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

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}

logging/BUILD.gn

@@ -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",
     ]

logging/rtc_event_log/logged_events.cc

@@ -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

logging/rtc_event_log/logged_events.h

@@ -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_

logging/rtc_event_log/rtc_event_log_parser.cc

@@ -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()) {
+  for (auto& packet :
-    clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
+       parsed_log.GetPacketInfos(PacketDirection::kOutgoingPacket)) {
-    if (clock.TimeInMicroseconds() >= NextRtpTime()) {
+    if (packet.log_feedback_time.IsFinite() &&
-      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
+        packet.reported_recv_time.IsFinite()) {
-      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) {
       rtp_rtcp_matched.emplace_back(
-            clock.TimeInMilliseconds(), packet.send_time_ms,
+          packet.log_feedback_time.ms(), packet.log_packet_time.ms(),
-            packet.arrival_time_ms, packet.payload_size);
+          packet.reported_recv_time.ms(), packet.size);
     }
-      ++rtcp_iterator;
-    }
-    time_us = std::min(NextRtpTime(), NextRtcpTime());
   }
   return rtp_rtcp_matched;
 }

logging/rtc_event_log/rtc_event_log_parser.h

@@ -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)