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Revert "Revert "Revert "Reland "Moved congestion controller to task queue.""""

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This reverts commit 65792c5a5c542201f7b9feefded505842692e6ed.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Revert "Revert "Reland "Moved congestion controller to task queue."""
> 
> This reverts commit 4e849f6925b2ac44b0957a228d7131fc391fca54.
> 
> Reason for revert: <INSERT REASONING HERE>
> 
> Original change's description:
> > Revert "Reland "Moved congestion controller to task queue.""
> > 
> > This reverts commit 57daeb7ac7f3d80992905b53fea500953fcfd793.
> > 
> > Reason for revert: Cause increased congestion and deadlocks in downstream project
> > 
> > Original change's description:
> > > Reland "Moved congestion controller to task queue."
> > > 
> > > This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9.
> > > 
> > > Original change's description:
> > > > Moved congestion controller to task queue.
> > > > 
> > > > The goal of this work is to make it easier to experiment with the
> > > > bandwidth estimation implementation. For this reason network control
> > > > functionality is moved from SendSideCongestionController(SSCC),
> > > > PacedSender and BitrateController to the newly created
> > > > GoogCcNetworkController which implements the newly created
> > > > NetworkControllerInterface. This allows the implementation to be
> > > > replaced at runtime in the future.
> > > > 
> > > > This is the first part of a split of a larger CL, see:
> > > > https://webrtc-review.googlesource.com/c/src/+/39788/8
> > > > For further explanations.
> > > > 
> > > > Bug: webrtc:8415
> > > > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> > > > Reviewed-on: https://webrtc-review.googlesource.com/43840
> > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#21868}
> > > 
> > > Bug: webrtc:8415
> > > Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da
> > > Reviewed-on: https://webrtc-review.googlesource.com/48000
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21899}
> > 
> > TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
> > 
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> > 
> > Bug: webrtc:8415
> > Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83
> > Reviewed-on: https://webrtc-review.googlesource.com/52980
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22017}
> 
> TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
> 
> Change-Id: I3393b74370c4f4d0955f50728005b2b925be169b
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8415
> Reviewed-on: https://webrtc-review.googlesource.com/53262
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22023}

TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org

Change-Id: Id68ad986ee51142b7be3381d0793709b4392fe2c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/53360
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22024}
This commit is contained in:
Sebastian Jansson
2018-02-14 16:53:38 +00:00
committed by Commit Bot
parent 65792c5a5c
commit ea86bb74fc
57 changed files with 844 additions and 3006 deletions
Download Patch File Download Diff File Expand all files Collapse all files

59
modules/bitrate_controller/send_side_bandwidth_estimation.cc
View File

@ -105,7 +105,7 @@ bool ReadBweLossExperimentParameters(float* low_loss_threshold,
} // namespace
SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
: lost_packets_since_last_loss_update_(0),
: lost_packets_since_last_loss_update_Q8_(0),
expected_packets_since_last_loss_update_(0),
current_bitrate_bps_(0),
min_bitrate_configured_(congestion_controller::GetMinBitrateBps()),
@ -125,7 +125,6 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
initially_lost_packets_(0),
bitrate_at_2_seconds_kbps_(0),
uma_update_state_(kNoUpdate),
uma_rtt_state_(kNoUpdate),
rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
event_log_(event_log),
last_rtc_event_log_ms_(-1),
@ -207,28 +206,24 @@ void SendSideBandwidthEstimation::UpdateDelayBasedEstimate(
}
void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
int64_t rtt_ms,
int64_t rtt,
int number_of_packets,
int64_t now_ms) {
const int kRoundingConstant = 128;
int packets_lost = (static_cast<int>(fraction_loss) * number_of_packets +
kRoundingConstant) >>
8;
UpdatePacketsLost(packets_lost, number_of_packets, now_ms);
UpdateRtt(rtt_ms, now_ms);
}
void SendSideBandwidthEstimation::UpdatePacketsLost(int packets_lost,
int number_of_packets,
int64_t now_ms) {
last_feedback_ms_ = now_ms;
if (first_report_time_ms_ == -1)
first_report_time_ms_ = now_ms;
// Update RTT if we were able to compute an RTT based on this RTCP.
// FlexFEC doesn't send RTCP SR, which means we won't be able to compute RTT.
if (rtt > 0)
last_round_trip_time_ms_ = rtt;
// Check sequence number diff and weight loss report
if (number_of_packets > 0) {
// Calculate number of lost packets.
const int num_lost_packets_Q8 = fraction_loss * number_of_packets;
// Accumulate reports.
lost_packets_since_last_loss_update_ += packets_lost;
lost_packets_since_last_loss_update_Q8_ += num_lost_packets_Q8;
expected_packets_since_last_loss_update_ += number_of_packets;
// Don't generate a loss rate until it can be based on enough packets.
@ -236,22 +231,21 @@ void SendSideBandwidthEstimation::UpdatePacketsLost(int packets_lost,
return;
has_decreased_since_last_fraction_loss_ = false;
int64_t lost_q8 = lost_packets_since_last_loss_update_ << 8;
int64_t expected = expected_packets_since_last_loss_update_;
last_fraction_loss_ = std::min<int>(lost_q8 / expected, 255);
last_fraction_loss_ = lost_packets_since_last_loss_update_Q8_ /
expected_packets_since_last_loss_update_;
// Reset accumulators.
lost_packets_since_last_loss_update_ = 0;
lost_packets_since_last_loss_update_Q8_ = 0;
expected_packets_since_last_loss_update_ = 0;
last_packet_report_ms_ = now_ms;
UpdateEstimate(now_ms);
}
UpdateUmaStatsPacketsLost(now_ms, packets_lost);
UpdateUmaStats(now_ms, rtt, (fraction_loss * number_of_packets) >> 8);
}
void SendSideBandwidthEstimation::UpdateUmaStatsPacketsLost(int64_t now_ms,
int packets_lost) {
void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
int64_t rtt,
int lost_packets) {
int bitrate_kbps = static_cast<int>((current_bitrate_bps_ + 500) / 1000);
for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
if (!rampup_uma_stats_updated_[i] &&
@ -262,12 +256,14 @@ void SendSideBandwidthEstimation::UpdateUmaStatsPacketsLost(int64_t now_ms,
}
}
if (IsInStartPhase(now_ms)) {
initially_lost_packets_ += packets_lost;
initially_lost_packets_ += lost_packets;
} else if (uma_update_state_ == kNoUpdate) {
uma_update_state_ = kFirstDone;
bitrate_at_2_seconds_kbps_ = bitrate_kbps;
RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets",
initially_lost_packets_, 0, 100, 50);
RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", static_cast<int>(rtt), 0,
2000, 50);
RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
bitrate_at_2_seconds_kbps_, 0, 2000, 50);
} else if (uma_update_state_ == kFirstDone &&
@ -280,19 +276,6 @@ void SendSideBandwidthEstimation::UpdateUmaStatsPacketsLost(int64_t now_ms,
}
}
void SendSideBandwidthEstimation::UpdateRtt(int64_t rtt_ms, int64_t now_ms) {
// Update RTT if we were able to compute an RTT based on this RTCP.
// FlexFEC doesn't send RTCP SR, which means we won't be able to compute RTT.
if (rtt_ms > 0)
last_round_trip_time_ms_ = rtt_ms;
if (!IsInStartPhase(now_ms) && uma_rtt_state_ == kNoUpdate) {
uma_rtt_state_ = kDone;
RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", static_cast<int>(rtt_ms), 0,
2000, 50);
}
}
void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
uint32_t new_bitrate = current_bitrate_bps_;
// We trust the REMB and/or delay-based estimate during the first 2 seconds if
@ -374,7 +357,7 @@ void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
new_bitrate *= 0.8;
// Reset accumulators since we've already acted on missing feedback and
// shouldn't to act again on these old lost packets.
lost_packets_since_last_loss_update_ = 0;
lost_packets_since_last_loss_update_Q8_ = 0;
expected_packets_since_last_loss_update_ = 0;
last_timeout_ms_ = now_ms;
}