Reland "Reland "Improving robustness of feedback matching code in event log parser.""

This is a reland of 0870c70b0471c3bae16ad9a6732d812ee25446dd

Original change's description:
> Reland "Improving robustness of feedback matching code in event log parser."
> 
> This is a reland of a1e4fbb25371867349a0c2ed6ba62224735a2ec7
> 
> Original change's description:
> > Improving robustness of feedback matching code in event log parser.
> > 
> > Removes the dependency on TransportFeedbackAdapter thereby removing
> > some of the complexity that came with it, in particular, we don't fill
> > in missing packets. This makes the code easier to debug and avoids some
> > confusing logging that's not relevant for the parser.
> > 
> > Bug: webrtc:9883
> > Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27739}
> 
> Bug: webrtc:9883
> Change-Id: I460d0c576626614fb4ce2c3d5e3ddbb5d1c122cf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134106
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27763}

Bug: webrtc:9883
Change-Id: I1f80ed1f63ad75fbb97f5f401fe486d19c057f75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134462
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27829}
This commit is contained in:
Sebastian Jansson
2019-05-02 17:07:22 +02:00
committed by Commit Bot
parent 4fb12b0cae
commit b468616a69
7 changed files with 192 additions and 99 deletions

View File

@ -335,8 +335,8 @@ if (rtc_enable_protobuf) {
"../api/units:time_delta",
"../api/units:timestamp",
"../call:video_stream_api",
"../modules:module_api",
"../modules/audio_coding:audio_network_adaptor",
"../modules/congestion_controller/rtp:transport_feedback",
"../modules/remote_bitrate_estimator",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",

View File

@ -392,7 +392,9 @@ struct LoggedRtcpPacketPli {
};
struct LoggedRtcpPacketTransportFeedback {
LoggedRtcpPacketTransportFeedback() = default;
LoggedRtcpPacketTransportFeedback()
: transport_feedback(/*include_timestamps=*/true, /*include_lost*/ true) {
}
LoggedRtcpPacketTransportFeedback(
int64_t timestamp_us,
const rtcp::TransportFeedback& transport_feedback)

View File

@ -30,7 +30,7 @@
#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/include/module_common_types.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@ -1918,7 +1918,6 @@ std::vector<LoggedPacketInfo> ParsedRtcEventLog::GetPacketInfos(
AddSendStreamInfos(&streams, video_send_configs(), LoggedMediaType::kVideo);
}
TransportFeedbackAdapter feedback_adapter;
std::vector<OverheadChangeEvent> overheads =
GetOverheadChangingEvents(GetRouteChanges(), direction);
auto overhead_iter = overheads.begin();
@ -1926,6 +1925,7 @@ std::vector<LoggedPacketInfo> ParsedRtcEventLog::GetPacketInfos(
std::map<int64_t, size_t> indices;
uint16_t current_overhead = kDefaultOverhead;
Timestamp last_log_time = Timestamp::Zero();
SequenceNumberUnwrapper seq_num_unwrapper;
auto advance_time = [&](Timestamp new_log_time) {
if (overhead_iter != overheads.end() &&
@ -1959,72 +1959,82 @@ std::vector<LoggedPacketInfo> ParsedRtcEventLog::GetPacketInfos(
}
LoggedPacketInfo logged(rtp, stream->media_type, stream->rtx, capture_time);
logged.overhead = current_overhead;
if (rtp.header.extension.hasTransportSequenceNumber) {
if (logged.has_transport_seq_no) {
logged.log_feedback_time = Timestamp::PlusInfinity();
RtpPacketSendInfo packet_info;
packet_info.ssrc = rtp.header.ssrc;
packet_info.transport_sequence_number =
rtp.header.extension.transportSequenceNumber;
packet_info.rtp_sequence_number = rtp.header.sequenceNumber;
packet_info.has_rtp_sequence_number = true;
packet_info.length = rtp.total_length;
feedback_adapter.AddPacket(packet_info,
0u, // Should this be current_overhead?
Timestamp::ms(rtp.log_time_ms()));
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;
auto sent_packet_msg = feedback_adapter.ProcessSentPacket(sent_packet);
RTC_CHECK(sent_packet_msg);
indices[sent_packet_msg->sequence_number] = packets.size();
int64_t unwrapped_seq_num =
seq_num_unwrapper.Unwrap(logged.transport_seq_no);
indices[unwrapped_seq_num] = packets.size();
}
packets.push_back(logged);
};
auto feedback_handler = [&](const LoggedRtcpPacketTransportFeedback& logged) {
advance_time(Timestamp::ms(logged.log_time_ms()));
auto msg = feedback_adapter.ProcessTransportFeedback(
logged.transport_feedback, Timestamp::ms(logged.log_time_ms()));
if (!msg.has_value() || msg->packet_feedbacks.empty())
return;
Timestamp feedback_base_time = Timestamp::MinusInfinity();
absl::optional<int64_t> last_feedback_base_time_us;
auto& last_fb = msg->packet_feedbacks.back();
Timestamp last_recv_time = last_fb.receive_time;
// This can happen if send time info is missing for the real last packet in
// the feedback, allowing the reported last packet to med indicated as lost.
if (last_recv_time.IsInfinite())
RTC_LOG(LS_WARNING) << "No receive time for last packet in feedback.";
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;
// If we have received feedback with a valid receive time for this packet
// before, we keep the previous values.
if (sent->log_feedback_time.IsFinite() &&
sent->reported_recv_time.IsFinite())
continue;
sent->log_feedback_time = msg->feedback_time;
if (last_recv_time.IsFinite()) {
if (direction == PacketDirection::kOutgoingPacket) {
sent->feedback_hold_duration = last_recv_time - fb.receive_time;
auto feedback_handler =
[&](const LoggedRtcpPacketTransportFeedback& logged_rtcp) {
auto log_feedback_time = Timestamp::ms(logged_rtcp.log_time_ms());
advance_time(log_feedback_time);
const auto& feedback = logged_rtcp.transport_feedback;
// Add timestamp deltas to a local time base selected on first packet
// arrival. This won't be the true time base, but makes it easier to
// manually inspect time stamps.
if (!last_feedback_base_time_us) {
feedback_base_time = log_feedback_time;
} else {
sent->feedback_hold_duration =
Timestamp::ms(logged.log_time_ms()) - sent->log_packet_time;
feedback_base_time += TimeDelta::us(
feedback.GetBaseDeltaUs(*last_feedback_base_time_us));
}
}
sent->last_in_feedback = (&fb == &last_fb);
}
};
last_feedback_base_time_us = feedback.GetBaseTimeUs();
std::vector<LoggedPacketInfo*> packet_feedbacks;
packet_feedbacks.reserve(feedback.GetAllPackets().size());
Timestamp receive_timestamp = feedback_base_time;
for (const auto& packet : feedback.GetAllPackets()) {
int64_t unwrapped_seq_num =
seq_num_unwrapper.Unwrap(packet.sequence_number());
auto it = indices.find(unwrapped_seq_num);
if (it == indices.end()) {
RTC_LOG(LS_WARNING) << "Received feedback for unknown packet: "
<< unwrapped_seq_num;
continue;
}
LoggedPacketInfo* sent = &packets[it->second];
if (log_feedback_time - sent->log_packet_time >
TimeDelta::seconds(60)) {
RTC_LOG(LS_WARNING)
<< "Received very late feedback, possibly due to wraparound.";
continue;
}
if (packet.received()) {
receive_timestamp += TimeDelta::us(packet.delta_us());
if (sent->reported_recv_time.IsInfinite()) {
sent->reported_recv_time = Timestamp::ms(receive_timestamp.ms());
sent->log_feedback_time = log_feedback_time;
}
} else {
if (sent->reported_recv_time.IsInfinite() &&
sent->log_feedback_time.IsInfinite()) {
sent->reported_recv_time = Timestamp::PlusInfinity();
sent->log_feedback_time = log_feedback_time;
}
}
packet_feedbacks.push_back(sent);
}
if (packet_feedbacks.empty())
return;
LoggedPacketInfo* last = packet_feedbacks.back();
last->last_in_feedback = true;
for (LoggedPacketInfo* fb : packet_feedbacks) {
if (direction == PacketDirection::kOutgoingPacket) {
fb->feedback_hold_duration =
last->reported_recv_time - fb->reported_recv_time;
} else {
fb->feedback_hold_duration =
log_feedback_time - fb->log_packet_time;
}
}
};
RtcEventProcessor process;
for (const auto& rtp_packets : rtp_packets_by_ssrc(direction)) {