Elad Alon 4a87e1c211 Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead
RtcEventLogImpl no longer hard-codes the way encoding is done. It now relies on RtcEventEncoder for it. This gives two benefits:
1. We can decide between the current encoding and the new encoding (which is still WIP) without code duplication (no need for RtcEventLogImplNew).
2. Encoding is done only when the event needs to be written to a file. This both avoids unnecessary encoding of events which don't end up getting written to a file, as well as is useful for the new, delta-based encoding, which is stateful.

BUG=webrtc:8111

Change-Id: I9517132e5f96b8059002a66fde8d42d3a678c3bb
Reviewed-on: https://webrtc-review.googlesource.com/1365
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20118}
2017-10-03 15:26:56 +00:00

322 lines
12 KiB
C++

/*
* 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 "modules/congestion_controller/delay_based_bwe.h"
#include <algorithm>
#include <cmath>
#include <string>
#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/pacing/paced_sender.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
#include "typedefs.h" // NOLINT(build/include)
namespace {
constexpr int kTimestampGroupLengthMs = 5;
constexpr int kAbsSendTimeFraction = 18;
constexpr int kAbsSendTimeInterArrivalUpshift = 8;
constexpr int kInterArrivalShift =
kAbsSendTimeFraction + kAbsSendTimeInterArrivalUpshift;
constexpr double kTimestampToMs =
1000.0 / static_cast<double>(1 << kInterArrivalShift);
// This ssrc is used to fulfill the current API but will be removed
// after the API has been changed.
constexpr uint32_t kFixedSsrc = 0;
// Parameters for linear least squares fit of regression line to noisy data.
constexpr size_t kDefaultTrendlineWindowSize = 20;
constexpr double kDefaultTrendlineSmoothingCoeff = 0.9;
constexpr double kDefaultTrendlineThresholdGain = 4.0;
constexpr int kMaxConsecutiveFailedLookups = 5;
const char kBweSparseUpdateExperiment[] = "WebRTC-BweSparseUpdateExperiment";
bool BweSparseUpdateExperimentIsEnabled() {
std::string experiment_string =
webrtc::field_trial::FindFullName(kBweSparseUpdateExperiment);
return experiment_string == "Enabled";
}
} // namespace
namespace webrtc {
DelayBasedBwe::Result::Result()
: updated(false),
probe(false),
target_bitrate_bps(0),
recovered_from_overuse(false) {}
DelayBasedBwe::Result::Result(bool probe, uint32_t target_bitrate_bps)
: updated(true),
probe(probe),
target_bitrate_bps(target_bitrate_bps),
recovered_from_overuse(false) {}
DelayBasedBwe::Result::~Result() {}
DelayBasedBwe::DelayBasedBwe(RtcEventLog* event_log, const Clock* clock)
: event_log_(event_log),
clock_(clock),
inter_arrival_(),
trendline_estimator_(),
detector_(),
last_seen_packet_ms_(-1),
uma_recorded_(false),
probe_bitrate_estimator_(event_log),
trendline_window_size_(kDefaultTrendlineWindowSize),
trendline_smoothing_coeff_(kDefaultTrendlineSmoothingCoeff),
trendline_threshold_gain_(kDefaultTrendlineThresholdGain),
consecutive_delayed_feedbacks_(0),
prev_bitrate_(0),
prev_state_(BandwidthUsage::kBwNormal),
in_sparse_update_experiment_(BweSparseUpdateExperimentIsEnabled()) {
LOG(LS_INFO) << "Using Trendline filter for delay change estimation.";
}
DelayBasedBwe::~DelayBasedBwe() {}
DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector,
rtc::Optional<uint32_t> acked_bitrate_bps) {
RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
packet_feedback_vector.end(),
PacketFeedbackComparator()));
RTC_DCHECK_RUNS_SERIALIZED(&network_race_);
// TOOD(holmer): An empty feedback vector here likely means that
// all acks were too late and that the send time history had
// timed out. We should reduce the rate when this occurs.
if (packet_feedback_vector.empty()) {
LOG(LS_WARNING) << "Very late feedback received.";
return DelayBasedBwe::Result();
}
if (!uma_recorded_) {
RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram,
BweNames::kSendSideTransportSeqNum,
BweNames::kBweNamesMax);
uma_recorded_ = true;
}
bool overusing = false;
bool delayed_feedback = true;
bool recovered_from_overuse = false;
BandwidthUsage prev_detector_state = detector_.State();
for (const auto& packet_feedback : packet_feedback_vector) {
if (packet_feedback.send_time_ms < 0)
continue;
delayed_feedback = false;
IncomingPacketFeedback(packet_feedback);
if (!in_sparse_update_experiment_)
overusing |= (detector_.State() == BandwidthUsage::kBwOverusing);
if (prev_detector_state == BandwidthUsage::kBwUnderusing &&
detector_.State() == BandwidthUsage::kBwNormal) {
recovered_from_overuse = true;
}
prev_detector_state = detector_.State();
}
if (in_sparse_update_experiment_)
overusing = (detector_.State() == BandwidthUsage::kBwOverusing);
if (delayed_feedback) {
++consecutive_delayed_feedbacks_;
if (consecutive_delayed_feedbacks_ >= kMaxConsecutiveFailedLookups) {
consecutive_delayed_feedbacks_ = 0;
return OnLongFeedbackDelay(packet_feedback_vector.back().arrival_time_ms);
}
} else {
consecutive_delayed_feedbacks_ = 0;
return MaybeUpdateEstimate(overusing, acked_bitrate_bps,
recovered_from_overuse);
}
return Result();
}
DelayBasedBwe::Result DelayBasedBwe::OnLongFeedbackDelay(
int64_t arrival_time_ms) {
// Estimate should always be valid since a start bitrate always is set in the
// Call constructor. An alternative would be to return an empty Result here,
// or to estimate the throughput based on the feedback we received.
RTC_DCHECK(rate_control_.ValidEstimate());
rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2,
arrival_time_ms);
Result result;
result.updated = true;
result.probe = false;
result.target_bitrate_bps = rate_control_.LatestEstimate();
LOG(LS_WARNING) << "Long feedback delay detected, reducing BWE to "
<< result.target_bitrate_bps;
return result;
}
void DelayBasedBwe::IncomingPacketFeedback(
const PacketFeedback& packet_feedback) {
int64_t now_ms = clock_->TimeInMilliseconds();
// Reset if the stream has timed out.
if (last_seen_packet_ms_ == -1 ||
now_ms - last_seen_packet_ms_ > kStreamTimeOutMs) {
inter_arrival_.reset(
new InterArrival((kTimestampGroupLengthMs << kInterArrivalShift) / 1000,
kTimestampToMs, true));
trendline_estimator_.reset(new TrendlineEstimator(
trendline_window_size_, trendline_smoothing_coeff_,
trendline_threshold_gain_));
}
last_seen_packet_ms_ = now_ms;
uint32_t send_time_24bits =
static_cast<uint32_t>(
((static_cast<uint64_t>(packet_feedback.send_time_ms)
<< kAbsSendTimeFraction) +
500) /
1000) &
0x00FFFFFF;
// Shift up send time to use the full 32 bits that inter_arrival works with,
// so wrapping works properly.
uint32_t timestamp = send_time_24bits << kAbsSendTimeInterArrivalUpshift;
uint32_t ts_delta = 0;
int64_t t_delta = 0;
int size_delta = 0;
if (inter_arrival_->ComputeDeltas(timestamp, packet_feedback.arrival_time_ms,
now_ms, packet_feedback.payload_size,
&ts_delta, &t_delta, &size_delta)) {
double ts_delta_ms = (1000.0 * ts_delta) / (1 << kInterArrivalShift);
trendline_estimator_->Update(t_delta, ts_delta_ms,
packet_feedback.arrival_time_ms);
detector_.Detect(trendline_estimator_->trendline_slope(), ts_delta_ms,
trendline_estimator_->num_of_deltas(),
packet_feedback.arrival_time_ms);
}
if (packet_feedback.pacing_info.probe_cluster_id !=
PacedPacketInfo::kNotAProbe) {
probe_bitrate_estimator_.HandleProbeAndEstimateBitrate(packet_feedback);
}
}
DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
bool overusing,
rtc::Optional<uint32_t> acked_bitrate_bps,
bool recovered_from_overuse) {
Result result;
int64_t now_ms = clock_->TimeInMilliseconds();
rtc::Optional<int> probe_bitrate_bps =
probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps();
// Currently overusing the bandwidth.
if (overusing) {
if (acked_bitrate_bps &&
rate_control_.TimeToReduceFurther(now_ms, *acked_bitrate_bps)) {
result.updated = UpdateEstimate(now_ms, acked_bitrate_bps, overusing,
&result.target_bitrate_bps);
} else if (!acked_bitrate_bps && rate_control_.ValidEstimate() &&
rate_control_.TimeToReduceFurther(
now_ms, rate_control_.LatestEstimate() / 2 - 1)) {
// Overusing before we have a measured acknowledged bitrate. We check
// TimeToReduceFurther (with a fake acknowledged bitrate) to avoid
// reducing too often.
// TODO(tschumim): Improve this and/or the acknowledged bitrate estimator
// so that we (almost) always have a bitrate estimate.
rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, now_ms);
result.updated = true;
result.probe = false;
result.target_bitrate_bps = rate_control_.LatestEstimate();
}
} else {
if (probe_bitrate_bps) {
result.probe = true;
result.updated = true;
result.target_bitrate_bps = *probe_bitrate_bps;
rate_control_.SetEstimate(*probe_bitrate_bps, now_ms);
} else {
result.updated = UpdateEstimate(now_ms, acked_bitrate_bps, overusing,
&result.target_bitrate_bps);
result.recovered_from_overuse = recovered_from_overuse;
}
}
if ((result.updated && prev_bitrate_ != result.target_bitrate_bps) ||
detector_.State() != prev_state_) {
uint32_t bitrate_bps =
result.updated ? result.target_bitrate_bps : prev_bitrate_;
BWE_TEST_LOGGING_PLOT(1, "target_bitrate_bps", now_ms, bitrate_bps);
if (event_log_) {
event_log_->Log(rtc::MakeUnique<RtcEventBweUpdateDelayBased>(
bitrate_bps, detector_.State()));
}
prev_bitrate_ = bitrate_bps;
prev_state_ = detector_.State();
}
return result;
}
bool DelayBasedBwe::UpdateEstimate(int64_t now_ms,
rtc::Optional<uint32_t> acked_bitrate_bps,
bool overusing,
uint32_t* target_bitrate_bps) {
// TODO(terelius): RateControlInput::noise_var is deprecated and will be
// removed. In the meantime, we set it to zero.
const RateControlInput input(
overusing ? BandwidthUsage::kBwOverusing : detector_.State(),
acked_bitrate_bps, 0);
uint32_t prev_target_bitrate_bps = rate_control_.LatestEstimate();
*target_bitrate_bps = rate_control_.Update(&input, now_ms);
return rate_control_.ValidEstimate() &&
prev_target_bitrate_bps != *target_bitrate_bps;
}
void DelayBasedBwe::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
rate_control_.SetRtt(avg_rtt_ms);
}
bool DelayBasedBwe::LatestEstimate(std::vector<uint32_t>* ssrcs,
uint32_t* bitrate_bps) const {
// Currently accessed from both the process thread (see
// ModuleRtpRtcpImpl::Process()) and the configuration thread (see
// Call::GetStats()). Should in the future only be accessed from a single
// thread.
RTC_DCHECK(ssrcs);
RTC_DCHECK(bitrate_bps);
if (!rate_control_.ValidEstimate())
return false;
*ssrcs = {kFixedSsrc};
*bitrate_bps = rate_control_.LatestEstimate();
return true;
}
void DelayBasedBwe::SetStartBitrate(int start_bitrate_bps) {
LOG(LS_WARNING) << "BWE Setting start bitrate to: " << start_bitrate_bps;
rate_control_.SetStartBitrate(start_bitrate_bps);
}
void DelayBasedBwe::SetMinBitrate(int min_bitrate_bps) {
// Called from both the configuration thread and the network thread. Shouldn't
// be called from the network thread in the future.
rate_control_.SetMinBitrate(min_bitrate_bps);
}
int64_t DelayBasedBwe::GetExpectedBwePeriodMs() const {
return rate_control_.GetExpectedBandwidthPeriodMs();
}
} // namespace webrtc