11200 Commits

Author SHA1 Message Date
ehmaldonado
eaaae9e91b base->rtc_base: Update .c, .mm and .java files.
TBR=kwiberg@webrtc.org
BUG=webrtc:7634

Review-Url: https://codereview.webrtc.org/2974613003
Cr-Commit-Position: refs/heads/master@{#18926}
2017-07-07 10:09:51 +00:00
ilnik
f04afde85a Report interframe delay sum in old GetStats
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2965033002
Cr-Commit-Position: refs/heads/master@{#18924}
2017-07-07 08:26:24 +00:00
jbauch
5b361730d0 Support re-entrant calls to MessageQueueManager::Clear.
BUG=webrtc:7908

Review-Url: https://codereview.webrtc.org/2968753002
Cr-Commit-Position: refs/heads/master@{#18923}
2017-07-07 06:51:37 +00:00
braveyao
4a494ffd12 desktop_capture: crop border in window_capture on Win8/10
On Windows8/10, we prefer cropping desired window out from a whole screen
capture due to some reasons. The problem is Win10 has an invisible border
around the window. If we leave the border, it will expose background a bit.

This cl is about to always remove the border of desired window on Win8/10.
This will help a lot to capturing still windows during window sharing.
This cl still can't handle the background exposure issue when you move the
target window around during capturing. More investigation is needed.

BUG=chromium:737278

Review-Url: https://codereview.webrtc.org/2973853002
Cr-Commit-Position: refs/heads/master@{#18921}
2017-07-07 03:20:27 +00:00
Edward Lemur
c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00
Sebastian Jansson
9e3f1e4ca2 Fixed a miscalculation of sent bitrate caused by mixup of time units
Bug: webrtc:7949
Change-Id: Ia57fdd3d1de0952b80e77c30b0a6cfe44515eff2
Reviewed-on: https://chromium-review.googlesource.com/561460
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18917}
2017-07-06 15:22:58 +00:00
mbonadei
d66072b71b Moving asm code out of common_audio_c sources list
BUG=webrtc:7743

Review-Url: https://codereview.webrtc.org/2966173002
Cr-Commit-Position: refs/heads/master@{#18916}
2017-07-06 14:44:14 +00:00
oprypin
3b03476233 Remove MAIN_NIB_FILE from Info.plist because the substitution is broken
BUG=webrtc:7943

Review-Url: https://codereview.webrtc.org/2965193002
Cr-Commit-Position: refs/heads/master@{#18915}
2017-07-06 14:09:57 +00:00
henrik.lundin
a44910787b Let NetEq reset the AudioFrame during muted state
In practice, this change will make AudioFrame::muted_ replicate the
explicit muted variable, passed as a pointer to NetEq::GetAudio.

BUG=webrtc:7944

Review-Url: https://codereview.webrtc.org/2965203002
Cr-Commit-Position: refs/heads/master@{#18914}
2017-07-06 12:23:53 +00:00
sprang
02569adfd4 Update screen simulcast config
Lower then bitrate so that the delta between the highest layer of the
lower stream and lowest layer of the higher stream is not too large.

BUG=webrtc:4172

This is a reland of the following CL:

Review-Url: https://codereview.webrtc.org/2791273002
Cr-Commit-Position: refs/heads/master@{#18232}
Committed: dceb42da3e

https: //codereview.webrtc.org/2883963002
Review-Url: https://codereview.webrtc.org/2966833002
Cr-Commit-Position: refs/heads/master@{#18913}
2017-07-06 12:05:50 +00:00
sprang
168794c43c Implement RTP keepalive in native stack.
BUG=webrtc:7907

Review-Url: https://codereview.webrtc.org/2960363002
Cr-Commit-Position: refs/heads/master@{#18912}
2017-07-06 11:38:06 +00:00
mbonadei
5c0d703382 Moving asm code out of isac_fix_c sources list
BUG=webrtc:7743

Review-Url: https://codereview.webrtc.org/2973613002
Cr-Commit-Position: refs/heads/master@{#18911}
2017-07-06 10:48:55 +00:00
ehmaldonado
05db21d5b3 Reland of move webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2973493002/ )
Remove webrtc/tools
https://chromium-review.googlesource.com/c/558980/ has been submitted. It should be safe to
reland now.

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
BUG=webrtc:7855

Review-Url: https://codereview.webrtc.org/2969093003
Cr-Commit-Position: refs/heads/master@{#18910}
2017-07-06 10:34:35 +00:00
ilnik
2edc6845ac Report timing frames info in GetStats.
Some frames are already marked as 'timing frames' via video-timing RTP header extension. Timestamps along full WebRTC pipeline are gathered for these frames. This CL implements reporting of these timestamps for a single
timing frame since the last GetStats(). The frame with the longest end-to-end delay between two consecutive GetStats calls is reported.

The purpose of this timing information is not to provide a realtime statistics but to provide debugging information as it will help identify problematic places in video pipeline for outliers (frames which took longest to process).

BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2946413002
Cr-Commit-Position: refs/heads/master@{#18909}
2017-07-06 10:06:50 +00:00
tommi
5b7fc8ce42 A few simplifications to CodecDatabase and VCMGenericDecoder.
* Remove the ReleaseDecoder and Release methods that were used in combination with deleting the decoder object. Now simply deleting the object does the right thing.
* Remove 'friend' relationship between the two classes since they don't need to touch each other's state directly anymore.
* Use std::unique_ptr for holding pointers and transferring ownership.

These changes were previously reviewed here:
https://codereview.webrtc.org/2764573002/

BUG=webrtc:7361, 695438

Review-Url: https://codereview.webrtc.org/2966823002
Cr-Commit-Position: refs/heads/master@{#18908}
2017-07-05 23:45:57 +00:00
bdodson
6aa95117d8 Fix null ref in NetworkMonitorAutoDetect if Connectivity Manager service is unavailable
BUG=webrtc:7917
TBR=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2963363002
Cr-Commit-Position: refs/heads/master@{#18906}
2017-07-05 16:55:09 +00:00
minyue-webrtc
b16a01f14f Revert "Reland "Adding ANA config event to debug dump.""
This reverts commit 2d54784d890be462a7fbf0fcfdc633bc4791982a.

Reason for revert: upstream conflicts

Original change's description:
> Reland "Adding ANA config event to debug dump."
> 
> Originally review in https://chromium-review.googlesource.com/c/535554/
> 
> Reverted in https://chromium-review.googlesource.com/c/539737/ due to upstreaming failure.
> 
> BUG=webrtc:7854
> 
> Change-Id: Ie4ad6ecfaf0f6b556dc662512d0be8ce94f8a4a8
> Reviewed-on: https://chromium-review.googlesource.com/541436
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18865}

TBR=minyue@webrtc.org,ossu@webrtc.org

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

Bug: webrtc:7854
Change-Id: I28841ed088664d2965454dc52196f83c9d81773e
Reviewed-on: https://chromium-review.googlesource.com/559429
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18904}
2017-07-05 14:50:32 +00:00
henrik.lundin
63d146b743 NetEq: Rectify the implementation of PacketBuffer::DiscardOldPackets
The implementation of this method did not follow the description in
the method comment. It was supposed to delete all packets in a range
[A, B], but if at least one packet in the buffer had a timestamp lower
than A, then no packets at all were discarded. This is now fixed.

BUG=webrtc:7937

Review-Url: https://codereview.webrtc.org/2969123003
Cr-Commit-Position: refs/heads/master@{#18903}
2017-07-05 14:03:34 +00:00
gnish
191113a46b Added implementation of four functions in the BBR congestion controller:
1) Function responsible for receiving feedback, digesting data and deciding switch scenarios.
2) Function which enters Startup mode.
3) Function which exits Startup mode.
4) Function which calculates, whether or not full bandwidth is reached.

BUG=webrtc:7713

Review-Url: https://codereview.webrtc.org/2924603002
Cr-Commit-Position: refs/heads/master@{#18901}
2017-07-05 12:00:46 +00:00
minyue-webrtc
fae474c9cd Implement packet discard rate in NetEq.
BUG=webrtc:7903

Change-Id: I819c9362671ca0b02c602d53e4dc39afdd8ec465
Reviewed-on: https://chromium-review.googlesource.com/555311
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18899}
2017-07-05 10:18:00 +00:00
stefan
889d9654f7 Fix issue with zero rtt reports when using FlexFEC and add perf test.
BUG=webrtc:7938

Review-Url: https://codereview.webrtc.org/2966153002
Cr-Commit-Position: refs/heads/master@{#18898}
2017-07-05 10:03:02 +00:00
henrika
070efc088e Improves WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches UMA stat
BUG=b/38018041

Review-Url: https://codereview.webrtc.org/2972743003
Cr-Commit-Position: refs/heads/master@{#18897}
2017-07-05 09:34:31 +00:00
philipel
f720704493 Added philipel@webrtc.org to webrtc/modules/remote_bitrate_estimator/OWNERS.
BUG=none
NOTRY=true

Review-Url: https://codereview.webrtc.org/2966043002
Cr-Commit-Position: refs/heads/master@{#18894}
2017-07-04 14:57:46 +00:00
henrika
cb576c50ee Fixes build issue based on usage of Android O specific API
BUG=NONE
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2967043002
Cr-Commit-Position: refs/heads/master@{#18893}
2017-07-04 14:02:35 +00:00
Gustavo Garcia
c43f68e52c Fix do not unregister bluetooth receiver if it was not registered
Bug: webrtc:7890
Change-Id: Ib46b4a4407fa030500930ed03a093b26c71f8963
Reviewed-on: https://chromium-review.googlesource.com/550617
Commit-Queue: Henrik Andreasson <henrika@webrtc.org>
Reviewed-by: Henrik Andreasson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18892}
2017-07-04 13:50:15 +00:00
magjed
cc8856c9c2 Remove unused static VideoEncoder functions
BUG=None
TBR=stefan

Review-Url: https://codereview.webrtc.org/2967853002
Cr-Commit-Position: refs/heads/master@{#18891}
2017-07-04 13:03:41 +00:00
henrika
8eadead3f4 Adds support for USB audio devices in AppRTCMobile on Android.
This change extends the definition of wired headset to also include USB
devices. The effect is that audio will now be routed to USB audio devices
when used in combination with AppRTCMobile.

BUG=webrtc:7931

Review-Url: https://codereview.webrtc.org/2971613003
Cr-Commit-Position: refs/heads/master@{#18889}
2017-07-04 12:10:48 +00:00
terelius
a9521e248e Reduce send rate to 50% if overusing before we have an acknowledged bitrate.
Check TimeToReducefurther to avoid reducing too often.

BUG=webrtc:7884

Review-Url: https://codereview.webrtc.org/2954923003
Cr-Commit-Position: refs/heads/master@{#18888}
2017-07-04 11:52:58 +00:00
peah
2c3161c86e Changed default value for the duration of the echo in echocanceller 3
BUG=webrtc:7519

Review-Url: https://codereview.webrtc.org/2971683002
Cr-Commit-Position: refs/heads/master@{#18887}
2017-07-04 11:33:11 +00:00
saza
0d7f04daa0 Reland of Add received audio/video call duration metrics based on packets.
Original issue:
https://codereview.webrtc.org/2957073002/

Reason for reland:
Failed Android unit tests and failed Windows compile.
The tests seemed related at the time, but not after more consideration.

Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.

BUG=webrtc:7882
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2970793003
Cr-Commit-Position: refs/heads/master@{#18886}
2017-07-04 11:05:06 +00:00
ehmaldonado
38fecafa48 Revert of Remove webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2970743003/ )
Reason for revert:
This should wait until https://chromium-review.googlesource.com/c/558980/ is submitted.

Original issue's description:
> Remove webrtc/tools
>
> BUG=webrtc:7855
> TBR=kwiberg@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2970743003
> Cr-Commit-Position: refs/heads/master@{#18883}
> Committed: ed56680adb

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7855

Review-Url: https://codereview.webrtc.org/2973493002
Cr-Commit-Position: refs/heads/master@{#18885}
2017-07-04 09:02:49 +00:00
peah
d3588cfb31 Improved low-level echo handling in echo canceller 3
This CL addresses the issue of echo leakage of low level
echoes by making the echo canceller more restrictive for
that scenario.

BUG=webrtc:7930

Review-Url: https://codereview.webrtc.org/2969943002
Cr-Commit-Position: refs/heads/master@{#18884}
2017-07-04 08:54:37 +00:00
ehmaldonado
ed56680adb Remove webrtc/tools
BUG=webrtc:7855
TBR=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2970743003
Cr-Commit-Position: refs/heads/master@{#18883}
2017-07-04 08:34:47 +00:00
saza
382f21cd9c Revert of Add received audio and video call duration metrics based on packets. (patchset #4 id:140001 of https://codereview.webrtc.org/2957073002/ )
Reason for revert:
The following, seemingly related, unit tests crash on Android32 (M Nexus5X).
org.webrtc.PeerConnectionTest#testCompleteSession
org.webrtc.PeerConnectionTest#testDataChannelOnlySession

A Windows build fails with a mysterious compile error.

Original issue's description:
> Add received audio/video call duration metrics based on packets.
>
> Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.
>
> BUG=webrtc:7882
>
> Review-Url: https://codereview.webrtc.org/2957073002
> Cr-Commit-Position: refs/heads/master@{#18881}
> Committed: 746749237a

TBR=stefan@webrtc.org,aleloi@webrtc.org,asapersson@webrtc.org,holmer@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7882

Review-Url: https://codereview.webrtc.org/2972613002
Cr-Commit-Position: refs/heads/master@{#18882}
2017-07-04 08:11:49 +00:00
saza
746749237a Add received audio/video call duration metrics based on packets.
Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.

BUG=webrtc:7882

Review-Url: https://codereview.webrtc.org/2957073002
Cr-Commit-Position: refs/heads/master@{#18881}
2017-07-04 07:19:22 +00:00
eladalon
2a2b297aa6 Add underscore at end of Call members' names
BUG=None

Review-Url: https://codereview.webrtc.org/2971583002
Cr-Commit-Position: refs/heads/master@{#18880}
2017-07-03 16:25:27 +00:00
peah
4235d78b57 Disabling flaky complexity tests for the audio processing module.
The complexity test for the audio processing module have long proven
to give false alarms of complexity regressions for which no related
changes can be identified. Attempts to address that has improved the
that, but the tests do still give false alarms.

This CL deactivates the complexity tests until a better way of
testing this is available.

BUG=chromium:713507, webrtc:5846,webrtc:6685,webrtc:7712

Review-Url: https://codereview.webrtc.org/2897403006
Cr-Commit-Position: refs/heads/master@{#18879}
2017-07-03 16:11:22 +00:00
eladalon
7ab7fd66c4 Fix gmock warnings emanating from FlexfecReceiveStreamTest
BUG=None

Review-Url: https://codereview.webrtc.org/2966963002
Cr-Commit-Position: refs/heads/master@{#18878}
2017-07-03 13:57:13 +00:00
brandtr
7c7796b8ec Register FlexFEC SSRC to receive RTCP on sending side.
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2965883002
Cr-Commit-Position: refs/heads/master@{#18877}
2017-07-03 13:02:53 +00:00
Alex Loiko
48587f91f8 Changing AudioConferenceMixer logging to base/logging.h
We'd like to remove all occurrences of WEBRTC_TRACE and delete the
macro! One logging mechanism is enough.

AudioConferenceMixer is scheduled for removal and is one of the 
things tracked by bugs.webrtc.org/4690. The logging is changed to not
block webrtc:5118

NOTRY=True

Bug: webrtc:5118
Change-Id: Ibad1ae45e8af1ba5bbe253d4c693ecf9e7c422ac
Reviewed-on: https://chromium-review.googlesource.com/518172
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18876}
2017-07-03 12:35:46 +00:00
ilnik
4257ab2e02 Add received interframe delay UMA metrics
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2966733002
Cr-Commit-Position: refs/heads/master@{#18875}
2017-07-03 08:15:58 +00:00
jansson
ad515c459b fix comment length
BUG=NONE
NOTRY=True

Review-Url: https://codereview.webrtc.org/2966743002
Cr-Commit-Position: refs/heads/master@{#18874}
2017-07-03 07:47:44 +00:00
Henrik Kjellander
a80c16a67c Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
This reverts commit c3771cc4d37f5573fe53b7c7cff295a4f0f9560f.
(breaks downstream internal project)

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2972463002 .
Cr-Commit-Position: refs/heads/master@{#18873}
2017-07-01 14:48:18 +00:00
Henrik Kjellander
dca1e09db7 Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"
This reverts commit c8fa692ec44fd6ba4fa3d085ac3161a262fc18c5.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2964773002 .
Cr-Commit-Position: refs/heads/master@{#18872}
2017-07-01 14:42:25 +00:00
kjellander
c8fa692ec4 Update includes for webrtc/{base => rtc_base} rename (1/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
2017-06-30 21:02:00 +00:00
kjellander
c3771cc4d3 Update includes for webrtc/{base => rtc_base} rename (2/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.

Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
2017-06-30 20:42:44 +00:00
sprang
89c4a7e57d Wire up experiment for improved screenshare bwe.
Also adds some full stack test variants with the experiment enabled.

BUG=webrtc:7694

Review-Url: https://codereview.webrtc.org/2949553002
Cr-Commit-Position: refs/heads/master@{#18869}
2017-06-30 20:27:40 +00:00
kjellander
e96c45b662 Reland "Update includes for webrtc/{base => rtc_base} rename (3/3)"
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

Changes since last attempt: Some system headers were moved back to their original location since on Windows compilation breaks otherwise.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
TBR=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2966523003
Cr-Commit-Position: refs/heads/master@{#18868}
2017-06-30 17:45:21 +00:00
Magnus Jedvert
224e65939a Reland of "VideoFrameBuffer: Remove deprecated functions"
This reverts commit f1e34832b84798d7665d2aad9a5b3f33cbe5a274.

Reason for reland: Chomium code has been updated.

Original change's description:
> Revert "VideoFrameBuffer: Remove deprecated functions"
> 
> This reverts commit 428c9e218538278e6b0db42d1b734431bb432e1a.
> 
> Reason for revert: Breaks Chromium WebRTC FYI on Mac Builder. http://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/25788
> 
> Original change's description:
> > VideoFrameBuffer: Remove deprecated functions
> > 
> > Bug: webrtc:7632
> > Change-Id: I06f97bacd51f94d1f90b5286cc39e06a1697bb9b
> > Reviewed-on: https://chromium-review.googlesource.com/535479
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#18832}
> 
> TBR=magjed@webrtc.org,nisse@webrtc.org
> 
> Change-Id: I2e6617420746bba3e4637019d3bce03be12a4643
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7632
> Reviewed-on: https://chromium-review.googlesource.com/555550
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18834}

TBR=magjed@webrtc.org,nisse@webrtc.org

Change-Id: I41c7b31ab52ba162fd0a9ab03a4b45aecb97cb09
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7632
Reviewed-on: https://chromium-review.googlesource.com/558244
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18867}
2017-06-30 16:39:27 +00:00
terelius
e75d96b5bd Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ )
Reason for revert:
Resetting the estimate means that we need to start gathering data from scratch again. The combination of
1) DelayBasedEstimator not reacting to overuse unless there is a valid estimate of the acknowledged bitrate, and
2) AcknowledgedBitrateEstimator needing a significant amount of time/data to obtain an provide an estimate
causes poor performance in simulations/tests. It is not clear whether this will affect real networks negatively, but I suggest reverting this to be on the safe side.
See also https://bugs.chromium.org/p/webrtc/issues/detail?id=7884

Original issue's description:
> Test and fix for huge bwe drop after alr state.
>
> BUG=webrtc:7746
>
> Review-Url: https://codereview.webrtc.org/2931873002
> Cr-Commit-Position: refs/heads/master@{#18692}
> Committed: 37aa8ba616

TBR=solenberg@webrtc.org,kwiberg@webrtc.org,minyue@webrtc.org,holmer@chromium.org,philipel@webrtc.org,oprypin@webrtc.org,holmer@google.com,stefan@webrtc.org,tschumim@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7746

Review-Url: https://codereview.webrtc.org/2964213002
Cr-Commit-Position: refs/heads/master@{#18866}
2017-06-30 15:11:44 +00:00