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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

15
modules/bitrate_controller/send_side_bandwidth_estimation.h
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@ -42,18 +42,10 @@ class SendSideBandwidthEstimation {
// Call when we receive a RTCP message with a ReceiveBlock.
void UpdateReceiverBlock(uint8_t fraction_loss,
int64_t rtt_ms,
int64_t rtt,
int number_of_packets,
int64_t now_ms);
// Call when we receive a RTCP message with a ReceiveBlock.
void UpdatePacketsLost(int packets_lost,
int number_of_packets,
int64_t now_ms);
// Call when we receive a RTCP message with a ReceiveBlock.
void UpdateRtt(int64_t rtt, int64_t now_ms);
void SetBitrates(int send_bitrate,
int min_bitrate,
int max_bitrate);
@ -66,7 +58,7 @@ class SendSideBandwidthEstimation {
bool IsInStartPhase(int64_t now_ms) const;
void UpdateUmaStatsPacketsLost(int64_t now_ms, int packets_lost);
void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets);
// Updates history of min bitrates.
// After this method returns min_bitrate_history_.front().second contains the
@ -80,7 +72,7 @@ class SendSideBandwidthEstimation {
std::deque<std::pair<int64_t, uint32_t> > min_bitrate_history_;
// incoming filters
int lost_packets_since_last_loss_update_;
int lost_packets_since_last_loss_update_Q8_;
int expected_packets_since_last_loss_update_;
uint32_t current_bitrate_bps_;
@ -103,7 +95,6 @@ class SendSideBandwidthEstimation {
int initially_lost_packets_;
int bitrate_at_2_seconds_kbps_;
UmaState uma_update_state_;
UmaState uma_rtt_state_;
std::vector<bool> rampup_uma_stats_updated_;
RtcEventLog* event_log_;
int64_t last_rtc_event_log_ms_;