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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}
186 lines
7.7 KiB
C++
186 lines
7.7 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/congestion_controller/probe_bitrate_estimator.h"
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#include <algorithm>
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#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
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#include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
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#include "logging/rtc_event_log/rtc_event_log.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/ptr_util.h"
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namespace {
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// The minumum number of probes we need to receive feedback about in percent
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// in order to have a valid estimate.
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constexpr int kMinReceivedProbesPercent = 80;
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// The minumum number of bytes we need to receive feedback about in percent
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// in order to have a valid estimate.
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constexpr int kMinReceivedBytesPercent = 80;
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// The maximum |receive rate| / |send rate| ratio for a valid estimate.
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constexpr float kMaxValidRatio = 2.0f;
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// The minimum |receive rate| / |send rate| ratio assuming that the link is
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// not saturated, i.e. we assume that we will receive at least
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// kMinRatioForUnsaturatedLink * |send rate| if |send rate| is less than the
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// link capacity.
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constexpr float kMinRatioForUnsaturatedLink = 0.9f;
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// The target utilization of the link. If we know true link capacity
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// we'd like to send at 95% of that rate.
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constexpr float kTargetUtilizationFraction = 0.95f;
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// The maximum time period over which the cluster history is retained.
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// This is also the maximum time period beyond which a probing burst is not
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// expected to last.
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constexpr int kMaxClusterHistoryMs = 1000;
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// The maximum time interval between first and the last probe on a cluster
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// on the sender side as well as the receive side.
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constexpr int kMaxProbeIntervalMs = 1000;
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} // namespace
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namespace webrtc {
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ProbeBitrateEstimator::ProbeBitrateEstimator(RtcEventLog* event_log)
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: event_log_(event_log) {}
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ProbeBitrateEstimator::~ProbeBitrateEstimator() = default;
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int ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
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const PacketFeedback& packet_feedback) {
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int cluster_id = packet_feedback.pacing_info.probe_cluster_id;
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RTC_DCHECK_NE(cluster_id, PacedPacketInfo::kNotAProbe);
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EraseOldClusters(packet_feedback.arrival_time_ms - kMaxClusterHistoryMs);
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int payload_size_bits = packet_feedback.payload_size * 8;
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AggregatedCluster* cluster = &clusters_[cluster_id];
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if (packet_feedback.send_time_ms < cluster->first_send_ms) {
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cluster->first_send_ms = packet_feedback.send_time_ms;
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}
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if (packet_feedback.send_time_ms > cluster->last_send_ms) {
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cluster->last_send_ms = packet_feedback.send_time_ms;
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cluster->size_last_send = payload_size_bits;
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}
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if (packet_feedback.arrival_time_ms < cluster->first_receive_ms) {
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cluster->first_receive_ms = packet_feedback.arrival_time_ms;
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cluster->size_first_receive = payload_size_bits;
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}
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if (packet_feedback.arrival_time_ms > cluster->last_receive_ms) {
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cluster->last_receive_ms = packet_feedback.arrival_time_ms;
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}
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cluster->size_total += payload_size_bits;
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cluster->num_probes += 1;
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RTC_DCHECK_GT(packet_feedback.pacing_info.probe_cluster_min_probes, 0);
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RTC_DCHECK_GT(packet_feedback.pacing_info.probe_cluster_min_bytes, 0);
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int min_probes = packet_feedback.pacing_info.probe_cluster_min_probes *
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kMinReceivedProbesPercent / 100;
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int min_bytes = packet_feedback.pacing_info.probe_cluster_min_bytes *
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kMinReceivedBytesPercent / 100;
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if (cluster->num_probes < min_probes || cluster->size_total < min_bytes * 8)
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return -1;
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float send_interval_ms = cluster->last_send_ms - cluster->first_send_ms;
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float receive_interval_ms =
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cluster->last_receive_ms - cluster->first_receive_ms;
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if (send_interval_ms <= 0 || send_interval_ms > kMaxProbeIntervalMs ||
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receive_interval_ms <= 0 || receive_interval_ms > kMaxProbeIntervalMs) {
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LOG(LS_INFO) << "Probing unsuccessful, invalid send/receive interval"
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<< " [cluster id: " << cluster_id
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<< "] [send interval: " << send_interval_ms << " ms]"
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<< " [receive interval: " << receive_interval_ms << " ms]";
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if (event_log_) {
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event_log_->Log(rtc::MakeUnique<RtcEventProbeResultFailure>(
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cluster_id, ProbeFailureReason::kInvalidSendReceiveInterval));
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}
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return -1;
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}
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// Since the |send_interval_ms| does not include the time it takes to actually
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// send the last packet the size of the last sent packet should not be
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// included when calculating the send bitrate.
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RTC_DCHECK_GT(cluster->size_total, cluster->size_last_send);
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float send_size = cluster->size_total - cluster->size_last_send;
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float send_bps = send_size / send_interval_ms * 1000;
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// Since the |receive_interval_ms| does not include the time it takes to
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// actually receive the first packet the size of the first received packet
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// should not be included when calculating the receive bitrate.
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RTC_DCHECK_GT(cluster->size_total, cluster->size_first_receive);
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float receive_size = cluster->size_total - cluster->size_first_receive;
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float receive_bps = receive_size / receive_interval_ms * 1000;
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float ratio = receive_bps / send_bps;
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if (ratio > kMaxValidRatio) {
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LOG(LS_INFO) << "Probing unsuccessful, receive/send ratio too high"
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<< " [cluster id: " << cluster_id << "] [send: " << send_size
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<< " bytes / " << send_interval_ms
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<< " ms = " << send_bps / 1000 << " kb/s]"
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<< " [receive: " << receive_size << " bytes / "
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<< receive_interval_ms << " ms = " << receive_bps / 1000
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<< " kb/s]"
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<< " [ratio: " << receive_bps / 1000 << " / "
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<< send_bps / 1000 << " = " << ratio << " > kMaxValidRatio ("
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<< kMaxValidRatio << ")]";
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if (event_log_) {
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event_log_->Log(rtc::MakeUnique<RtcEventProbeResultFailure>(
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cluster_id, ProbeFailureReason::kInvalidSendReceiveRatio));
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}
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return -1;
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}
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LOG(LS_INFO) << "Probing successful"
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<< " [cluster id: " << cluster_id << "] [send: " << send_size
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<< " bytes / " << send_interval_ms << " ms = " << send_bps / 1000
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<< " kb/s]"
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<< " [receive: " << receive_size << " bytes / "
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<< receive_interval_ms << " ms = " << receive_bps / 1000
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<< " kb/s]";
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float res = std::min(send_bps, receive_bps);
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// If we're receiving at significantly lower bitrate than we were sending at,
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// it suggests that we've found the true capacity of the link. In this case,
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// set the target bitrate slightly lower to not immediately overuse.
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if (receive_bps < kMinRatioForUnsaturatedLink * send_bps) {
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RTC_DCHECK_GT(send_bps, receive_bps);
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res = kTargetUtilizationFraction * receive_bps;
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}
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if (event_log_) {
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event_log_->Log(
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rtc::MakeUnique<RtcEventProbeResultSuccess>(cluster_id, res));
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}
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estimated_bitrate_bps_ = rtc::Optional<int>(res);
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return *estimated_bitrate_bps_;
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}
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rtc::Optional<int>
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ProbeBitrateEstimator::FetchAndResetLastEstimatedBitrateBps() {
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rtc::Optional<int> estimated_bitrate_bps = estimated_bitrate_bps_;
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estimated_bitrate_bps_.reset();
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return estimated_bitrate_bps;
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}
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void ProbeBitrateEstimator::EraseOldClusters(int64_t timestamp_ms) {
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for (auto it = clusters_.begin(); it != clusters_.end();) {
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if (it->second.last_receive_ms < timestamp_ms) {
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it = clusters_.erase(it);
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} else {
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++it;
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}
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}
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}
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} // namespace webrtc
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