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Author SHA1 Message Date
Filip Hlasek
f2fe43b655 Don't round the computed time deltas to nearest ms. 
Resolving https://bugs.chromium.org/p/webrtc/issues/detail?id=14023

At the moment, in DelayBasedBwe the time deltas are rounded to the
nearest millisecond. This change makes sure the numbers are passed as
doubles as expected by the TrendlineEstimator.

Change-Id: I68882547fb19af0e67e7b5d8de4159083a54d7eb
Bug: webrtc:14023
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261320
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36806}
 2022-05-09 07:02:46 +00:00
Jonas Oreland
e62c2f2c77 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 
rename WebRtcKeyValueConfig to FieldTrialsView

Bug: webrtc:10335
Change-Id: If725bd498c4c3daf144bee638230fa089fdde833
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36365}
 2022-03-29 10:14:00 +00:00
Diep Bui
4ceea65848 Integrate trendline estimator into loss based bwe v2. 
Bug: webrtc:12707
Change-Id: I510d3799c14599344d1714178e42b29e7c0c06d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254380
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36236}
 2022-03-17 13:07:44 +00:00
Per Kjellander
c6d3a7a691 Ensure returned delay based estimate from probe can be clamped by 
AimdRateControl

AimdRateControl can potentially clamp bitrate in SetEstimate.
DelayBasedBwe::MaybeUpdateEstimate should therefore check the result before using the probe bitrate.
Otherwise, BWE may drop on next update.

Bug: none
Change-Id: I8b1b3549a2bcd981e941b1cc802c984828d68261
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252444
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36099}
 2022-03-01 09:45:30 +00:00
Per Kjellander
6b667a8fe2 Clean up expreiment WebRTC-Bwe-NewInterArrivalDelta 
The experiment has been per default enabled since
https://webrtc.googlesource.com/src/+/62b340545f80baaa449c2159a8e44052c74116c9
submitted 20210914.

Bug: webrtc:12269
Change-Id: I730b603f4d0e382758fd4a6df6ccef9d8b76ea82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246105
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35683}
 2022-01-13 13:32:24 +00:00
Christoffer Rodbro
62b340545f Default the safetyped calculation of packet interarrival times. 
This defaults the calculation landed in cl 196502. The less readable legacy calculation method will be deleted in a future CL.


Bug: none
Change-Id: Ida02a5208e354835b964c69355ad1e9d5bba18aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231956
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35027}
 2021-09-17 13:58:55 +00:00
Per Kjellander
ea969d287b Reland Addd class InterArrivalDelta to goog_cc 
This time the class is added but only used if the field trial "WebRTC-Bwe-NewInterArrivalDelta/Enabled/" is enabled.
Original cl description:

This cl copies modules/remote_bitrate_estimator/inter_arrival.x to inter_arrival.h and interrival_delta.cc in goog_cc
but modified to use webrtc::Timestamp and webrtc::Timedelta in order to avoid having to use 24 bit time repressentation.

patchset 1 is a pure revert of the revert https://webrtc-review.googlesource.com/c/src/+/196343
patchset 2 contains a modification to allow running it behind an experiment.


Bug: webrtc:12269
Change-Id: Ide80e9f5243362799a2cc1f0fcf7e613e707d851
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196502
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32784}
 2020-12-07 11:02:22 +00:00
Christoffer Rodbro
d13178cb55 Cleanup obsolete filtering of small packets in delay based estimator. 
Also deletes unused constructor in Results struct.

Bug: webrtc:10932
Change-Id: Id33f57db30df49aa23fb0b5959812cc3834f1eaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196508
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32777}
 2020-12-04 16:02:55 +00:00
Per Kjellander
a760bca072 Revert "Add class InterArrivalDelta to goog_cc" 
This reverts commit 0496a41211.

Reason for revert: Causes unexpected changes in perf tests.

Original change's description:
> Add class InterArrivalDelta to goog_cc
>
> This cl copies modules/remote_bitrate_estimator/inter_arrival.x to inter_arrival.h and interrival_delta.cc in goog_cc in the first patchset.
> In the following- this class is modified to use webrtc::Timestamp and webrtc::Timedelta in order to avoid having to use 24 bit time repressentation.
>
> Bug: none
> Change-Id: I9befe6e3e283cf7e21efa974ae33e8a83e26cbe6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194004
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32733}

TBR=perkj@webrtc.org,crodbro@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: none
Change-Id: I725b246f6ec0c293cb3ada39b1a65a14ef9a001e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196343
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32765}
 2020-12-03 22:50:42 +00:00
Per Kjellander
0496a41211 Add class InterArrivalDelta to goog_cc 
This cl copies modules/remote_bitrate_estimator/inter_arrival.x to inter_arrival.h and interrival_delta.cc in goog_cc in the first patchset.
In the following- this class is modified to use webrtc::Timestamp and webrtc::Timedelta in order to avoid having to use 24 bit time repressentation.

Bug: none
Change-Id: I9befe6e3e283cf7e21efa974ae33e8a83e26cbe6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194004
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32733}
 2020-12-01 20:13:12 +00:00
Niels Möller
0d863f72a8 Cleanup of bwe_defines.h 
Delete unused macros BWE_MIN and BWE_MAX.

Move enum RateControlState: Make it a private enum class in
AimdRateControl, the only user.

Change users of the header file that only need BandwidthUsage, to
instead include api/network_state_predictor.h, the file defining this
class. As a result, fewer dependencies on
modules/remote_bitrate_estimator.

Bug: None
Change-Id: I4450c79dd58e1875d64dd74d1ae2cb7d911a14b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195222
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32704}
 2020-11-26 12:26:02 +00:00
Mirko Bonadei
57cabed0b0 Replace std::string::find() == 0 with absl::StartsWith. 
Bug: None
Change-Id: I070c4a5d19455f3a5c5d3ccc05f418545c351987
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172584
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30960}
 2020-04-01 11:15:00 +00:00
Björn Terelius
987ef48258 Adds field trial to separate audio and video packets for delay-based overuse detection. 
The decision to route audio packets to a separate overuse detector
is off by default and requires the field trial
WebRTC-Bwe-SeparateAudioPackets/enabled,packet_threshold:10,time_threshold:1000ms/
The parameters control the threshold for switching over to the
audio overuse detector if we stop receiving feedback for video.

Bug: webrtc:10932
Change-Id: Icdde35bc7a98b18b1a344bd2d620a890fd9421d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168342
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30694}
 2020-03-05 16:29:55 +00:00
Danil Chapovalov
5528402ef8 Use newer version of TimeDelta and TimeStamp factories in modules/ 
This change generated with following commands:
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I117d64a54950be040d996035c54bc0043310943a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168340
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30489}
 2020-02-10 11:49:57 +00:00
Björn Terelius
fd0e32a87a Fix filtering of small packets in delay-based BWE 
crodbro@ found that the previous field trial, which filtered the deltas
in the trendline estimator, can increase the noise caused by varying
packet sizes. Moving the filtering to the DelayBasedBwe class fixes the
issue.

To avoid confusion, we've updated the field trial name, so e.g.
WebRTC-BweIgnoreSmallPacketsFix/small:200bytes,large:200bytes,
                                fraction_large:0.25,smoothing:0.1/
should be used to enable the feature.

Bug: webrtc:10932
Change-Id: If77e83043c37fff909038405f634e541ce41abb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159711
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29804}
 2019-11-15 14:53:59 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique. 
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
 2019-09-17 15:47:29 +00:00
Per Kjellander
dc7d2c6fd7 Backoff to acked bitrate during first overuse detection 
In DelayBasedBwe, in experiment WebRTC-Bwe-AlrLimitedBackoff, back off relative the BWE only after the first detected overuse. The first time overuse is detected, back down to the acked bitrate.

The idea is to faster drop BWE in the beginning of the call when the initial BWE guess may be too high. Withouth this, it may take a too long time to initially back down.

BUG=webrtc:10542

Change-Id: I2a11457d2391ad25658e7c13d9cae02a38973ecb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152541
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29163}
 2019-09-12 10:51:45 +00:00
Danil Chapovalov
16cb1f61c0 Stop using rtc_event.h forward header 
Bug: webrtc:10206
Change-Id: I16905ec745673178195d6715fda6175c31500163
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151601
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29149}
 2019-09-11 08:20:29 +00:00
Bjorn Terelius
602942f14c Filter out small packets from delay-based overuse detection. 
The change is behind a field trial. The intention is to use this
to (heuristically) base the bandwidth estimate only on video packets
even if both audio and video packets have transport sequence numbers.

Bug: webrtc:10932
Change-Id: I6cc5bb9ab6f1a3f25b84ee6ac78e4abb4112032e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150787
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29031}
 2019-09-01 17:57:01 +00:00
Danil Chapovalov
83bbe91398 Delete deprecated rtc_event_log header 
Bug: webrtc:10206
Change-Id: I9ed3148843c647372993729b87c0e74741ab540b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28791}
 2019-08-07 10:58:17 +00:00
Sebastian Jansson
88290ae358 Reland "Cleanup of RTP references in GoogCC implementation." 
This is a reland of fa79081dca

It crashed due to inability to handle small timestamps in probe
estimator. This was fixed by moving history window check to avoid
subtracting from the timestamp.

Original change's description:
> Cleanup of RTP references in GoogCC implementation.
>
> As the send time congestion controller now has been removed,
> we don't need the RTP related constructs anymore.
>
> Bug: webrtc:9510
> Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28330}

Bug: webrtc:9510
Change-Id: I3bf91222068e4fbb6aa159bfeb7a73e00bb6a0d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143165
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28347}
 2019-06-24 09:10:52 +00:00
Sebastian Jansson
7953ad5dab Revert "Cleanup of RTP references in GoogCC implementation." 
This reverts commit fa79081dca.

Reason for revert: Breaks downstream project.

Original change's description:
> Cleanup of RTP references in GoogCC implementation.
> 
> As the send time congestion controller now has been removed,
> we don't need the RTP related constructs anymore.
> 
> Bug: webrtc:9510
> Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28330}

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

Change-Id: I562365fc5d1da68326d603338ccc6371114d7e12
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9510
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143164
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28331}
 2019-06-20 10:21:51 +00:00
Sebastian Jansson
fa79081dca Cleanup of RTP references in GoogCC implementation. 
As the send time congestion controller now has been removed,
we don't need the RTP related constructs anymore.

Bug: webrtc:9510
Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28330}
 2019-06-20 10:08:29 +00:00
Sebastian Jansson
acab559c7b Adds overuse predictor to GoogCC. 
Bug: webrtc:10498
Change-Id: Ic97c16d28cbc1e30609f6c1daa3a61423d44641c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136924
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28012}
 2019-05-21 16:50:39 +00:00
Per Kjellander
eb9bf411f3 Fix problem in WebRTC-Bwe-AlrLimitedBackoff experiment 
When backing down, the delay based estimator can still use acked bitrate instead of the last set estimate.

Original code was reviewed in:
https://webrtc-review.googlesource.com/c/src/+/113880

BUG=webrtc:10144

Change-Id: Ia6e2d6d7d05f88f7e51d61b6e37c61a89adccf8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135950
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27912}
 2019-05-10 13:58:27 +00:00
Sebastian Jansson
5e3d0f88c8 Moves trendline estimation configuration to trendline_estimator.cc 
Bug: webrtc:9883
Change-Id: I5b2139de0c085e1c5ec7c55b5c5ff9a95067e170
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134205
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27767}
 2019-04-25 12:27:19 +00:00
Per Kjellander
416d5db75d Add field trial to AimdRateController to only increase while not in ALR 
The idea is that when ALR is detected, the encoder can not produce the bitrate
needed for the delay based estimator to detect overuse and thus the delay based
estimator should not be allowed to increase further.
Likewise, if ALR is not detected, the delay based estimator is allowed to
increase the BWE to ensure that there is no region where the BWE can get stuck.

BUG=webrtc:10542

Change-Id: Ic94b708461c9077fd09132ee4ecb6279ffcd5f99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133190
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27661}
 2019-04-17 12:05:24 +00:00
Per Kjellander
494947bbcf Remove direct use of FieldTrials from modules/remote_bitrate_estimator 
Instead use WebRtcKeyValueConfig and FieldTrialBasedConfig

BUG=webrtc:10335

Change-Id: Ie148cb466f86d8fa1ded5c7f125fbcccf6e7dbe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132714
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27642}
 2019-04-16 13:23:12 +00:00
Sebastian Jansson
df88cc014a Allow injection of network estimator into GoogCC. 
Bug: webrtc:10498
Change-Id: Ie9225411db201dfcfa0a37a3c40992acbdc215bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132402
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27624}
 2019-04-15 14:12:08 +00:00
Ying Wang
0810a7c25a Add base class NetworkPredictor and NetworkPredictorFactory and wire up. 
Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe.

Bug: webrtc:10492
Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27543}
 2019-04-10 12:38:58 +00:00
Niels Möller
6ec2f547d7 Fix mis-spelled TODO items 
No-Try: true
Tbr: kwiberg@webrtc.org
Bug: webrtc:10198
Change-Id: Iedcafb89d3fd39a812d410db1b2ed6fac8fade38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125724
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26972}
 2019-03-05 13:45:39 +00:00
Sebastian Jansson
95edb037a4 Adds WebRtcKeyValueConfig interface 
The WebRtcKeyValueConfig interface allows providing custom key value
configurations that changes per instance of GoogCcNetworkController.

Bug: webrtc:10009
Change-Id: I520fff030d1c3c755455ec8f67896fe8a6b4d970
Reviewed-on: https://webrtc-review.googlesource.com/c/116989
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26312}
 2019-01-18 08:45:08 +00:00
Erik Språng
b3564c1cb2 Back off relative to current estimate rather than ack rate when in ALR. 
If we're in ALR, the acked rate is going to be significantly lower than
the current estimate for the link capacity. If we need to back off in
this situation (usually caused by latency spikes), this CL makes us back
off relative to current estimate if. We then immediately send a new
probe just in case the network did actually change.

All of this is behind experiment flags for now.

Bug: webrtc:10144
Change-Id: I062a259c36417eea2211d44592ef7fc979aa22b7
Reviewed-on: https://webrtc-review.googlesource.com/c/113880
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26045}
 2018-12-18 12:33:08 +00:00
Bjorn Terelius
24779d8229 Missing packet send time should not cause BWE backoff. 
The removed coded causes problems if the same RTCP packet is forwarded
to the congestion controller multiple times.

Bug: webrtc:10125
Change-Id: I659d8f8f3ce3c643710156fa81176ceeaedd714a
Reviewed-on: https://webrtc-review.googlesource.com/c/114165
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26016}
 2018-12-14 14:48:48 +00:00
Bjorn Terelius
6b3d18164b Remove unused BWE field trial strings. 
Bug: None
Change-Id: I38d2e5495ddfe0b9f1493efc38ef7df95e7fd207
Reviewed-on: https://webrtc-review.googlesource.com/c/111258
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25798}
 2018-11-27 13:05:43 +00:00
Sebastian Jansson
885cf60106 Moves ProbeBitrateEstimator from DelayBasedBwe. 
This prepares for providing an additional implementation of delay based
rate control. By moving the probe controller, less code will have to be
added in the upcoming CL.

Bug: webrtc:9718
Change-Id: I64eb2c8f5f7950b6e9d209f110dc0a757c710b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/111860
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25770}
 2018-11-23 13:43:51 +00:00
Sebastian Jansson
5f00995964 Using unit classes in AimdRateControl. 
Bug: webrtc:9718
Change-Id: I1efed4e55c9d1ccec3c32ed012cb3cd82d7f4ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/110788
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25705}
 2018-11-20 08:04:11 +00:00
Sebastian Jansson
b6787bcd79 Using data unit classes in DelayBasedBwe. 
Bug: webrtc:9718
Change-Id: I1b6ed37afd7680dfad6267addfe46155c378525d
Reviewed-on: https://webrtc-review.googlesource.com/c/110903
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25702}
 2018-11-19 20:18:36 +00:00
Sebastian Jansson
af6d741fe1 Makes send time information in feedback non-optional. 
This makes it safer to reason about the common case where send
time information is available. We don't have to either assume that
it's available, or check it everywhere the PacketResult struct is used.

To achieve this, a new field is added to TransportPacketsFeedback
and a new interface is introduced to clearly separate which field is
used. A possible followup would be to introduce a separate struct.
That would complicate the signature of ProcessTransportFeedback.

Bug: webrtc:9934
Change-Id: I2b319e4df2b557fbd4de66b812744bca7d91ca15
Reviewed-on: https://webrtc-review.googlesource.com/c/107080
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25465}
 2018-11-01 12:39:56 +00:00
Yves Gerey
988cc0870b [Cleanup] Add missing #include. Remove useless ones. 
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
 2018-10-23 11:32:56 +00:00
Sebastian Jansson
c0af56b9fb Cleanup in congestion controller. 
This CL removes some indirection and moves some constants. This
is done to simplify understanding and debugging of the code.

Bug: webrtc:9718
Change-Id: Ibe2b1da0163b4c97ffd1a5bc157f6aa59582d697
Reviewed-on: https://webrtc-review.googlesource.com/98240
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24732}
 2018-09-13 15:46:37 +00:00
Niels Möller
a12c42a6b2 Delete root header file typedef.h. 
Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24100}
 2018-07-25 14:59:26 +00:00
Bjorn Terelius
0c7ec80927 Limit BWE reductions before first measured throughput. 
After detecting overuse of the network capacity, the target
bitrate is reduced. Normally, this should happen at most once
per RTT to prevent repeated reductions from the same overuse
signal. This CL fixes a bug that allowed repeated reductions
if an overuse was detected before it had the first reliable
throughput measurement.

The fix is guarded by a field trial. To enable the fix, use
WebRTC-BweInitialBackOffInterval/Enabled-200/

Bug: webrtc:9493
Change-Id: Iae566227fd94ebb8a4449406572158a8b79d9c53
Reviewed-on: https://webrtc-review.googlesource.com/88765
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24021}
 2018-07-18 13:51:05 +00:00
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly 
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
 2018-07-05 10:59:49 +00:00
Sebastian Jansson
04b18cb365 Removes redundant delay based bwe. 
This removes the legacy DelayBasedBwe to reduce code redundancy and
avoid the risk of applying changes on only one version.

Bug: webrtc:8415
Change-Id: I88aba03adbb77ee0ff0a97a8b3be6ddf028af48a
Reviewed-on: https://webrtc-review.googlesource.com/85364
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23798}
 2018-07-02 09:11:33 +00:00
Bjorn Terelius
43d0b98fe5 Clean up RateControlInput struct, used by bandwidth estimation. 
Remove unused member noise_var from RateControlInput struct.

Rename incoming_bitrate to estimated_throughput_bps to reflect
that the AimdRateControl may be running on either the send side
or the receive side.

Bug: webrtc:9411
Change-Id: Ie1ae0c29dc9559ef389993144e69fcd41684f1a4
Reviewed-on: https://webrtc-review.googlesource.com/83728
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Anastasia Koloskova <koloskova@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23783}
 2018-06-29 10:47:37 +00:00
Danil Chapovalov
0040b66ad3 Replace rtc::Optional with absl::optional 
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script from modules with parameters
'pacing video_coding congestion_controller remote_bitrate_estimator':

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118
Reviewed-on: https://webrtc-review.googlesource.com/83900
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23640}
 2018-06-18 10:24:48 +00:00
Per Kjellander
dd3eae5f94 Revert "Configure and use max bitrate to limit the AIMD controller estimates." 
This reverts commit 18d7c7ea7e.

Reason for revert: 
This seems to cause the auto roller to Chrome to fail on Linux and Mac on the browsertest
WebRtcSimulcastBrowserTest.TestVgaReturnsTwoSimulcastStreams

https://chromium-review.googlesource.com/c/chromium/src/+/1064736


Original change's description:
> Configure and use max bitrate to limit the AIMD controller estimates.
> 
> Bug: webrtc:9275
> Change-Id: I9625cd473e1cb198abe08020f5462f1bd64bf2a5
> Reviewed-on: https://webrtc-review.googlesource.com/77081
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23287}

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

Change-Id: I8ed827ab6b2f7d2b70b9889e5a88701bfb974d35
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9275
Reviewed-on: https://webrtc-review.googlesource.com/77660
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23291}
 2018-05-18 07:12:26 +00:00
Bjorn Terelius
18d7c7ea7e Configure and use max bitrate to limit the AIMD controller estimates. 
Bug: webrtc:9275
Change-Id: I9625cd473e1cb198abe08020f5462f1bd64bf2a5
Reviewed-on: https://webrtc-review.googlesource.com/77081
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23287}
 2018-05-17 16:07:42 +00:00
Sebastian Jansson
fc7ec8e9f8 Reland "Moved congestion controller to goog_cc folder." 
This is a reland of e6cefdf9c5.

Original change's description:
> Moved congestion controller to goog_cc folder.
> 
> Bug: webrtc:8415
> Change-Id: I2070da0cacf1dbfc4b6a89285af3e68fd03497ab
> Reviewed-on: https://webrtc-review.googlesource.com/43841
> 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@{#21928}

Bug: webrtc:8415
Change-Id: Ib5cf8641466655d64ac80f720561817f4cab49a9
Reviewed-on: https://webrtc-review.googlesource.com/53062
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22244}
 2018-03-01 10:16:12 +00:00
Renamed from modules/congestion_controller/rtp/delay_based_bwe.cc (Browse further)