rtc_event = static_cast<const RtcEventProbeResultSuccess&>(event);
return EncodeProbeResultSuccess(rtc_event);
}
case RtcEvent::Type::RtcpPacketIncoming: {
auto& rtc_event = static_cast<const RtcEventRtcpPacketIncoming&>(event);
return EncodeRtcpPacketIncoming(rtc_event);
}
case RtcEvent::Type::RtcpPacketOutgoing: {
auto& rtc_event = static_cast<const RtcEventRtcpPacketOutgoing&>(event);
return EncodeRtcpPacketOutgoing(rtc_event);
}
case RtcEvent::Type::RtpPacketIncoming: {
auto& rtc_event = static_cast<const RtcEventRtpPacketIncoming&>(event);
return EncodeRtpPacketIncoming(rtc_event);
}
case RtcEvent::Type::RtpPacketOutgoing: {
auto& rtc_event = static_cast<const RtcEventRtpPacketOutgoing&>(event);
return EncodeRtpPacketOutgoing(rtc_event);
}
case RtcEvent::Type::VideoReceiveStreamConfig: {
auto& rtc_event =
static_cast<const RtcEventVideoReceiveStreamConfig&>(event);
return EncodeVideoReceiveStreamConfig(rtc_event);
}
case RtcEvent::Type::VideoSendStreamConfig: {
auto& rtc_event =
static_cast<const RtcEventVideoSendStreamConfig&>(event);
return EncodeVideoSendStreamConfig(rtc_event);
}
}
int event_type = static_cast<int>(event.GetType());
RTC_NOTREACHED() << "Unknown event type (" << event_type << ")";
return "";
}
std::string RtcEventLogEncoderLegacy::EncodeAudioNetworkAdaptation(
const RtcEventAudioNetworkAdaptation& event) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(event.timestamp_us_);
rtclog_event.set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
auto audio_network_adaptation =
rtclog_event.mutable_audio_network_adaptation();
if (event.config_->bitrate_bps)
audio_network_adaptation->set_bitrate_bps(*event.config_->bitrate_bps);
if (event.config_->frame_length_ms)
audio_network_adaptation->set_frame_length_ms(
*event.config_->frame_length_ms);
if (event.config_->uplink_packet_loss_fraction) {
audio_network_adaptation->set_uplink_packet_loss_fraction(
*event.config_->uplink_packet_loss_fraction);
}
if (event.config_->enable_fec)
audio_network_adaptation->set_enable_fec(*event.config_->enable_fec);
if (event.config_->enable_dtx)
audio_network_adaptation->set_enable_dtx(*event.config_->enable_dtx);
if (event.config_->num_channels)
audio_network_adaptation->set_num_channels(*event.config_->num_channels);
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeAudioPlayout(
const RtcEventAudioPlayout& event) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(event.timestamp_us_);
rtclog_event.set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT);
auto playout_event = rtclog_event.mutable_audio_playout_event();
playout_event->set_local_ssrc(event.ssrc_);
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeAudioReceiveStreamConfig(
const RtcEventAudioReceiveStreamConfig& event) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(event.timestamp_us_);
rtclog_event.set_type(rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT);
rtclog::AudioReceiveConfig* receiver_config =
rtclog_event.mutable_audio_receiver_config();
receiver_config->set_remote_ssrc(event.config_->remote_ssrc);
receiver_config->set_local_ssrc(event.config_->local_ssrc);
for (const auto& e : event.config_->rtp_extensions) {
rtclog::RtpHeaderExtension* extension =
receiver_config->add_header_extensions();
extension->set_name(e.uri);
extension->set_id(e.id);
}
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeAudioSendStreamConfig(
const RtcEventAudioSendStreamConfig& event) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(event.timestamp_us_);
rtclog_event.set_type(rtclog::Event::AUDIO_SENDER_CONFIG_EVENT);
rtclog::AudioSendConfig* sender_config =
rtclog_event.mutable_audio_sender_config();
sender_config->set_ssrc(event.config_->local_ssrc);
for (const auto& e : event.config_->rtp_extensions) {
rtclog::RtpHeaderExtension* extension =
sender_config->add_header_extensions();
extension->set_name(e.uri);
extension->set_id(e.id);
}
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeBweUpdateDelayBased(
const RtcEventBweUpdateDelayBased& event) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(event.timestamp_us_);
rtclog_event.set_type(rtclog::Event::DELAY_BASED_BWE_UPDATE);
auto bwe_event = rtclog_event.mutable_delay_based_bwe_update();
bwe_event->set_bitrate_bps(event.bitrate_bps_);
bwe_event->set_detector_state(ConvertDetectorState(event.detector_state_));
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeBweUpdateLossBased(
const RtcEventBweUpdateLossBased& event) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(event.timestamp_us_);
rtclog_event.set_type(rtclog::Event::LOSS_BASED_BWE_UPDATE);
auto bwe_event = rtclog_event.mutable_loss_based_bwe_update();
bwe_event->set_bitrate_bps(event.bitrate_bps_);
bwe_event->set_fraction_loss(event.fraction_loss_);
bwe_event->set_total_packets(event.total_packets_);
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeLoggingStarted(
const RtcEventLoggingStarted& event) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(event.timestamp_us_);
rtclog_event.set_type(rtclog::Event::LOG_START);
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeLoggingStopped(
const RtcEventLoggingStopped& event) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(event.timestamp_us_);
rtclog_event.set_type(rtclog::Event::LOG_END);
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeProbeClusterCreated(
const RtcEventProbeClusterCreated& event) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(event.timestamp_us_);
rtclog_event.set_type(rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT);
auto probe_cluster = rtclog_event.mutable_probe_cluster();
probe_cluster->set_id(event.id_);
probe_cluster->set_bitrate_bps(event.bitrate_bps_);
probe_cluster->set_min_packets(event.min_probes_);
probe_cluster->set_min_bytes(event.min_bytes_);
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeProbeResultFailure(
const RtcEventProbeResultFailure& event) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(event.timestamp_us_);
rtclog_event.set_type(rtclog::Event::BWE_PROBE_RESULT_EVENT);
auto probe_result = rtclog_event.mutable_probe_result();
probe_result->set_id(event.id_);
probe_result->set_result(ConvertProbeResultType(event.failure_reason_));
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeProbeResultSuccess(
const RtcEventProbeResultSuccess& event) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(event.timestamp_us_);
rtclog_event.set_type(rtclog::Event::BWE_PROBE_RESULT_EVENT);
auto probe_result = rtclog_event.mutable_probe_result();
probe_result->set_id(event.id_);
probe_result->set_result(rtclog::BweProbeResult::SUCCESS);
probe_result->set_bitrate_bps(event.bitrate_bps_);
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeRtcpPacketIncoming(
const RtcEventRtcpPacketIncoming& event) {
return EncodeRtcpPacket(event.timestamp_us_, event.packet_, true);
}
std::string RtcEventLogEncoderLegacy::EncodeRtcpPacketOutgoing(
const RtcEventRtcpPacketOutgoing& event) {
return EncodeRtcpPacket(event.timestamp_us_, event.packet_, false);
}
std::string RtcEventLogEncoderLegacy::EncodeRtpPacketIncoming(
const RtcEventRtpPacketIncoming& event) {
return EncodeRtpPacket(event.timestamp_us_, event.header_,
event.packet_length_, PacedPacketInfo::kNotAProbe,
true);
}
std::string RtcEventLogEncoderLegacy::EncodeRtpPacketOutgoing(
const RtcEventRtpPacketOutgoing& event) {
return EncodeRtpPacket(event.timestamp_us_, event.header_,
event.packet_length_, event.probe_cluster_id_, false);
}
std::string RtcEventLogEncoderLegacy::EncodeVideoReceiveStreamConfig(
const RtcEventVideoReceiveStreamConfig& event) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(event.timestamp_us_);
rtclog_event.set_type(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
rtclog::VideoReceiveConfig* receiver_config =
rtclog_event.mutable_video_receiver_config();
receiver_config->set_remote_ssrc(event.config_->remote_ssrc);
receiver_config->set_local_ssrc(event.config_->local_ssrc);
// TODO(perkj): Add field for rsid.
receiver_config->set_rtcp_mode(ConvertRtcpMode(event.config_->rtcp_mode));
receiver_config->set_remb(event.config_->remb);
for (const auto& e : event.config_->rtp_extensions) {
rtclog::RtpHeaderExtension* extension =
receiver_config->add_header_extensions();
extension->set_name(e.uri);
extension->set_id(e.id);
}
for (const auto& d : event.config_->codecs) {
rtclog::DecoderConfig* decoder = receiver_config->add_decoders();
decoder->set_name(d.payload_name);
decoder->set_payload_type(d.payload_type);
if (d.rtx_payload_type != 0) {
rtclog::RtxMap* rtx = receiver_config->add_rtx_map();
rtx->set_payload_type(d.payload_type);
rtx->mutable_config()->set_rtx_ssrc(event.config_->rtx_ssrc);
rtx->mutable_config()->set_rtx_payload_type(d.rtx_payload_type);
}
}
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeVideoSendStreamConfig(
const RtcEventVideoSendStreamConfig& event) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(event.timestamp_us_);
rtclog_event.set_type(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
rtclog::VideoSendConfig* sender_config =
rtclog_event.mutable_video_sender_config();
// TODO(perkj): rtclog::VideoSendConfig should only contain one SSRC.
sender_config->add_ssrcs(event.config_->local_ssrc);
if (event.config_->rtx_ssrc != 0) {
sender_config->add_rtx_ssrcs(event.config_->rtx_ssrc);
}
for (const auto& e : event.config_->rtp_extensions) {
rtclog::RtpHeaderExtension* extension =
sender_config->add_header_extensions();
extension->set_name(e.uri);
extension->set_id(e.id);
}
// TODO(perkj): rtclog::VideoSendConfig should contain many possible codec
// configurations.
for (const auto& codec : event.config_->codecs) {
sender_config->set_rtx_payload_type(codec.rtx_payload_type);
rtclog::EncoderConfig* encoder = sender_config->mutable_encoder();
encoder->set_name(codec.payload_name);
encoder->set_payload_type(codec.payload_type);
if (event.config_->codecs.size() > 1) {
LOG(WARNING) << "LogVideoSendStreamConfig currently only supports one "
<< "codec. Logging codec :" << codec.payload_name;
break;
}
}
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeRtcpPacket(
int64_t timestamp_us,
const rtc::Buffer& packet,
bool is_incoming) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(timestamp_us);
rtclog_event.set_type(rtclog::Event::RTCP_EVENT);
rtclog_event.mutable_rtcp_packet()->set_incoming(is_incoming);
rtcp::CommonHeader header;
const uint8_t* block_begin = packet.data();
const uint8_t* packet_end = packet.data() + packet.size();
RTC_DCHECK(packet.size() <= IP_PACKET_SIZE);
uint8_t buffer[IP_PACKET_SIZE];
uint32_t buffer_length = 0;
while (block_begin < packet_end) {
if (!header.Parse(block_begin, packet_end - block_begin)) {
break; // Incorrect message header.
}
const uint8_t* next_block = header.NextPacket();
uint32_t block_size = next_block - block_begin;
switch (header.type()) {
case rtcp::Bye::kPacketType:
case rtcp::ExtendedJitterReport::kPacketType:
case rtcp::ExtendedReports::kPacketType:
case rtcp::Psfb::kPacketType:
case rtcp::ReceiverReport::kPacketType:
case rtcp::Rtpfb::kPacketType:
case rtcp::SenderReport::kPacketType:
// We log sender reports, receiver reports, bye messages
// inter-arrival jitter, third-party loss reports, payload-specific
// feedback and extended reports.
memcpy(buffer + buffer_length, block_begin, block_size);
buffer_length += block_size;
break;
case rtcp::App::kPacketType:
case rtcp::Sdes::kPacketType:
default:
// We don't log sender descriptions, application defined messages
// or message blocks of unknown type.
break;
}
block_begin += block_size;
}
rtclog_event.mutable_rtcp_packet()->set_packet_data(buffer, buffer_length);
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::EncodeRtpPacket(
int64_t timestamp_us,
const webrtc::RtpPacket& header,
size_t packet_length,
int probe_cluster_id,
bool is_incoming) {
rtclog::Event rtclog_event;
rtclog_event.set_timestamp_us(timestamp_us);
rtclog_event.set_type(rtclog::Event::RTP_EVENT);
rtclog_event.mutable_rtp_packet()->set_incoming(is_incoming);
rtclog_event.mutable_rtp_packet()->set_packet_length(packet_length);
rtclog_event.mutable_rtp_packet()->set_header(header.data(), header.size());
if (probe_cluster_id != PacedPacketInfo::kNotAProbe) {
RTC_DCHECK(!is_incoming);
rtclog_event.mutable_rtp_packet()->set_probe_cluster_id(probe_cluster_id);
}
return Serialize(&rtclog_event);
}
std::string RtcEventLogEncoderLegacy::Serialize(rtclog::Event* event) {
// Even though we're only serializing a single event during this call, what
// we intend to get is a list of events, with a tag and length preceding
// each actual event. To produce that, we serialize a list of a single event.
// If we later concatenate several results from this function, the result will
// be a proper concatenation of all those events.
rtclog::EventStream event_stream;
event_stream.add_stream();
// As a tweak, we swap the new event into the event-stream, write that to
// file, then swap back. This saves on some copying, while making sure that
// the caller wouldn't be surprised by Serialize() modifying the object.
rtclog::Event* output_event = event_stream.mutable_stream(0);
output_event->Swap(event);
std::string output_string = event_stream.SerializeAsString();
RTC_DCHECK(!output_string.empty());
// When the function returns, the original Event will be unchanged.
output_event->Swap(event);
return output_string;
}
} // namespace webrtc
#endif // ENABLE_RTC_EVENT_LOG