mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-14 22:30:40 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/logging/rtc_event_log/rtc_event_log_parser_new.cc
Minyue Li 27e2b7d177 Plot NetEq stats in RTC event log visualizer. 
Bug: webrtc:9147
Change-Id: I61ec7bc5299201e25e1efc503b73b84d5be3ebbf
Reviewed-on: https://webrtc-review.googlesource.com/71740
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23151}
2018-05-07 17:01:48 +00:00

1251 lines
48 KiB
C++
Raw Blame History

/*
* Copyright (c) 2016 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/rtc_event_log_parser_new.h"
#include <stdint.h>
#include <string.h>
#include <algorithm>
#include <fstream>
#include <istream> // no-presubmit-check TODO(webrtc:8982)
#include <limits>
#include <map>
#include <utility>
#include "api/rtp_headers.h"
#include "api/rtpparameters.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/byte_io.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/logging.h"
#include "rtc_base/protobuf_utils.h"
#include "rtc_base/ptr_util.h"
namespace webrtc {
namespace {
RtcpMode GetRuntimeRtcpMode(rtclog::VideoReceiveConfig::RtcpMode rtcp_mode) {
switch (rtcp_mode) {
case rtclog::VideoReceiveConfig::RTCP_COMPOUND:
return RtcpMode::kCompound;
case rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE:
return RtcpMode::kReducedSize;
}
RTC_NOTREACHED();
return RtcpMode::kOff;
}
ParsedRtcEventLogNew::EventType GetRuntimeEventType(
rtclog::Event::EventType event_type) {
switch (event_type) {
case rtclog::Event::UNKNOWN_EVENT:
return ParsedRtcEventLogNew::EventType::UNKNOWN_EVENT;
case rtclog::Event::LOG_START:
return ParsedRtcEventLogNew::EventType::LOG_START;
case rtclog::Event::LOG_END:
return ParsedRtcEventLogNew::EventType::LOG_END;
case rtclog::Event::RTP_EVENT:
return ParsedRtcEventLogNew::EventType::RTP_EVENT;
case rtclog::Event::RTCP_EVENT:
return ParsedRtcEventLogNew::EventType::RTCP_EVENT;
case rtclog::Event::AUDIO_PLAYOUT_EVENT:
return ParsedRtcEventLogNew::EventType::AUDIO_PLAYOUT_EVENT;
case rtclog::Event::LOSS_BASED_BWE_UPDATE:
return ParsedRtcEventLogNew::EventType::LOSS_BASED_BWE_UPDATE;
case rtclog::Event::DELAY_BASED_BWE_UPDATE:
return ParsedRtcEventLogNew::EventType::DELAY_BASED_BWE_UPDATE;
case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT:
return ParsedRtcEventLogNew::EventType::VIDEO_RECEIVER_CONFIG_EVENT;
case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT:
return ParsedRtcEventLogNew::EventType::VIDEO_SENDER_CONFIG_EVENT;
case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT:
return ParsedRtcEventLogNew::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
return ParsedRtcEventLogNew::EventType::AUDIO_SENDER_CONFIG_EVENT;
case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT:
return ParsedRtcEventLogNew::EventType::AUDIO_NETWORK_ADAPTATION_EVENT;
case rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT:
return ParsedRtcEventLogNew::EventType::BWE_PROBE_CLUSTER_CREATED_EVENT;
case rtclog::Event::BWE_PROBE_RESULT_EVENT:
return ParsedRtcEventLogNew::EventType::BWE_PROBE_RESULT_EVENT;
case rtclog::Event::ALR_STATE_EVENT:
return ParsedRtcEventLogNew::EventType::ALR_STATE_EVENT;
case rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG:
return ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_CONFIG;
case rtclog::Event::ICE_CANDIDATE_PAIR_EVENT:
return ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_EVENT;
}
return ParsedRtcEventLogNew::EventType::UNKNOWN_EVENT;
}
BandwidthUsage GetRuntimeDetectorState(
rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
switch (detector_state) {
case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
return BandwidthUsage::kBwNormal;
case rtclog::DelayBasedBweUpdate::BWE_UNDERUSING:
return BandwidthUsage::kBwUnderusing;
case rtclog::DelayBasedBweUpdate::BWE_OVERUSING:
return BandwidthUsage::kBwOverusing;
}
RTC_NOTREACHED();
return BandwidthUsage::kBwNormal;
}
IceCandidatePairEventType GetRuntimeIceCandidatePairConfigType(
rtclog::IceCandidatePairConfig::IceCandidatePairConfigType type) {
switch (type) {
case rtclog::IceCandidatePairConfig::ADDED:
return IceCandidatePairEventType::kAdded;
case rtclog::IceCandidatePairConfig::UPDATED:
return IceCandidatePairEventType::kUpdated;
case rtclog::IceCandidatePairConfig::DESTROYED:
return IceCandidatePairEventType::kDestroyed;
case rtclog::IceCandidatePairConfig::SELECTED:
return IceCandidatePairEventType::kSelected;
}
RTC_NOTREACHED();
return IceCandidatePairEventType::kAdded;
}
IceCandidateType GetRuntimeIceCandidateType(
rtclog::IceCandidatePairConfig::IceCandidateType type) {
switch (type) {
case rtclog::IceCandidatePairConfig::LOCAL:
return IceCandidateType::kLocal;
case rtclog::IceCandidatePairConfig::STUN:
return IceCandidateType::kStun;
case rtclog::IceCandidatePairConfig::PRFLX:
return IceCandidateType::kPrflx;
case rtclog::IceCandidatePairConfig::RELAY:
return IceCandidateType::kRelay;
case rtclog::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE:
return IceCandidateType::kUnknown;
}
RTC_NOTREACHED();
return IceCandidateType::kUnknown;
}
IceCandidatePairProtocol GetRuntimeIceCandidatePairProtocol(
rtclog::IceCandidatePairConfig::Protocol protocol) {
switch (protocol) {
case rtclog::IceCandidatePairConfig::UDP:
return IceCandidatePairProtocol::kUdp;
case rtclog::IceCandidatePairConfig::TCP:
return IceCandidatePairProtocol::kTcp;
case rtclog::IceCandidatePairConfig::SSLTCP:
return IceCandidatePairProtocol::kSsltcp;
case rtclog::IceCandidatePairConfig::TLS:
return IceCandidatePairProtocol::kTls;
case rtclog::IceCandidatePairConfig::UNKNOWN_PROTOCOL:
return IceCandidatePairProtocol::kUnknown;
}
RTC_NOTREACHED();
return IceCandidatePairProtocol::kUnknown;
}
IceCandidatePairAddressFamily GetRuntimeIceCandidatePairAddressFamily(
rtclog::IceCandidatePairConfig::AddressFamily address_family) {
switch (address_family) {
case rtclog::IceCandidatePairConfig::IPV4:
return IceCandidatePairAddressFamily::kIpv4;
case rtclog::IceCandidatePairConfig::IPV6:
return IceCandidatePairAddressFamily::kIpv6;
case rtclog::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY:
return IceCandidatePairAddressFamily::kUnknown;
}
RTC_NOTREACHED();
return IceCandidatePairAddressFamily::kUnknown;
}
IceCandidateNetworkType GetRuntimeIceCandidateNetworkType(
rtclog::IceCandidatePairConfig::NetworkType network_type) {
switch (network_type) {
case rtclog::IceCandidatePairConfig::ETHERNET:
return IceCandidateNetworkType::kEthernet;
case rtclog::IceCandidatePairConfig::LOOPBACK:
return IceCandidateNetworkType::kLoopback;
case rtclog::IceCandidatePairConfig::WIFI:
return IceCandidateNetworkType::kWifi;
case rtclog::IceCandidatePairConfig::VPN:
return IceCandidateNetworkType::kVpn;
case rtclog::IceCandidatePairConfig::CELLULAR:
return IceCandidateNetworkType::kCellular;
case rtclog::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE:
return IceCandidateNetworkType::kUnknown;
}
RTC_NOTREACHED();
return IceCandidateNetworkType::kUnknown;
}
IceCandidatePairEventType GetRuntimeIceCandidatePairEventType(
rtclog::IceCandidatePairEvent::IceCandidatePairEventType type) {
switch (type) {
case rtclog::IceCandidatePairEvent::CHECK_SENT:
return IceCandidatePairEventType::kCheckSent;
case rtclog::IceCandidatePairEvent::CHECK_RECEIVED:
return IceCandidatePairEventType::kCheckReceived;
case rtclog::IceCandidatePairEvent::CHECK_RESPONSE_SENT:
return IceCandidatePairEventType::kCheckResponseSent;
case rtclog::IceCandidatePairEvent::CHECK_RESPONSE_RECEIVED:
return IceCandidatePairEventType::kCheckResponseReceived;
}
RTC_NOTREACHED();
return IceCandidatePairEventType::kCheckSent;
}
// 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<AudioLevel>(webrtc::RtpExtension::kAudioLevelDefaultId);
default_map.Register<TransmissionOffset>(
webrtc::RtpExtension::kTimestampOffsetDefaultId);
default_map.Register<AbsoluteSendTime>(
webrtc::RtpExtension::kAbsSendTimeDefaultId);
default_map.Register<VideoOrientation>(
webrtc::RtpExtension::kVideoRotationDefaultId);
default_map.Register<VideoContentTypeExtension>(
webrtc::RtpExtension::kVideoContentTypeDefaultId);
default_map.Register<VideoTimingExtension>(
webrtc::RtpExtension::kVideoTimingDefaultId);
default_map.Register<TransportSequenceNumber>(
webrtc::RtpExtension::kTransportSequenceNumberDefaultId);
default_map.Register<PlayoutDelayLimits>(
webrtc::RtpExtension::kPlayoutDelayDefaultId);
return default_map;
}
std::pair<uint64_t, bool> ParseVarInt(
std::istream& stream) { // no-presubmit-check TODO(webrtc:8982)
uint64_t varint = 0;
for (size_t bytes_read = 0; bytes_read < 10; ++bytes_read) {
// The most significant bit of each byte is 0 if it is the last byte in
// the varint and 1 otherwise. Thus, we take the 7 least significant bits
// of each byte and shift them 7 bits for each byte read previously to get
// the (unsigned) integer.
int byte = stream.get();
if (stream.eof()) {
return std::make_pair(varint, false);
}
RTC_DCHECK_GE(byte, 0);
RTC_DCHECK_LE(byte, 255);
varint |= static_cast<uint64_t>(byte & 0x7F) << (7 * bytes_read);
if ((byte & 0x80) == 0) {
return std::make_pair(varint, true);
}
}
return std::make_pair(varint, false);
}
void GetHeaderExtensions(std::vector<RtpExtension>* header_extensions,
const RepeatedPtrField<rtclog::RtpHeaderExtension>&
proto_header_extensions) {
header_extensions->clear();
for (auto& p : proto_header_extensions) {
RTC_CHECK(p.has_name());
RTC_CHECK(p.has_id());
const std::string& name = p.name();
int id = p.id();
header_extensions->push_back(RtpExtension(name, id));
}
}
} // namespace
ParsedRtcEventLogNew::ParsedRtcEventLogNew(
UnconfiguredHeaderExtensions parse_unconfigured_header_extensions)
: parse_unconfigured_header_extensions_(
parse_unconfigured_header_extensions) {
Clear();
}
void ParsedRtcEventLogNew::Clear() {
events_.clear();
default_extension_map_ = GetDefaultHeaderExtensionMap();
incoming_rtx_ssrcs_.clear();
incoming_video_ssrcs_.clear();
incoming_audio_ssrcs_.clear();
outgoing_rtx_ssrcs_.clear();
outgoing_video_ssrcs_.clear();
outgoing_audio_ssrcs_.clear();
incoming_rtp_packets_map_.clear();
outgoing_rtp_packets_map_.clear();
incoming_rtp_packets_by_ssrc_.clear();
outgoing_rtp_packets_by_ssrc_.clear();
incoming_rtp_packet_views_by_ssrc_.clear();
outgoing_rtp_packet_views_by_ssrc_.clear();
incoming_rtcp_packets_.clear();
outgoing_rtcp_packets_.clear();
incoming_rr_.clear();
outgoing_rr_.clear();
incoming_sr_.clear();
outgoing_sr_.clear();
incoming_nack_.clear();
outgoing_nack_.clear();
incoming_remb_.clear();
outgoing_remb_.clear();
incoming_transport_feedback_.clear();
outgoing_transport_feedback_.clear();
start_log_events_.clear();
stop_log_events_.clear();
audio_playout_events_.clear();
audio_network_adaptation_events_.clear();
bwe_probe_cluster_created_events_.clear();
bwe_probe_result_events_.clear();
bwe_delay_updates_.clear();
bwe_loss_updates_.clear();
alr_state_events_.clear();
ice_candidate_pair_configs_.clear();
ice_candidate_pair_events_.clear();
audio_recv_configs_.clear();
audio_send_configs_.clear();
video_recv_configs_.clear();
video_send_configs_.clear();
memset(last_incoming_rtcp_packet_, 0, IP_PACKET_SIZE);
last_incoming_rtcp_packet_length_ = 0;
first_timestamp_ = std::numeric_limits<int64_t>::max();
last_timestamp_ = std::numeric_limits<int64_t>::min();
incoming_rtp_extensions_maps_.clear();
outgoing_rtp_extensions_maps_.clear();
}
bool ParsedRtcEventLogNew::ParseFile(const std::string& filename) {
std::ifstream file( // no-presubmit-check TODO(webrtc:8982)
filename, std::ios_base::in | std::ios_base::binary);
if (!file.good() || !file.is_open()) {
RTC_LOG(LS_WARNING) << "Could not open file for reading.";
return false;
}
return ParseStream(file);
}
bool ParsedRtcEventLogNew::ParseString(const std::string& s) {
std::istringstream stream( // no-presubmit-check TODO(webrtc:8982)
s, std::ios_base::in | std::ios_base::binary);
return ParseStream(stream);
}
bool ParsedRtcEventLogNew::ParseStream(
std::istream& stream) { // no-presubmit-check TODO(webrtc:8982)
Clear();
const size_t kMaxEventSize = (1u << 16) - 1;
std::vector<char> tmp_buffer(kMaxEventSize);
uint64_t tag;
uint64_t message_length;
bool success;
RTC_DCHECK(stream.good());
while (1) {
// Check whether we have reached end of file.
stream.peek();
if (stream.eof()) {
break;
}
// Read the next message tag. The tag number is defined as
// (fieldnumber << 3) | wire_type. In our case, the field number is
// supposed to be 1 and the wire type for an
// length-delimited field is 2.
const uint64_t kExpectedTag = (1 << 3) | 2;
std::tie(tag, success) = ParseVarInt(stream);
if (!success) {
RTC_LOG(LS_WARNING)
<< "Missing field tag from beginning of protobuf event.";
return false;
} else if (tag != kExpectedTag) {
RTC_LOG(LS_WARNING)
<< "Unexpected field tag at beginning of protobuf event.";
return false;
}
// Read the length field.
std::tie(message_length, success) = ParseVarInt(stream);
if (!success) {
RTC_LOG(LS_WARNING) << "Missing message length after protobuf field tag.";
return false;
} else if (message_length > kMaxEventSize) {
RTC_LOG(LS_WARNING) << "Protobuf message length is too large.";
return false;
}
// Read the next protobuf event to a temporary char buffer.
stream.read(tmp_buffer.data(), message_length);
if (stream.gcount() != static_cast<int>(message_length)) {
RTC_LOG(LS_WARNING) << "Failed to read protobuf message from file.";
return false;
}
// Parse the protobuf event from the buffer.
rtclog::Event event;
if (!event.ParseFromArray(tmp_buffer.data(), message_length)) {
RTC_LOG(LS_WARNING) << "Failed to parse protobuf message.";
return false;
}
StoreParsedEvent(event);
events_.push_back(event);
}
// Move packets_streams from map to vector.
incoming_rtp_packets_by_ssrc_.reserve(incoming_rtp_packets_map_.size());
for (const auto& kv : incoming_rtp_packets_map_) {
incoming_rtp_packets_by_ssrc_.emplace_back(LoggedRtpStreamIncoming());
incoming_rtp_packets_by_ssrc_.back().ssrc = kv.first;
incoming_rtp_packets_by_ssrc_.back().incoming_packets =
std::move(kv.second);
}
outgoing_rtp_packets_by_ssrc_.reserve(outgoing_rtp_packets_map_.size());
for (const auto& kv : outgoing_rtp_packets_map_) {
outgoing_rtp_packets_by_ssrc_.emplace_back(LoggedRtpStreamOutgoing());
outgoing_rtp_packets_by_ssrc_.back().ssrc = kv.first;
outgoing_rtp_packets_by_ssrc_.back().outgoing_packets =
std::move(kv.second);
}
// Build PacketViews for easier iteration over RTP packets
for (const auto& stream : incoming_rtp_packets_by_ssrc_) {
incoming_rtp_packet_views_by_ssrc_.emplace_back(
LoggedRtpStreamView(stream.ssrc, stream.incoming_packets.data(),
stream.incoming_packets.size()));
}
for (const auto& stream : outgoing_rtp_packets_by_ssrc_) {
outgoing_rtp_packet_views_by_ssrc_.emplace_back(
LoggedRtpStreamView(stream.ssrc, stream.outgoing_packets.data(),
stream.outgoing_packets.size()));
}
return true;
}
void ParsedRtcEventLogNew::StoreParsedEvent(const rtclog::Event& event) {
if (event.type() != rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT &&
event.type() != rtclog::Event::VIDEO_SENDER_CONFIG_EVENT &&
event.type() != rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT &&
event.type() != rtclog::Event::AUDIO_SENDER_CONFIG_EVENT &&
event.type() != rtclog::Event::LOG_START &&
event.type() != rtclog::Event::LOG_END) {
RTC_CHECK(event.has_timestamp_us());
int64_t timestamp = event.timestamp_us();
first_timestamp_ = std::min(first_timestamp_, timestamp);
last_timestamp_ = std::max(last_timestamp_, timestamp);
}
switch (event.type()) {
case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT: {
rtclog::StreamConfig config = GetVideoReceiveConfig(event);
video_recv_configs_.emplace_back(GetTimestamp(event), config);
incoming_rtp_extensions_maps_[config.remote_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
// TODO(terelius): I don't understand the reason for configuring header
// extensions for the local SSRC. I think it should be removed, but for
// now I want to preserve the previous functionality.
incoming_rtp_extensions_maps_[config.local_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
incoming_video_ssrcs_.insert(config.remote_ssrc);
incoming_video_ssrcs_.insert(config.rtx_ssrc);
incoming_rtx_ssrcs_.insert(config.rtx_ssrc);
break;
}
case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT: {
std::vector<rtclog::StreamConfig> configs = GetVideoSendConfig(event);
video_send_configs_.emplace_back(GetTimestamp(event), configs);
for (const auto& config : configs) {
outgoing_rtp_extensions_maps_[config.local_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
outgoing_rtp_extensions_maps_[config.rtx_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
outgoing_video_ssrcs_.insert(config.local_ssrc);
outgoing_video_ssrcs_.insert(config.rtx_ssrc);
outgoing_rtx_ssrcs_.insert(config.rtx_ssrc);
}
break;
}
case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT: {
rtclog::StreamConfig config = GetAudioReceiveConfig(event);
audio_recv_configs_.emplace_back(GetTimestamp(event), config);
incoming_rtp_extensions_maps_[config.remote_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
incoming_rtp_extensions_maps_[config.local_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
incoming_audio_ssrcs_.insert(config.remote_ssrc);
break;
}
case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT: {
rtclog::StreamConfig config = GetAudioSendConfig(event);
audio_send_configs_.emplace_back(GetTimestamp(event), config);
outgoing_rtp_extensions_maps_[config.local_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
outgoing_audio_ssrcs_.insert(config.local_ssrc);
break;
}
case rtclog::Event::RTP_EVENT: {
PacketDirection direction;
uint8_t header[IP_PACKET_SIZE];
size_t header_length;
size_t total_length;
const RtpHeaderExtensionMap* extension_map = GetRtpHeader(
event, &direction, header, &header_length, &total_length, nullptr);
RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
RTPHeader parsed_header;
if (extension_map != nullptr) {
rtp_parser.Parse(&parsed_header, extension_map);
} else {
// Use the default extension map.
// TODO(ivoc): Once configuration of audio streams is stored in the
// event log, this can be removed.
// Tracking bug: webrtc:6399
rtp_parser.Parse(&parsed_header, &default_extension_map_);
}
RTC_CHECK(event.has_timestamp_us());
uint64_t timestamp_us = event.timestamp_us();
if (direction == kIncomingPacket) {
incoming_rtp_packets_map_[parsed_header.ssrc].push_back(
LoggedRtpPacketIncoming(timestamp_us, parsed_header, header_length,
total_length));
} else {
outgoing_rtp_packets_map_[parsed_header.ssrc].push_back(
LoggedRtpPacketOutgoing(timestamp_us, parsed_header, header_length,
total_length));
}
break;
}
case rtclog::Event::RTCP_EVENT: {
PacketDirection direction;
uint8_t packet[IP_PACKET_SIZE];
size_t total_length;
GetRtcpPacket(event, &direction, packet, &total_length);
uint64_t timestamp_us = GetTimestamp(event);
RTC_CHECK_LE(total_length, IP_PACKET_SIZE);
if (direction == kIncomingPacket) {
// Currently incoming RTCP packets are logged twice, both for audio and
// video. Only act on one of them. Compare against the previous parsed
// incoming RTCP packet.
if (total_length == last_incoming_rtcp_packet_length_ &&
memcmp(last_incoming_rtcp_packet_, packet, total_length) == 0)
break;
incoming_rtcp_packets_.push_back(
LoggedRtcpPacketIncoming(timestamp_us, packet, total_length));
last_incoming_rtcp_packet_length_ = total_length;
memcpy(last_incoming_rtcp_packet_, packet, total_length);
} else {
outgoing_rtcp_packets_.push_back(
LoggedRtcpPacketOutgoing(timestamp_us, packet, total_length));
}
rtcp::CommonHeader header;
const uint8_t* packet_end = packet + total_length;
for (const uint8_t* block = packet; block < packet_end;
block = header.NextPacket()) {
RTC_CHECK(header.Parse(block, packet_end - block));
if (header.type() == rtcp::TransportFeedback::kPacketType &&
header.fmt() == rtcp::TransportFeedback::kFeedbackMessageType) {
if (direction == kIncomingPacket) {
incoming_transport_feedback_.emplace_back();
LoggedRtcpPacketTransportFeedback& parsed_block =
incoming_transport_feedback_.back();
parsed_block.timestamp_us = GetTimestamp(event);
if (!parsed_block.transport_feedback.Parse(header))
incoming_transport_feedback_.pop_back();
} else {
outgoing_transport_feedback_.emplace_back();
LoggedRtcpPacketTransportFeedback& parsed_block =
outgoing_transport_feedback_.back();
parsed_block.timestamp_us = GetTimestamp(event);
if (!parsed_block.transport_feedback.Parse(header))
outgoing_transport_feedback_.pop_back();
}
} else if (header.type() == rtcp::SenderReport::kPacketType) {
LoggedRtcpPacketSenderReport parsed_block;
parsed_block.timestamp_us = GetTimestamp(event);
if (parsed_block.sr.Parse(header)) {
if (direction == kIncomingPacket)
incoming_sr_.push_back(std::move(parsed_block));
else
outgoing_sr_.push_back(std::move(parsed_block));
}
} else if (header.type() == rtcp::ReceiverReport::kPacketType) {
LoggedRtcpPacketReceiverReport parsed_block;
parsed_block.timestamp_us = GetTimestamp(event);
if (parsed_block.rr.Parse(header)) {
if (direction == kIncomingPacket)
incoming_rr_.push_back(std::move(parsed_block));
else
outgoing_rr_.push_back(std::move(parsed_block));
}
} else if (header.type() == rtcp::Remb::kPacketType &&
header.fmt() == rtcp::Remb::kFeedbackMessageType) {
LoggedRtcpPacketRemb parsed_block;
parsed_block.timestamp_us = GetTimestamp(event);
if (parsed_block.remb.Parse(header)) {
if (direction == kIncomingPacket)
incoming_remb_.push_back(std::move(parsed_block));
else
outgoing_remb_.push_back(std::move(parsed_block));
}
} else if (header.type() == rtcp::Nack::kPacketType &&
header.fmt() == rtcp::Nack::kFeedbackMessageType) {
LoggedRtcpPacketNack parsed_block;
parsed_block.timestamp_us = GetTimestamp(event);
if (parsed_block.nack.Parse(header)) {
if (direction == kIncomingPacket)
incoming_nack_.push_back(std::move(parsed_block));
else
outgoing_nack_.push_back(std::move(parsed_block));
}
}
}
break;
}
case ParsedRtcEventLogNew::LOG_START: {
start_log_events_.push_back(LoggedStartEvent(GetTimestamp(event)));
break;
}
case ParsedRtcEventLogNew::LOG_END: {
stop_log_events_.push_back(LoggedStopEvent(GetTimestamp(event)));
break;
}
case ParsedRtcEventLogNew::AUDIO_PLAYOUT_EVENT: {
LoggedAudioPlayoutEvent playout_event = GetAudioPlayout(event);
audio_playout_events_[playout_event.ssrc].push_back(playout_event);
break;
}
case ParsedRtcEventLogNew::LOSS_BASED_BWE_UPDATE: {
bwe_loss_updates_.push_back(GetLossBasedBweUpdate(event));
break;
}
case ParsedRtcEventLogNew::DELAY_BASED_BWE_UPDATE: {
bwe_delay_updates_.push_back(GetDelayBasedBweUpdate(event));
break;
}
case ParsedRtcEventLogNew::AUDIO_NETWORK_ADAPTATION_EVENT: {
LoggedAudioNetworkAdaptationEvent ana_event =
GetAudioNetworkAdaptation(event);
audio_network_adaptation_events_.push_back(ana_event);
break;
}
case ParsedRtcEventLogNew::BWE_PROBE_CLUSTER_CREATED_EVENT: {
bwe_probe_cluster_created_events_.push_back(
GetBweProbeClusterCreated(event));
break;
}
case ParsedRtcEventLogNew::BWE_PROBE_RESULT_EVENT: {
bwe_probe_result_events_.push_back(GetBweProbeResult(event));
break;
}
case ParsedRtcEventLogNew::ALR_STATE_EVENT: {
alr_state_events_.push_back(GetAlrState(event));
break;
}
case ParsedRtcEventLogNew::ICE_CANDIDATE_PAIR_CONFIG: {
ice_candidate_pair_configs_.push_back(GetIceCandidatePairConfig(event));
break;
}
case ParsedRtcEventLogNew::ICE_CANDIDATE_PAIR_EVENT: {
ice_candidate_pair_events_.push_back(GetIceCandidatePairEvent(event));
break;
}
case ParsedRtcEventLogNew::UNKNOWN_EVENT: {
break;
}
}
}
size_t ParsedRtcEventLogNew::GetNumberOfEvents() const {
return events_.size();
}
int64_t ParsedRtcEventLogNew::GetTimestamp(size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
return GetTimestamp(event);
}
int64_t ParsedRtcEventLogNew::GetTimestamp(const rtclog::Event& event) const {
RTC_CHECK(event.has_timestamp_us());
return event.timestamp_us();
}
ParsedRtcEventLogNew::EventType ParsedRtcEventLogNew::GetEventType(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
RTC_CHECK(event.has_type());
return GetRuntimeEventType(event.type());
}
// The header must have space for at least IP_PACKET_SIZE bytes.
const webrtc::RtpHeaderExtensionMap* ParsedRtcEventLogNew::GetRtpHeader(
size_t index,
PacketDirection* incoming,
uint8_t* header,
size_t* header_length,
size_t* total_length,
int* probe_cluster_id) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
return GetRtpHeader(event, incoming, header, header_length, total_length,
probe_cluster_id);
}
const webrtc::RtpHeaderExtensionMap* ParsedRtcEventLogNew::GetRtpHeader(
const rtclog::Event& event,
PacketDirection* incoming,
uint8_t* header,
size_t* header_length,
size_t* total_length,
int* probe_cluster_id) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::RTP_EVENT);
RTC_CHECK(event.has_rtp_packet());
const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
// Get direction of packet.
RTC_CHECK(rtp_packet.has_incoming());
if (incoming != nullptr) {
*incoming = rtp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
}
// Get packet length.
RTC_CHECK(rtp_packet.has_packet_length());
if (total_length != nullptr) {
*total_length = rtp_packet.packet_length();
}
// Get header length.
RTC_CHECK(rtp_packet.has_header());
if (header_length != nullptr) {
*header_length = rtp_packet.header().size();
}
if (probe_cluster_id != nullptr) {
if (rtp_packet.has_probe_cluster_id()) {
*probe_cluster_id = rtp_packet.probe_cluster_id();
RTC_CHECK_NE(*probe_cluster_id, PacedPacketInfo::kNotAProbe);
} else {
*probe_cluster_id = PacedPacketInfo::kNotAProbe;
}
}
// Get header contents.
if (header != nullptr) {
const size_t kMinRtpHeaderSize = 12;
RTC_CHECK_GE(rtp_packet.header().size(), kMinRtpHeaderSize);
RTC_CHECK_LE(rtp_packet.header().size(),
static_cast<size_t>(IP_PACKET_SIZE));
memcpy(header, rtp_packet.header().data(), rtp_packet.header().size());
uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(header + 8);
auto& extensions_maps = rtp_packet.incoming()
? incoming_rtp_extensions_maps_
: outgoing_rtp_extensions_maps_;
auto it = extensions_maps.find(ssrc);
if (it != extensions_maps.end()) {
return &(it->second);
}
if (parse_unconfigured_header_extensions_ ==
UnconfiguredHeaderExtensions::kAttemptWebrtcDefaultConfig) {
RTC_LOG(LS_WARNING) << "Using default header extension map for SSRC "
<< ssrc;
extensions_maps.insert(std::make_pair(ssrc, default_extension_map_));
return &default_extension_map_;
}
}
return nullptr;
}
// The packet must have space for at least IP_PACKET_SIZE bytes.
void ParsedRtcEventLogNew::GetRtcpPacket(size_t index,
PacketDirection* incoming,
uint8_t* packet,
size_t* length) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
GetRtcpPacket(event, incoming, packet, length);
}
void ParsedRtcEventLogNew::GetRtcpPacket(const rtclog::Event& event,
PacketDirection* incoming,
uint8_t* packet,
size_t* length) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::RTCP_EVENT);
RTC_CHECK(event.has_rtcp_packet());
const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
// Get direction of packet.
RTC_CHECK(rtcp_packet.has_incoming());
if (incoming != nullptr) {
*incoming = rtcp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
}
// Get packet length.
RTC_CHECK(rtcp_packet.has_packet_data());
if (length != nullptr) {
*length = rtcp_packet.packet_data().size();
}
// Get packet contents.
if (packet != nullptr) {
RTC_CHECK_LE(rtcp_packet.packet_data().size(),
static_cast<unsigned>(IP_PACKET_SIZE));
memcpy(packet, rtcp_packet.packet_data().data(),
rtcp_packet.packet_data().size());
}
}
rtclog::StreamConfig ParsedRtcEventLogNew::GetVideoReceiveConfig(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
return GetVideoReceiveConfig(events_[index]);
}
rtclog::StreamConfig ParsedRtcEventLogNew::GetVideoReceiveConfig(
const rtclog::Event& event) const {
rtclog::StreamConfig config;
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
RTC_CHECK(event.has_video_receiver_config());
const rtclog::VideoReceiveConfig& receiver_config =
event.video_receiver_config();
// Get SSRCs.
RTC_CHECK(receiver_config.has_remote_ssrc());
config.remote_ssrc = receiver_config.remote_ssrc();
RTC_CHECK(receiver_config.has_local_ssrc());
config.local_ssrc = receiver_config.local_ssrc();
config.rtx_ssrc = 0;
// Get RTCP settings.
RTC_CHECK(receiver_config.has_rtcp_mode());
config.rtcp_mode = GetRuntimeRtcpMode(receiver_config.rtcp_mode());
RTC_CHECK(receiver_config.has_remb());
config.remb = receiver_config.remb();
// Get RTX map.
std::map<uint32_t, const rtclog::RtxConfig> rtx_map;
for (int i = 0; i < receiver_config.rtx_map_size(); i++) {
const rtclog::RtxMap& map = receiver_config.rtx_map(i);
RTC_CHECK(map.has_payload_type());
RTC_CHECK(map.has_config());
RTC_CHECK(map.config().has_rtx_ssrc());
RTC_CHECK(map.config().has_rtx_payload_type());
rtx_map.insert(std::make_pair(map.payload_type(), map.config()));
}
// Get header extensions.
GetHeaderExtensions(&config.rtp_extensions,
receiver_config.header_extensions());
// Get decoders.
config.codecs.clear();
for (int i = 0; i < receiver_config.decoders_size(); i++) {
RTC_CHECK(receiver_config.decoders(i).has_name());
RTC_CHECK(receiver_config.decoders(i).has_payload_type());
int rtx_payload_type = 0;
auto rtx_it = rtx_map.find(receiver_config.decoders(i).payload_type());
if (rtx_it != rtx_map.end()) {
rtx_payload_type = rtx_it->second.rtx_payload_type();
if (config.rtx_ssrc != 0 &&
config.rtx_ssrc != rtx_it->second.rtx_ssrc()) {
RTC_LOG(LS_WARNING)
<< "RtcEventLog protobuf contained different SSRCs for "
"different received RTX payload types. Will only use "
"rtx_ssrc = "
<< config.rtx_ssrc << ".";
} else {
config.rtx_ssrc = rtx_it->second.rtx_ssrc();
}
}
config.codecs.emplace_back(receiver_config.decoders(i).name(),
receiver_config.decoders(i).payload_type(),
rtx_payload_type);
}
return config;
}
std::vector<rtclog::StreamConfig> ParsedRtcEventLogNew::GetVideoSendConfig(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
return GetVideoSendConfig(events_[index]);
}
std::vector<rtclog::StreamConfig> ParsedRtcEventLogNew::GetVideoSendConfig(
const rtclog::Event& event) const {
std::vector<rtclog::StreamConfig> configs;
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
RTC_CHECK(event.has_video_sender_config());
const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
if (sender_config.rtx_ssrcs_size() > 0 &&
sender_config.ssrcs_size() != sender_config.rtx_ssrcs_size()) {
RTC_LOG(WARNING)
<< "VideoSendConfig is configured for RTX but the number of "
"SSRCs doesn't match the number of RTX SSRCs.";
}
configs.resize(sender_config.ssrcs_size());
for (int i = 0; i < sender_config.ssrcs_size(); i++) {
// Get SSRCs.
configs[i].local_ssrc = sender_config.ssrcs(i);
if (sender_config.rtx_ssrcs_size() > 0 &&
i < sender_config.rtx_ssrcs_size()) {
RTC_CHECK(sender_config.has_rtx_payload_type());
configs[i].rtx_ssrc = sender_config.rtx_ssrcs(i);
}
// Get header extensions.
GetHeaderExtensions(&configs[i].rtp_extensions,
sender_config.header_extensions());
// Get the codec.
RTC_CHECK(sender_config.has_encoder());
RTC_CHECK(sender_config.encoder().has_name());
RTC_CHECK(sender_config.encoder().has_payload_type());
configs[i].codecs.emplace_back(
sender_config.encoder().name(), sender_config.encoder().payload_type(),
sender_config.has_rtx_payload_type() ? sender_config.rtx_payload_type()
: 0);
}
return configs;
}
rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioReceiveConfig(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
return GetAudioReceiveConfig(events_[index]);
}
rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioReceiveConfig(
const rtclog::Event& event) const {
rtclog::StreamConfig config;
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT);
RTC_CHECK(event.has_audio_receiver_config());
const rtclog::AudioReceiveConfig& receiver_config =
event.audio_receiver_config();
// Get SSRCs.
RTC_CHECK(receiver_config.has_remote_ssrc());
config.remote_ssrc = receiver_config.remote_ssrc();
RTC_CHECK(receiver_config.has_local_ssrc());
config.local_ssrc = receiver_config.local_ssrc();
// Get header extensions.
GetHeaderExtensions(&config.rtp_extensions,
receiver_config.header_extensions());
return config;
}
rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioSendConfig(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
return GetAudioSendConfig(events_[index]);
}
rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioSendConfig(
const rtclog::Event& event) const {
rtclog::StreamConfig config;
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_SENDER_CONFIG_EVENT);
RTC_CHECK(event.has_audio_sender_config());
const rtclog::AudioSendConfig& sender_config = event.audio_sender_config();
// Get SSRCs.
RTC_CHECK(sender_config.has_ssrc());
config.local_ssrc = sender_config.ssrc();
// Get header extensions.
GetHeaderExtensions(&config.rtp_extensions,
sender_config.header_extensions());
return config;
}
LoggedAudioPlayoutEvent ParsedRtcEventLogNew::GetAudioPlayout(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
return GetAudioPlayout(event);
}
LoggedAudioPlayoutEvent ParsedRtcEventLogNew::GetAudioPlayout(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_PLAYOUT_EVENT);
RTC_CHECK(event.has_audio_playout_event());
const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
LoggedAudioPlayoutEvent res;
res.timestamp_us = GetTimestamp(event);
RTC_CHECK(playout_event.has_local_ssrc());
res.ssrc = playout_event.local_ssrc();
return res;
}
LoggedBweLossBasedUpdate ParsedRtcEventLogNew::GetLossBasedBweUpdate(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
return GetLossBasedBweUpdate(event);
}
LoggedBweLossBasedUpdate ParsedRtcEventLogNew::GetLossBasedBweUpdate(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::LOSS_BASED_BWE_UPDATE);
RTC_CHECK(event.has_loss_based_bwe_update());
const rtclog::LossBasedBweUpdate& loss_event = event.loss_based_bwe_update();
LoggedBweLossBasedUpdate bwe_update;
bwe_update.timestamp_us = GetTimestamp(event);
RTC_CHECK(loss_event.has_bitrate_bps());
bwe_update.bitrate_bps = loss_event.bitrate_bps();
RTC_CHECK(loss_event.has_fraction_loss());
bwe_update.fraction_lost = loss_event.fraction_loss();
RTC_CHECK(loss_event.has_total_packets());
bwe_update.expected_packets = loss_event.total_packets();
return bwe_update;
}
LoggedBweDelayBasedUpdate ParsedRtcEventLogNew::GetDelayBasedBweUpdate(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
return GetDelayBasedBweUpdate(event);
}
LoggedBweDelayBasedUpdate ParsedRtcEventLogNew::GetDelayBasedBweUpdate(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::DELAY_BASED_BWE_UPDATE);
RTC_CHECK(event.has_delay_based_bwe_update());
const rtclog::DelayBasedBweUpdate& delay_event =
event.delay_based_bwe_update();
LoggedBweDelayBasedUpdate res;
res.timestamp_us = GetTimestamp(event);
RTC_CHECK(delay_event.has_bitrate_bps());
res.bitrate_bps = delay_event.bitrate_bps();
RTC_CHECK(delay_event.has_detector_state());
res.detector_state = GetRuntimeDetectorState(delay_event.detector_state());
return res;
}
LoggedAudioNetworkAdaptationEvent
ParsedRtcEventLogNew::GetAudioNetworkAdaptation(size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
return GetAudioNetworkAdaptation(event);
}
LoggedAudioNetworkAdaptationEvent
ParsedRtcEventLogNew::GetAudioNetworkAdaptation(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
RTC_CHECK(event.has_audio_network_adaptation());
const rtclog::AudioNetworkAdaptation& ana_event =
event.audio_network_adaptation();
LoggedAudioNetworkAdaptationEvent res;
res.timestamp_us = GetTimestamp(event);
if (ana_event.has_bitrate_bps())
res.config.bitrate_bps = ana_event.bitrate_bps();
if (ana_event.has_enable_fec())
res.config.enable_fec = ana_event.enable_fec();
if (ana_event.has_enable_dtx())
res.config.enable_dtx = ana_event.enable_dtx();
if (ana_event.has_frame_length_ms())
res.config.frame_length_ms = ana_event.frame_length_ms();
if (ana_event.has_num_channels())
res.config.num_channels = ana_event.num_channels();
if (ana_event.has_uplink_packet_loss_fraction())
res.config.uplink_packet_loss_fraction =
ana_event.uplink_packet_loss_fraction();
return res;
}
LoggedBweProbeClusterCreatedEvent
ParsedRtcEventLogNew::GetBweProbeClusterCreated(size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
return GetBweProbeClusterCreated(event);
}
LoggedBweProbeClusterCreatedEvent
ParsedRtcEventLogNew::GetBweProbeClusterCreated(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT);
RTC_CHECK(event.has_probe_cluster());
const rtclog::BweProbeCluster& pcc_event = event.probe_cluster();
LoggedBweProbeClusterCreatedEvent res;
res.timestamp_us = GetTimestamp(event);
RTC_CHECK(pcc_event.has_id());
res.id = pcc_event.id();
RTC_CHECK(pcc_event.has_bitrate_bps());
res.bitrate_bps = pcc_event.bitrate_bps();
RTC_CHECK(pcc_event.has_min_packets());
res.min_packets = pcc_event.min_packets();
RTC_CHECK(pcc_event.has_min_bytes());
res.min_bytes = pcc_event.min_bytes();
return res;
}
LoggedBweProbeResultEvent ParsedRtcEventLogNew::GetBweProbeResult(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
return GetBweProbeResult(event);
}
LoggedBweProbeResultEvent ParsedRtcEventLogNew::GetBweProbeResult(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PROBE_RESULT_EVENT);
RTC_CHECK(event.has_probe_result());
const rtclog::BweProbeResult& pr_event = event.probe_result();
LoggedBweProbeResultEvent res;
res.timestamp_us = GetTimestamp(event);
RTC_CHECK(pr_event.has_id());
res.id = pr_event.id();
RTC_CHECK(pr_event.has_result());
if (pr_event.result() == rtclog::BweProbeResult::SUCCESS) {
RTC_CHECK(pr_event.has_bitrate_bps());
res.bitrate_bps = pr_event.bitrate_bps();
} else if (pr_event.result() ==
rtclog::BweProbeResult::INVALID_SEND_RECEIVE_INTERVAL) {
res.failure_reason = ProbeFailureReason::kInvalidSendReceiveInterval;
} else if (pr_event.result() ==
rtclog::BweProbeResult::INVALID_SEND_RECEIVE_RATIO) {
res.failure_reason = ProbeFailureReason::kInvalidSendReceiveRatio;
} else if (pr_event.result() == rtclog::BweProbeResult::TIMEOUT) {
res.failure_reason = ProbeFailureReason::kTimeout;
} else {
RTC_NOTREACHED();
}
return res;
}
LoggedAlrStateEvent ParsedRtcEventLogNew::GetAlrState(size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
return GetAlrState(event);
}
LoggedAlrStateEvent ParsedRtcEventLogNew::GetAlrState(
const rtclog::Event& event) const {
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::ALR_STATE_EVENT);
RTC_CHECK(event.has_alr_state());
const rtclog::AlrState& alr_event = event.alr_state();
LoggedAlrStateEvent res;
res.timestamp_us = GetTimestamp(event);
RTC_CHECK(alr_event.has_in_alr());
res.in_alr = alr_event.in_alr();
return res;
}
LoggedIceCandidatePairConfig ParsedRtcEventLogNew::GetIceCandidatePairConfig(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& rtc_event = events_[index];
return GetIceCandidatePairConfig(rtc_event);
}
LoggedIceCandidatePairConfig ParsedRtcEventLogNew::GetIceCandidatePairConfig(
const rtclog::Event& rtc_event) const {
RTC_CHECK(rtc_event.has_type());
RTC_CHECK_EQ(rtc_event.type(), rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG);
LoggedIceCandidatePairConfig res;
const rtclog::IceCandidatePairConfig& config =
rtc_event.ice_candidate_pair_config();
res.timestamp_us = GetTimestamp(rtc_event);
RTC_CHECK(config.has_config_type());
res.type = GetRuntimeIceCandidatePairConfigType(config.config_type());
RTC_CHECK(config.has_candidate_pair_id());
res.candidate_pair_id = config.candidate_pair_id();
RTC_CHECK(config.has_local_candidate_type());
res.local_candidate_type =
GetRuntimeIceCandidateType(config.local_candidate_type());
RTC_CHECK(config.has_local_relay_protocol());
res.local_relay_protocol =
GetRuntimeIceCandidatePairProtocol(config.local_relay_protocol());
RTC_CHECK(config.has_local_network_type());
res.local_network_type =
GetRuntimeIceCandidateNetworkType(config.local_network_type());
RTC_CHECK(config.has_local_address_family());
res.local_address_family =
GetRuntimeIceCandidatePairAddressFamily(config.local_address_family());
RTC_CHECK(config.has_remote_candidate_type());
res.remote_candidate_type =
GetRuntimeIceCandidateType(config.remote_candidate_type());
RTC_CHECK(config.has_remote_address_family());
res.remote_address_family =
GetRuntimeIceCandidatePairAddressFamily(config.remote_address_family());
RTC_CHECK(config.has_candidate_pair_protocol());
res.candidate_pair_protocol =
GetRuntimeIceCandidatePairProtocol(config.candidate_pair_protocol());
return res;
}
LoggedIceCandidatePairEvent ParsedRtcEventLogNew::GetIceCandidatePairEvent(
size_t index) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& rtc_event = events_[index];
return GetIceCandidatePairEvent(rtc_event);
}
LoggedIceCandidatePairEvent ParsedRtcEventLogNew::GetIceCandidatePairEvent(
const rtclog::Event& rtc_event) const {
RTC_CHECK(rtc_event.has_type());
RTC_CHECK_EQ(rtc_event.type(), rtclog::Event::ICE_CANDIDATE_PAIR_EVENT);
LoggedIceCandidatePairEvent res;
const rtclog::IceCandidatePairEvent& event =
rtc_event.ice_candidate_pair_event();
res.timestamp_us = GetTimestamp(rtc_event);
RTC_CHECK(event.has_event_type());
res.type = GetRuntimeIceCandidatePairEventType(event.event_type());
RTC_CHECK(event.has_candidate_pair_id());
res.candidate_pair_id = event.candidate_pair_id();
return res;
}
// Returns the MediaType for registered SSRCs. Search from the end to use last
// registered types first.
ParsedRtcEventLogNew::MediaType ParsedRtcEventLogNew::GetMediaType(
uint32_t ssrc,
PacketDirection direction) const {
if (direction == kIncomingPacket) {
if (std::find(incoming_video_ssrcs_.begin(), incoming_video_ssrcs_.end(),
ssrc) != incoming_video_ssrcs_.end()) {
return MediaType::VIDEO;
}
if (std::find(incoming_audio_ssrcs_.begin(), incoming_audio_ssrcs_.end(),
ssrc) != incoming_audio_ssrcs_.end()) {
return MediaType::AUDIO;
}
} else {
if (std::find(outgoing_video_ssrcs_.begin(), outgoing_video_ssrcs_.end(),
ssrc) != outgoing_video_ssrcs_.end()) {
return MediaType::VIDEO;
}
if (std::find(outgoing_audio_ssrcs_.begin(), outgoing_audio_ssrcs_.end(),
ssrc) != outgoing_audio_ssrcs_.end()) {
return MediaType::AUDIO;
}
}
return MediaType::ANY;
}
} // namespace webrtc
Reference in a new issue View git blame Copy permalink