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Reland "Adds richer packet and ice processing to ParsedRtcEventLog."

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This reverts commit 5586d7fb57.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Revert "Adds richer packet and ice processing to ParsedRtcEventLog."
> 
> This reverts commit 4306a25dfc.
> 
> Reason for revert: Breaks downstream project
> 
> Original change's description:
> > Adds richer packet and ice processing to ParsedRtcEventLog.
> > 
> > Bug: webrtc:10170
> > Change-Id: I0f10a8c0b5656917a806cf0f3ad88b7a6baee000
> > Reviewed-on: https://webrtc-review.googlesource.com/c/116069
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26268}
> 
> TBR=terelius@webrtc.org,srte@webrtc.org
> 
> Change-Id: Ic50fdfb6b10c26e77728b594f553bc4aac4eb0ab
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10170
> Reviewed-on: https://webrtc-review.googlesource.com/c/117780
> Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26270}

TBR=terelius@webrtc.org,srte@webrtc.org,amithi@webrtc.org

Change-Id: I5e87fb472b91dd4b6fa177418f03a9031035ec60
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10170
Reviewed-on: https://webrtc-review.googlesource.com/c/117721
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26274}
This commit is contained in:
Sebastian Jansson 2019-01-16 07:44:11 +00:00 committed by Commit Bot
parent 77536a2b81
commit 20aae1de5c
5 changed files with 351 additions and 83 deletions
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5
logging/BUILD.gn
View file

@ -291,6 +291,7 @@ if (rtc_enable_protobuf) {
rtc_static_library("rtc_event_log_parser") {
visibility = [ "*" ]
sources = [
"rtc_event_log/logged_events.cc",
"rtc_event_log/logged_events.h",
"rtc_event_log/rtc_event_log_parser.cc",
"rtc_event_log/rtc_event_log_parser.h",
@ -307,6 +308,9 @@ if (rtc_enable_protobuf) {
":rtc_event_log_proto",
":rtc_stream_config",
"../api:libjingle_peerconnection_api",
"../api/units:data_rate",
"../api/units:time_delta",
"../api/units:timestamp",
"../call:video_stream_api",
"../modules/audio_coding:audio_network_adaptor",
"../modules/congestion_controller/rtp:transport_feedback",
@ -317,6 +321,7 @@ if (rtc_enable_protobuf) {
"../rtc_base:deprecation",
"../rtc_base:protobuf_utils",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_numerics",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]

36
logging/rtc_event_log/logged_events.cc Normal file
View file

@ -0,0 +1,36 @@
/*
* Copyright 2019 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/logged_events.h"
namespace webrtc {
LoggedPacketInfo::LoggedPacketInfo(const LoggedRtpPacket& rtp,
LoggedMediaType media_type,
bool rtx,
Timestamp capture_time)
: ssrc(rtp.header.ssrc),
stream_seq_no(rtp.header.sequenceNumber),
size(static_cast<uint16_t>(rtp.total_length)),
payload_type(rtp.header.payloadType),
media_type(media_type),
rtx(rtx),
marker_bit(rtp.header.markerBit),
has_transport_seq_no(rtp.header.extension.hasTransportSequenceNumber),
transport_seq_no(static_cast<uint16_t>(
has_transport_seq_no ? rtp.header.extension.transportSequenceNumber
: 0)),
// TODO(srte): Use logged sample rate when it is added to the format.
capture_time(capture_time),
log_packet_time(Timestamp::us(rtp.log_time_us())) {}
LoggedPacketInfo::LoggedPacketInfo(const LoggedPacketInfo&) = default;
LoggedPacketInfo::~LoggedPacketInfo() {}
} // namespace webrtc

70
logging/rtc_event_log/logged_events.h
View file

@ -13,7 +13,11 @@
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "logging/rtc_event_log/events/rtc_event_dtls_transport_state.h"
#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
@ -437,5 +441,71 @@ struct LoggedVideoSendConfig {
int64_t timestamp_us;
rtclog::StreamConfig config;
};
struct LoggedRouteChangeEvent {
uint32_t route_id;
int64_t timestamp_us;
uint16_t send_overhead;
uint16_t return_overhead;
};
enum class LoggedMediaType : uint8_t { kUnknown, kAudio, kVideo };
struct LoggedPacketInfo {
LoggedPacketInfo(const LoggedRtpPacket& rtp,
LoggedMediaType media_type,
bool rtx,
Timestamp capture_time);
LoggedPacketInfo(const LoggedPacketInfo&);
~LoggedPacketInfo();
uint32_t ssrc;
uint16_t stream_seq_no;
uint16_t size;
uint16_t overhead = 0;
uint8_t payload_type;
LoggedMediaType media_type = LoggedMediaType::kUnknown;
bool rtx = false;
bool marker_bit = false;
bool has_transport_seq_no = false;
bool last_in_feedback = false;
uint16_t transport_seq_no = 0;
// The RTP header timestamp unwrapped and converted from tick count to seconds
// based timestamp.
Timestamp capture_time;
// The time the packet was logged. This is the receive time for incoming
// packets and send time for outgoing.
Timestamp log_packet_time;
// The receive time that was reported in feedback. For incoming packets this
// corresponds to log_packet_time, but might be measured using another clock.
// PlusInfinity indicates that the packet was lost.
Timestamp reported_recv_time = Timestamp::MinusInfinity();
// The time feedback message was logged. This is the feedback send time for
// incoming packets and feedback receive time for outgoing.
// PlusInfinity indicates that feedback was expected but not received.
Timestamp log_feedback_time = Timestamp::MinusInfinity();
// The delay betweeen receiving an RTP packet and sending feedback for
// incoming packets. For outgoing packets we don't know the feedback send
// time, and this is instead calculated as the difference in reported receive
// time between this packet and the last packet in the same feedback message.
TimeDelta feedback_hold_duration = TimeDelta::MinusInfinity();
};
enum class LoggedIceEventType {
kAdded,
kUpdated,
kDestroyed,
kSelected,
kCheckSent,
kCheckReceived,
kCheckResponseSent,
kCheckResponseReceived,
};
struct LoggedIceEvent {
uint32_t candidate_pair_id;
int64_t timestamp_us;
LoggedIceEventType event_type;
};
} // namespace webrtc
#endif // LOGGING_RTC_EVENT_LOG_LOGGED_EVENTS_H_

331
logging/rtc_event_log/rtc_event_log_parser.cc
View file

@ -28,6 +28,7 @@
#include "logging/rtc_event_log/encoder/delta_encoding.h"
#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_common.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_event_processor.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
@ -39,6 +40,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/sequence_number_util.h"
#include "rtc_base/protobuf_utils.h"
using webrtc_event_logging::ToSigned;
@ -47,6 +49,69 @@ using webrtc_event_logging::ToUnsigned;
namespace webrtc {
namespace {
constexpr size_t kIpv4Overhead = 20;
constexpr size_t kIpv6Overhead = 40;
constexpr size_t kUdpOverhead = 8;
constexpr size_t kSrtpOverhead = 10;
constexpr size_t kStunOverhead = 4;
constexpr uint16_t kDefaultOverhead =
kUdpOverhead + kSrtpOverhead + kIpv4Overhead;
// Starting at a multiple of common audio sample rate (48000) and video tick
// rate (90000) to make a tick count of 0 to correspond to something without
// decimals in base 10.
constexpr uint64_t kStartingCaptureTimeTicks = 90 * 48 * 1000;
struct MediaStreamInfo {
MediaStreamInfo() : unwrap_capture_ticks(kStartingCaptureTimeTicks) {}
MediaStreamInfo(LoggedMediaType media_type, bool rtx)
: media_type(media_type),
rtx(rtx),
unwrap_capture_ticks(kStartingCaptureTimeTicks) {}
LoggedMediaType media_type = LoggedMediaType::kUnknown;
bool rtx = false;
SeqNumUnwrapper<uint32_t> unwrap_capture_ticks;
};
template <typename Iterable>
void AddRecvStreamInfos(std::map<uint32_t, MediaStreamInfo>* streams,
const Iterable configs,
LoggedMediaType media_type) {
for (auto& conf : configs) {
streams->insert({conf.config.remote_ssrc, {media_type, false}});
if (conf.config.rtx_ssrc != 0)
streams->insert({conf.config.rtx_ssrc, {media_type, true}});
}
}
template <typename Iterable>
void AddSendStreamInfos(std::map<uint32_t, MediaStreamInfo>* streams,
const Iterable configs,
LoggedMediaType media_type) {
for (auto& conf : configs) {
streams->insert({conf.config.local_ssrc, {media_type, false}});
if (conf.config.rtx_ssrc != 0)
streams->insert({conf.config.rtx_ssrc, {media_type, true}});
}
}
struct OverheadChangeEvent {
int64_t timestamp_us;
uint16_t overhead;
};
std::vector<OverheadChangeEvent> GetOverheadChangingEvents(
const std::vector<LoggedRouteChangeEvent>& route_changes,
PacketDirection direction) {
std::vector<OverheadChangeEvent> overheads;
for (auto& event : route_changes) {
uint16_t new_overhead = direction == PacketDirection::kIncomingPacket
? event.return_overhead
: event.send_overhead;
if (overheads.empty() || new_overhead != overheads.back().overhead) {
overheads.push_back({event.timestamp_us, new_overhead});
}
}
return overheads;
}
// Conversion functions for legacy wire format.
RtcpMode GetRuntimeRtcpMode(rtclog::VideoReceiveConfig::RtcpMode rtcp_mode) {
switch (rtcp_mode) {
@ -407,23 +472,6 @@ void GetHeaderExtensions(std::vector<RtpExtension>* header_extensions,
}
}
void SortPacketFeedbackVectorWithLoss(std::vector<PacketFeedback>* vec) {
class LossHandlingPacketFeedbackComparator {
public:
inline bool operator()(const PacketFeedback& lhs,
const PacketFeedback& rhs) {
if (lhs.arrival_time_ms != PacketFeedback::kNotReceived &&
rhs.arrival_time_ms != PacketFeedback::kNotReceived &&
lhs.arrival_time_ms != rhs.arrival_time_ms)
return lhs.arrival_time_ms < rhs.arrival_time_ms;
if (lhs.send_time_ms != rhs.send_time_ms)
return lhs.send_time_ms < rhs.send_time_ms;
return lhs.sequence_number < rhs.sequence_number;
}
};
std::sort(vec->begin(), vec->end(), LossHandlingPacketFeedbackComparator());
}
template <typename ProtoType, typename LoggedType>
void StoreRtpPackets(
const ProtoType& proto,
@ -1766,84 +1814,187 @@ ParsedRtcEventLog::MediaType ParsedRtcEventLog::GetMediaType(
return MediaType::ANY;
}
std::vector<LoggedRouteChangeEvent> ParsedRtcEventLog::GetRouteChanges() const {
std::vector<LoggedRouteChangeEvent> route_changes;
for (auto& candidate : ice_candidate_pair_configs()) {
if (candidate.type == IceCandidatePairConfigType::kSelected) {
LoggedRouteChangeEvent route;
route.route_id = candidate.candidate_pair_id;
route.timestamp_us = candidate.log_time_us();
route.send_overhead = kUdpOverhead + kSrtpOverhead + kIpv4Overhead;
if (candidate.remote_address_family ==
IceCandidatePairAddressFamily::kIpv6)
route.send_overhead += kIpv6Overhead - kIpv4Overhead;
if (candidate.remote_candidate_type != IceCandidateType::kLocal)
route.send_overhead += kStunOverhead;
route.return_overhead = kUdpOverhead + kSrtpOverhead + kIpv4Overhead;
if (candidate.remote_address_family ==
IceCandidatePairAddressFamily::kIpv6)
route.return_overhead += kIpv6Overhead - kIpv4Overhead;
if (candidate.remote_candidate_type != IceCandidateType::kLocal)
route.return_overhead += kStunOverhead;
route_changes.push_back(route);
}
}
return route_changes;
}
std::vector<LoggedPacketInfo> ParsedRtcEventLog::GetPacketInfos(
PacketDirection direction) const {
std::map<uint32_t, MediaStreamInfo> streams;
if (direction == PacketDirection::kIncomingPacket) {
AddRecvStreamInfos(&streams, audio_recv_configs(), LoggedMediaType::kAudio);
AddRecvStreamInfos(&streams, video_recv_configs(), LoggedMediaType::kVideo);
} else if (direction == PacketDirection::kOutgoingPacket) {
AddSendStreamInfos(&streams, audio_send_configs(), LoggedMediaType::kAudio);
AddSendStreamInfos(&streams, video_send_configs(), LoggedMediaType::kVideo);
}
// Using one second as an arbitrary starting point.
SimulatedClock clock(1000000);
TransportFeedbackAdapter feedback_adapter(&clock);
std::vector<OverheadChangeEvent> overheads =
GetOverheadChangingEvents(GetRouteChanges(), direction);
auto overhead_iter = overheads.begin();
std::vector<LoggedPacketInfo> packets;
std::map<int64_t, size_t> indices;
uint16_t current_overhead = kDefaultOverhead;
int64_t last_log_time_ms = 0;
auto advance_clock = [&](int64_t log_time_ms) {
if (overhead_iter != overheads.end() &&
log_time_ms * 1000 >= overhead_iter->timestamp_us) {
current_overhead = overhead_iter->overhead;
++overhead_iter;
}
RTC_CHECK_GE(log_time_ms, last_log_time_ms);
clock.AdvanceTimeMilliseconds(log_time_ms - last_log_time_ms);
last_log_time_ms = log_time_ms;
};
auto rtp_handler = [&](const LoggedRtpPacket& rtp) {
advance_clock(rtp.log_time_ms());
MediaStreamInfo* stream = &streams[rtp.header.ssrc];
uint64_t capture_ticks =
stream->unwrap_capture_ticks.Unwrap(rtp.header.timestamp);
// TODO(srte): Use logged sample rate when it is added to the format.
Timestamp capture_time = Timestamp::seconds(
capture_ticks /
(stream->media_type == LoggedMediaType::kAudio ? 48000.0 : 90000.0));
LoggedPacketInfo logged(rtp, stream->media_type, stream->rtx, capture_time);
logged.overhead = current_overhead;
if (rtp.header.extension.hasTransportSequenceNumber) {
logged.log_feedback_time = Timestamp::PlusInfinity();
rtc::SentPacket sent_packet;
sent_packet.send_time_ms = rtp.log_time_ms();
sent_packet.info.packet_size_bytes = rtp.total_length;
sent_packet.info.included_in_feedback = true;
sent_packet.packet_id = rtp.header.extension.transportSequenceNumber;
feedback_adapter.AddPacket(rtp.header.ssrc, sent_packet.packet_id,
rtp.total_length, PacedPacketInfo());
auto sent_packet_msg = feedback_adapter.ProcessSentPacket(sent_packet);
RTC_CHECK(sent_packet_msg);
indices[sent_packet_msg->sequence_number] = packets.size();
}
packets.push_back(logged);
};
auto feedback_handler = [&](const LoggedRtcpPacketTransportFeedback& logged) {
advance_clock(logged.log_time_ms());
auto msg =
feedback_adapter.ProcessTransportFeedback(logged.transport_feedback);
if (!msg.has_value() || msg->packet_feedbacks.empty())
return;
auto& last_fb = msg->packet_feedbacks.back();
Timestamp last_recv_time = last_fb.receive_time;
for (auto& fb : msg->packet_feedbacks) {
if (indices.find(fb.sent_packet.sequence_number) == indices.end()) {
RTC_LOG(LS_ERROR) << "Received feedback for unknown packet: "
<< fb.sent_packet.sequence_number;
continue;
}
LoggedPacketInfo* sent =
&packets[indices[fb.sent_packet.sequence_number]];
sent->reported_recv_time = fb.receive_time;
sent->log_feedback_time = msg->feedback_time;
if (direction == PacketDirection::kOutgoingPacket) {
sent->feedback_hold_duration = last_recv_time - fb.receive_time;
} else {
sent->feedback_hold_duration =
Timestamp::us(logged.log_time_us()) - sent->log_packet_time;
}
sent->last_in_feedback = (&fb == &last_fb);
}
};
RtcEventProcessor process;
for (const auto& rtp_packets : rtp_packets_by_ssrc(direction)) {
process.AddEvents(rtp_packets.packet_view, rtp_handler);
}
if (direction == PacketDirection::kOutgoingPacket) {
process.AddEvents(incoming_transport_feedback_, feedback_handler);
} else {
process.AddEvents(outgoing_transport_feedback_, feedback_handler);
}
process.ProcessEventsInOrder();
return packets;
}
std::vector<LoggedIceCandidatePairConfig> ParsedRtcEventLog::GetIceCandidates()
const {
std::vector<LoggedIceCandidatePairConfig> candidates;
std::set<uint32_t> added;
for (auto& candidate : ice_candidate_pair_configs()) {
if (added.find(candidate.candidate_pair_id) == added.end()) {
candidates.push_back(candidate);
added.insert(candidate.candidate_pair_id);
}
}
return candidates;
}
std::vector<LoggedIceEvent> ParsedRtcEventLog::GetIceEvents() const {
using CheckType = IceCandidatePairEventType;
using ConfigType = IceCandidatePairConfigType;
using Combined = LoggedIceEventType;
std::map<CheckType, Combined> check_map(
{{CheckType::kCheckSent, Combined::kCheckSent},
{CheckType::kCheckReceived, Combined::kCheckReceived},
{CheckType::kCheckResponseSent, Combined::kCheckResponseSent},
{CheckType::kCheckResponseReceived, Combined::kCheckResponseReceived}});
std::map<ConfigType, Combined> config_map(
{{ConfigType::kAdded, Combined::kAdded},
{ConfigType::kUpdated, Combined::kUpdated},
{ConfigType::kDestroyed, Combined::kDestroyed},
{ConfigType::kSelected, Combined::kSelected}});
std::vector<LoggedIceEvent> logged_events;
auto handle_check = [&](const LoggedIceCandidatePairEvent& check) {
logged_events.push_back(LoggedIceEvent{
check.candidate_pair_id, check.timestamp_us, check_map[check.type]});
};
auto handle_config = [&](const LoggedIceCandidatePairConfig& conf) {
logged_events.push_back(LoggedIceEvent{
conf.candidate_pair_id, conf.timestamp_us, config_map[conf.type]});
};
RtcEventProcessor process;
process.AddEvents(ice_candidate_pair_events(), handle_check);
process.AddEvents(ice_candidate_pair_configs(), handle_config);
return logged_events;
}
const std::vector<MatchedSendArrivalTimes> GetNetworkTrace(
const ParsedRtcEventLog& parsed_log) {
using RtpPacketType = LoggedRtpPacketOutgoing;
using TransportFeedbackType = LoggedRtcpPacketTransportFeedback;
std::multimap<int64_t, const RtpPacketType*> outgoing_rtp;
for (const auto& stream : parsed_log.outgoing_rtp_packets_by_ssrc()) {
for (const RtpPacketType& rtp_packet : stream.outgoing_packets)
outgoing_rtp.insert(
std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
}
const std::vector<TransportFeedbackType>& incoming_rtcp =
parsed_log.transport_feedbacks(kIncomingPacket);
SimulatedClock clock(0);
TransportFeedbackAdapter feedback_adapter(&clock);
auto rtp_iterator = outgoing_rtp.begin();
auto rtcp_iterator = incoming_rtcp.begin();
auto NextRtpTime = [&]() {
if (rtp_iterator != outgoing_rtp.end())
return static_cast<int64_t>(rtp_iterator->first);
return std::numeric_limits<int64_t>::max();
};
auto NextRtcpTime = [&]() {
if (rtcp_iterator != incoming_rtcp.end())
return static_cast<int64_t>(rtcp_iterator->log_time_us());
return std::numeric_limits<int64_t>::max();
};
int64_t time_us = std::min(NextRtpTime(), NextRtcpTime());
std::vector<MatchedSendArrivalTimes> rtp_rtcp_matched;
while (time_us != std::numeric_limits<int64_t>::max()) {
clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
if (clock.TimeInMicroseconds() >= NextRtpTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
const RtpPacketType& rtp_packet = *rtp_iterator->second;
rtc::SentPacket sent_packet;
sent_packet.send_time_ms = rtp_packet.rtp.log_time_ms();
sent_packet.info.packet_size_bytes = rtp_packet.rtp.total_length;
if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
feedback_adapter.AddPacket(
rtp_packet.rtp.header.ssrc,
rtp_packet.rtp.header.extension.transportSequenceNumber,
rtp_packet.rtp.total_length, PacedPacketInfo());
sent_packet.packet_id =
rtp_packet.rtp.header.extension.transportSequenceNumber;
sent_packet.info.included_in_feedback = true;
sent_packet.info.included_in_allocation = true;
feedback_adapter.ProcessSentPacket(sent_packet);
} else {
sent_packet.info.included_in_feedback = false;
// TODO(srte): Make it possible to indicate that all packets are part of
// allocation.
sent_packet.info.included_in_allocation = false;
feedback_adapter.ProcessSentPacket(sent_packet);
}
++rtp_iterator;
}
if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
feedback_adapter.ProcessTransportFeedback(
rtcp_iterator->transport_feedback);
std::vector<PacketFeedback> feedback =
feedback_adapter.GetTransportFeedbackVector();
SortPacketFeedbackVectorWithLoss(&feedback);
for (const PacketFeedback& packet : feedback) {
for (auto& packet :
parsed_log.GetPacketInfos(PacketDirection::kOutgoingPacket)) {
if (packet.log_feedback_time.IsFinite() &&
packet.reported_recv_time.IsFinite()) {
rtp_rtcp_matched.emplace_back(
clock.TimeInMilliseconds(), packet.send_time_ms,
packet.arrival_time_ms, packet.payload_size);
packet.log_feedback_time.ms(), packet.log_packet_time.ms(),
packet.reported_recv_time.ms(), packet.size);
}
++rtcp_iterator;
}
time_us = std::min(NextRtpTime(), NextRtcpTime());
}
return rtp_rtcp_matched;
}

6
logging/rtc_event_log/rtc_event_log_parser.h
View file

@ -466,7 +466,13 @@ class ParsedRtcEventLog {
int64_t first_timestamp() const { return first_timestamp_; }
int64_t last_timestamp() const { return last_timestamp_; }
std::vector<LoggedPacketInfo> GetPacketInfos(PacketDirection direction) const;
std::vector<LoggedIceCandidatePairConfig> GetIceCandidates() const;
std::vector<LoggedIceEvent> GetIceEvents() const;
private:
std::vector<LoggedRouteChangeEvent> GetRouteChanges() const;
bool ParseStreamInternal(
std::istream& stream); // no-presubmit-check TODO(webrtc:8982)