Reland "Adds richer packet and ice processing to ParsedRtcEventLog."
This reverts commit 5586d7fb5717fd772463c17a4675491980c5d017.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Adds richer packet and ice processing to ParsedRtcEventLog."
>
> This reverts commit 4306a25dfcaba7defe09f5d4b669736d374fe985.
>
> 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}diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index bc7b3c0..6b29094 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -291,6 +291,7 @@
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 @@
":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 @@
"../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",
]
diff --git a/logging/rtc_event_log/logged_events.cc b/logging/rtc_event_log/logged_events.cc
new file mode 100644
index 0000000..0a4b3df
--- /dev/null
+++ b/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
diff --git a/logging/rtc_event_log/logged_events.h b/logging/rtc_event_log/logged_events.h
index 7ff7e2d..4140265 100644
--- a/logging/rtc_event_log/logged_events.h
+++ b/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 @@
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_
diff --git a/logging/rtc_event_log/rtc_event_log_parser.cc b/logging/rtc_event_log/rtc_event_log_parser.cc
index abf6c7b..0cd87a0 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/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 @@
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 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 @@
return MediaType::ANY;
}
-const std::vector<MatchedSendArrivalTimes> GetNetworkTrace(
- const ParsedRtcEventLog& parsed_log) {
- using RtpPacketType = LoggedRtpPacketOutgoing;
- using TransportFeedbackType = LoggedRtcpPacketTransportFeedback;
+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();
- 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));
+ 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);
}
- const std::vector<TransportFeedbackType>& incoming_rtcp =
- parsed_log.transport_feedbacks(kIncomingPacket);
-
- SimulatedClock clock(0);
+ // 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 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 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 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;
+ 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_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);
+ 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_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);
+ sent->feedback_hold_duration =
+ Timestamp::us(logged.log_time_us()) - sent->log_packet_time;
}
- ++rtp_iterator;
+ sent->last_in_feedback = (&fb == &last_fb);
}
- 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.arrival_time_ms, packet.payload_size);
- }
- ++rtcp_iterator;
+ };
+
+ 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);
}
- time_us = std::min(NextRtpTime(), NextRtcpTime());
+ }
+ 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) {
+ std::vector<MatchedSendArrivalTimes> rtp_rtcp_matched;
+ for (auto& packet :
+ parsed_log.GetPacketInfos(PacketDirection::kOutgoingPacket)) {
+ if (packet.log_feedback_time.IsFinite() &&
+ packet.reported_recv_time.IsFinite()) {
+ rtp_rtcp_matched.emplace_back(
+ packet.log_feedback_time.ms(), packet.log_packet_time.ms(),
+ packet.reported_recv_time.ms(), packet.size);
+ }
}
return rtp_rtcp_matched;
}
diff --git a/logging/rtc_event_log/rtc_event_log_parser.h b/logging/rtc_event_log/rtc_event_log_parser.h
index dc23944..82807fb 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.h
+++ b/logging/rtc_event_log/rtc_event_log_parser.h
@@ -466,7 +466,13 @@
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)