fd1e9d1af4
[Stats] Add minimum RTCReceivedRtpStreamStats with jitter and packetsLost
...
Spec: https://www.w3.org/TR/webrtc-stats/#receivedrtpstats-dict *
According to the spec, |RTCReceivedRtpStreamStats| is the base class for |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats|. This structure isn't visible in JavaScript but it's important to bring it up to spec for the C++ part. This CL adds the barebone |RTCReceivedRtpStreamStats| with a bunch of TODOs for later migrations.
This commit makes the minimum |RTCReceivedRtpStreamStats| and rebase |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats| to use the new class as the parent class.
This commit also moves |jitter| and |packets_lost| to |RTCReceivedRtpStreamStats|, from |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats|. Moving these two first because they are the two that exist in both subclasses for now.
Bug: webrtc:12532
Change-Id: I0ec74fd241f16c1e1a6498b6baa621ca0489f279
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210340
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33435}
2021-03-11 11:58:58 +00:00
8639673f0c
Parse encoded frame QP if not provided by encoder
...
Bug: webrtc:12542
Change-Id: Ic70b46e226f158db7a478a9f20e1f940804febba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210966
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33434}
2021-03-11 11:48:38 +00:00
8097935df3
Revert "Reduce complexity in the APM pipeline when the output is not used"
...
This reverts commit aa6adffba325f4b698a1e94aeab020bfdc47adec.
Reason for revert: breaks webrtc-importer
Original change's description:
> Reduce complexity in the APM pipeline when the output is not used
>
> This CL selectively turns off parts of the audio processing when
> the output of APM is not used. The parts turned off are such that
> don't need to continuously need to be trained, but rather can be
> temporarily deactivated.
>
> The purpose of this CL is to allow CPU to be reduced when the
> client is muted.
>
> The CL will be follow by additional CLs, adding similar functionality
> in the echo canceller and the noiser suppressor
>
> Bug: b/177830919
> Change-Id: I72d24505197a53872562c0955f3e7b670c43df6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209703
> Commit-Queue: Per Åhgren <peah@webrtc.org >
> Reviewed-by: Sam Zackrisson <saza@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#33431}
Bug: b/177830919
Change-Id: I937cd61dedcd43150933eb1b9d65aebe68401e91
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211348
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33433}
2021-03-11 11:41:20 +00:00
be140b4187
Change ObjCNetworkMonitor::OnPathUpdate to use PostTask
...
Removes use of AsyncInvoker, replaced with PendingTaskSafetyFlag. The
latter is extended to support creation on a different thread than
where it will be used, and to support stop and restart.
Bug: webrtc:12339
Change-Id: I28b6e09b1542f50037e842ef5fe3a47d15704b46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211002
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Tommi <tommi@webrtc.org >
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33432}
2021-03-11 11:10:18 +00:00
aa6adffba3
Reduce complexity in the APM pipeline when the output is not used
...
This CL selectively turns off parts of the audio processing when
the output of APM is not used. The parts turned off are such that
don't need to continuously need to be trained, but rather can be
temporarily deactivated.
The purpose of this CL is to allow CPU to be reduced when the
client is muted.
The CL will be follow by additional CLs, adding similar functionality
in the echo canceller and the noiser suppressor
Bug: b/177830919
Change-Id: I72d24505197a53872562c0955f3e7b670c43df6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209703
Commit-Queue: Per Åhgren <peah@webrtc.org >
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33431}
2021-03-11 10:06:58 +00:00
54dbc3be3f
Revert "[Battery]: Delay start of TaskQueuePacedSender."
...
This reverts commit 89cb65ed663a9000b9f7c90a78039bd85731e9ae.
Reason for revert: Breaks downstream project
Original change's description:
> [Battery]: Delay start of TaskQueuePacedSender.
>
> To avoid unnecessary repeating tasks, TaskQueuePacedSender is started
> only upon RtpTransportControllerSend::EnsureStarted().
>
> More specifically, the repeating task happens in
> TaskQueuePacedSender::MaybeProcessPackets() every 500ms, using a self
> task_queue_.PostDelayedTask().
>
> Bug: chromium:1152887
> Change-Id: I72c96d2c4b491d5edb45a30b210b3797165cbf48
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208560
> Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org >
> Reviewed-by: Henrik Boström <hbos@webrtc.org >
> Reviewed-by: Erik Språng <sprang@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#33421}
Bug: chromium:1152887
Change-Id: I781d3bf614d5d0c03f292c0e478f24ede91624bb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211345
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33430}
2021-03-11 09:01:01 +00:00
3135772326
Changed setActive of RTCAudio Session, and it's working
...
Bug: webrtc:12018
Change-Id: I7ee5cf2df406e7a6d0edf1a95a3665c4b1d6958b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210720
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org >
Commit-Queue: Abby Yeh <abbyyeh@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33429}
2021-03-11 05:57:25 +00:00
fbb2dcfe38
Roll chromium_revision 43d5378f7f..fc9c86fd36 (861531:861807)
...
Change log: 43d5378f7f..fc9c86fd3643d5378f7f..fc9c86fd36a90f1db464..948e8c0a7cdb920d7bb8..1ed0793ff4de840b0cc0..ce92af86b1cb7510fb0e..5c0035ad66fc08a988c1..2f9fd21021https://chromium.googlesource.com/catapult.git/+log/fa0ace78a3..7a3a7363a7
* src/third_party/depot_tools: 7e3ad9eeb8..c2c576e94037518c39e3..c600d9d76c6e15c0788b..03d5140c9443d5378f7f..fc9c86fd36TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I10ff282bd54dc91ffeb06d999713b4c6ad6dc40c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211420
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#33428}
2021-03-11 03:23:53 +00:00
b2e71b8b35
Reland "Fix race between destroying SctpTransport and receiving notification on timer thread."
...
This reverts commit 8a38b1cf681cd77f0d59a68fb45d8dedbd7d4cee.
Reason for reland: Problem was identified; has something to do with
the unique_ptr with the custom deleter.
Original change's description:
> Revert "Fix race between destroying SctpTransport and receiving notification on timer thread."
>
> This reverts commit a88fe7be146b9b85575504d4d5193c007f2e3de4.
>
> Reason for revert: Breaks downstream test, still investigating.
>
> Original change's description:
> > Fix race between destroying SctpTransport and receiving notification on timer thread.
> >
> > This gets rid of the SctpTransportMap::Retrieve method and forces
> > everything to go through PostToTransportThread, which behaves safely
> > with relation to the transport's destruction.
> >
> > Bug: webrtc:12467
> > Change-Id: Id4a723c2c985be2a368d2cc5c5e62deb04c509ab
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208800
> > Reviewed-by: Niels Moller <nisse@webrtc.org >
> > Commit-Queue: Taylor <deadbeef@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#33364}
>
> TBR=nisse@webrtc.org
>
> Bug: webrtc:12467
> Change-Id: Ib5d815a2cbca4feb25f360bff7ed62c02d1910a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209820
> Reviewed-by: Taylor <deadbeef@webrtc.org >
> Commit-Queue: Taylor <deadbeef@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#33386}
Bug: webrtc:12467
Change-Id: I5f9fcd6df7a211e6edfa64577fc953833f4d9b79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210040
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Florent Castelli <orphis@webrtc.org >
Commit-Queue: Taylor <deadbeef@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33427}
2021-03-11 00:58:33 +00:00
34fdc92119
Add audioproc_f support for testing the runtime settings of whether the output is used
...
Bug: b/177830919
Change-Id: Iddcb79000f471eac165e3f44f14fad41435e6ccb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211241
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Commit-Queue: Per Åhgren <peah@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33426}
2021-03-10 23:19:30 +00:00
3d792e990a
Add controls for MTU size of virtual socket server
...
This will allow us to write tests for what happens when large datagrams
are discarded.
Bug: webrtc:12495
Change-Id: I5c8e9fe55917967ea4604e1b3abe3c590c330ffa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211044
Reviewed-by: Taylor <deadbeef@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33425}
2021-03-10 20:06:22 +00:00
bccfd26322
Allow webrtc mac cross compile
...
Chromium side changes: https://crrev.com/c/2743475
Bug: b/180976558
Change-Id: I4db49b91379d8b7f293147bdefb4296861faf0a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210684
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33424}
2021-03-10 18:42:58 +00:00
048adc7136
Add missing remote-outbound stats to RTCPReceiver::NTP
...
In order to add `RTCRemoteOutboundRtpStreamStats` (see [1]), the
following stats must be added:
- sender's packet count (see [2])
- sender's octet count (see [2])
- total number of RTCP SR blocks sent (see [3])
[1] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats
[2] https://tools.ietf.org/html/rfc3550#section-6.4.1
[3] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-reportssent
Bug: webrtc:12529
Change-Id: I47ac2f79ba53631965d1cd7c1062f3d0f158d66e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210963
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33423}
2021-03-10 16:36:48 +00:00
da2fd2a2b2
Fix problem with ipv4 over ipv6 on Android
...
This patch fixes a problem with using ipv4 over ipv6
addresses on Android. These addresses are discovered
using 'getifaddr' with interfaces called 'v4-wlan0' or
'v4-rmnet' but the Android API does not report them.
This leads to failure when BasicPortAllocator tries
to bind a socket to the ip-address, making the ipv4
address unusable.
This solution does the following
1) Insert BasicNetworkManager as NetworkBinderInterface
rather than AndroidNetworkManager.
2) When SocketServer calls BindSocketToNetwork,
BasicNetworkManager first lookup the interface name,
and then calls AndroidNetworkManager.
3) AndroidNetworkManager will then first try to bind
using the known ip-addresses, and if it can't find the network
it will instead match the interface names.
The patch has been tested on real android devices, and works fine.
And everything is disabled by default, and is enabled by field trial.
My plan is to rollout the feature, checking that it does not introduce
any problems, and if so, enabled for all.
Bug: webrtc:10707
Change-Id: I7081ba43d4ce17077acfa5fbab44eda127ac3971
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211003
Commit-Queue: Jonas Oreland <jonaso@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33422}
2021-03-10 16:08:18 +00:00
89cb65ed66
[Battery]: Delay start of TaskQueuePacedSender.
...
To avoid unnecessary repeating tasks, TaskQueuePacedSender is started
only upon RtpTransportControllerSend::EnsureStarted().
More specifically, the repeating task happens in
TaskQueuePacedSender::MaybeProcessPackets() every 500ms, using a self
task_queue_.PostDelayedTask().
Bug: chromium:1152887
Change-Id: I72c96d2c4b491d5edb45a30b210b3797165cbf48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208560
Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33421}
2021-03-10 16:03:28 +00:00
79011ef4a7
Remove ModuleRtpRtcpImpl2::LastReceivedNTP
...
`LastReceivedNTP()` does not need to be part of the public members of
`ModuleRtpRtcpImpl` and `ModuleRtpRtcpImpl2` since it is used only
once in the same class.
This change is requried by the child CL [1] which adds a public getter
needed to add remote-outbound stats.
[1] https://webrtc-review.googlesource.com/c/src/+/211041
Bug: webrtc:12529
Change-Id: I82cfea5ee795de37fffa3d759ce9f581ca775d55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211043
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33420}
2021-03-10 15:11:38 +00:00
ee8cd20ec5
Add a mutex free implementation of webrtc::ReceiveStatistics
...
The mutex is removed from the old existing implementation and instead a wrapper is implemented that ensure thread-safety.
Both the thread-safe and unsafe version share the same implementation of the logic.
There are two ways of construction:
webrtc::ReceiveStatistics::Create - thread-safe version.
webrtc::ReceiveStatistics::CreateUnLocked -thread-unsafe
Bug: none
Change-Id: Ica375919fda70180335c8f9ea666497811daf866
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211240
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Per Kjellander <perkj@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33419}
2021-03-10 14:16:38 +00:00
bb22066e60
Roll chromium_revision e1b9354ff4..43d5378f7f (854007:861531)
...
Change log: e1b9354ff4..43d5378f7fe1b9354ff4..43d5378f7f242517a010..a90f1db464251bfc5101..db920d7bb8228b1c1883..368c7dd2c9196ba1aaa8..c93be42a40d999d54f4b..cc80b4ac98d365fb8897..de840b0cc07dee089ce5..cb7510fb0ed15b5c89b3..fc08a988c1https://boringssl.googlesource.com/boringssl.git/+log/ce9b002ebd..dfe0b01b3e
* src/third_party/breakpad/breakpad: e3d485f73f..d6a6f52606https://chromium.googlesource.com/catapult.git/+log/35976bfcac..fa0ace78a3
* src/third_party/depot_tools: 7386a1e04f..7e3ad9eeb87849316ccf..85168499771b0cdaae57..07f486922153806e5b83..7686ff854b12825ed8aa..d879aac971https://aomedia.googlesource.com/aom.git/+log/e72d81ffbc..fb9f22ce8c
* src/third_party/libyuv: 93b1b332cd..a8c181050c7945040041..37518c39e3a6647318b5..991335be3dbd77bcca0d..6e15c0788be1b9354ff4..43d5378f7fe1b9354ff4..43d5378f7fTBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: Icc9f6934589bc950215d6166f1ae899d2c01761b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211280
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#33418}
2021-03-10 13:14:38 +00:00
213dc2cfc5
Temporarily disable Opus decode test.
...
Bug: webrtc:12518, webrtc:12543
Change-Id: I5481ee96fe2a3f9fd549e17cd9424441223a8b63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211245
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33417}
2021-03-10 12:47:18 +00:00
8bfa2756a5
Fix nullability of completion handlers in iOS SDK.
...
Bug: None
Change-Id: I74d3d976760fd620a8f749a3c187430dbe80ef57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210961
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com >
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33416}
2021-03-10 11:48:04 +00:00
89127190ce
Revert "Roll chromium_revision e1b9354ff4..74090df66c (854007:861144)"
...
This reverts commit cb7ff134321aef548034123339ec45c9c1cd0bfe.
Reason for revert: iOS 12 bot fails consistently: https://ci.chromium.org/ui/p/webrtc/builders/ci/iOS64%20Sim%20Debug%20(iOS%2012)/5497/overview
Original change's description:
> Roll chromium_revision e1b9354ff4..74090df66c (854007:861144)
>
> Change log: e1b9354ff4..74090df66ce1b9354ff4..74090df66c242517a010..b64741b2d4251bfc5101..6e85f7f8aa228b1c1883..368c7dd2c9196ba1aaa8..c93be42a40d999d54f4b..cc80b4ac98d365fb8897..171bf5f5ff7dee089ce5..f8f63f141ad15b5c89b3..68e9de796chttps://boringssl.googlesource.com/boringssl.git/+log/ce9b002ebd..dfe0b01b3e
> * src/third_party/breakpad/breakpad: e3d485f73f..d6a6f52606https://chromium.googlesource.com/catapult.git/+log/35976bfcac..4f82770e78
> * src/third_party/depot_tools: 7386a1e04f..d071f2a8a97849316ccf..fedd945c3e1b0cdaae57..07f486922153806e5b83..7686ff854b12825ed8aa..e6379c8802https://aomedia.googlesource.com/aom.git/+log/e72d81ffbc..fb9f22ce8c
> * src/third_party/libyuv: 93b1b332cd..a8c181050c7945040041..37518c39e3a6647318b5..6e00fe20dfbd77bcca0d..6cf14f8442e1b9354ff4..74090df66ce1b9354ff4..74090df66cTBR=titovartem@webrtc.org ,
> BUG=None
>
> Change-Id: Id2b5745ddc858a9cc4aa0ee7c4009fff62fc7fca
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210962
> Reviewed-by: Artem Titov <titovartem@webrtc.org >
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
> Commit-Queue: Artem Titov <titovartem@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#33407}
Bug: None
Change-Id: I7679a2545ac6996df641716d3db9af0143479d33
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211242
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33415}
2021-03-10 10:14:26 +00:00
6adb8d9282
Revert "Roll chromium_revision 74090df66c..f86a579769 (861144:861254)"
...
This reverts commit 14a626af56675c9d95f9fb207d53366a53a95fbd.
Reason for revert: iOS 12 bot fails consistently: https://ci.chromium.org/ui/p/webrtc/builders/ci/iOS64%20Sim%20Debug%20(iOS%2012)/5497/overview
Original change's description:
> Roll chromium_revision 74090df66c..f86a579769 (861144:861254)
>
> Change log: 74090df66c..f86a57976974090df66c..f86a579769b64741b2d4..7d68e0b0d06e85f7f8aa..6936526975171bf5f5ff..6a6c1b635af8f63f141a..6c578c1c0568e9de796c..4b716109f36cf14f8442..c6fd42a43074090df66c..f86a579769TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
> BUG=None
>
> Change-Id: Id07a53e236cf61a9cd68fa785c467bfd377f0228
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211101
> Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
> Cr-Commit-Position: refs/heads/master@{#33409}
Bug: None
Change-Id: Ie9600c1f91c3bb65f2cd1d4e08a1ee561fa31eb6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211243
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33414}
2021-03-10 10:11:56 +00:00
734ae5296b
Revert "Roll chromium_revision f86a579769..dc3e6d8b69 (861254:861387)"
...
This reverts commit 8114016f5919a1471a165073f2fe8d2df7a8b5dc.
Reason for revert: iOS 12 bot fails consistently: https://ci.chromium.org/ui/p/webrtc/builders/ci/iOS64%20Sim%20Debug%20(iOS%2012)/5497/overview
Original change's description:
> Roll chromium_revision f86a579769..dc3e6d8b69 (861254:861387)
>
> Change log: f86a579769..dc3e6d8b69f86a579769..dc3e6d8b697d68e0b0d0..9978a5f5776936526975..389d3f34d26a6c1b635a..2202f960816c578c1c05..ce9ed2c0174b716109f3..a6f935173bd071f2a8a9..0e8b3107e0e6379c8802..d879aac971c6fd42a430..4542cb38c4f86a579769..dc3e6d8b69TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
> BUG=None
>
> Change-Id: I6c9f488a7a8e013da069dad6f60f9abb816b176d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211160
> Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
> Cr-Commit-Position: refs/heads/master@{#33410}
Bug: None
Change-Id: Ic87a952750b25d1300c596c87a4238afe8664cee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211244
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33413}
2021-03-10 10:10:16 +00:00
6a55e7307b
Stop inheriting from has_slots in DtlsTransport.
...
Bug: webrtc:11943
Change-Id: Icb6cac30ba8a10528d75e87d9ccc3508d1cfb5a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208940
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33412}
2021-03-10 08:40:14 +00:00
68ef4e5b00
Update WebRTC code version (2021-03-10T04:02:37).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: I2418fb1a561d64d8a57c36f69fe2080f19e243e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211181
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#33411}
2021-03-10 06:10:44 +00:00
8114016f59
Roll chromium_revision f86a579769..dc3e6d8b69 (861254:861387)
...
Change log: f86a579769..dc3e6d8b69f86a579769..dc3e6d8b697d68e0b0d0..9978a5f5776936526975..389d3f34d26a6c1b635a..2202f960816c578c1c05..ce9ed2c0174b716109f3..a6f935173bd071f2a8a9..0e8b3107e0e6379c8802..d879aac971c6fd42a430..4542cb38c4f86a579769..dc3e6d8b69TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I6c9f488a7a8e013da069dad6f60f9abb816b176d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211160
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#33410}
2021-03-10 02:49:11 +00:00
14a626af56
Roll chromium_revision 74090df66c..f86a579769 (861144:861254)
...
Change log: 74090df66c..f86a57976974090df66c..f86a579769b64741b2d4..7d68e0b0d06e85f7f8aa..6936526975171bf5f5ff..6a6c1b635af8f63f141a..6c578c1c0568e9de796c..4b716109f36cf14f8442..c6fd42a43074090df66c..f86a579769TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: Id07a53e236cf61a9cd68fa785c467bfd377f0228
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211101
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#33409}
2021-03-09 21:52:43 +00:00
5265b9367a
Add build-id to libjingle_peerconnection_so.so
...
This is required by Firebase Crashlytics:
https://firebase.google.com/docs/crashlytics/ndk-reports#enable-symbol-uploading
Bug: None
Change-Id: Ie0d2c2e92477df78b26b7e1fc2273589b71efa81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210965
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org >
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33408}
2021-03-09 20:21:23 +00:00
cb7ff13432
Roll chromium_revision e1b9354ff4..74090df66c (854007:861144)
...
Change log: e1b9354ff4..74090df66ce1b9354ff4..74090df66c242517a010..b64741b2d4251bfc5101..6e85f7f8aa228b1c1883..368c7dd2c9196ba1aaa8..c93be42a40d999d54f4b..cc80b4ac98d365fb8897..171bf5f5ff7dee089ce5..f8f63f141ad15b5c89b3..68e9de796chttps://boringssl.googlesource.com/boringssl.git/+log/ce9b002ebd..dfe0b01b3e
* src/third_party/breakpad/breakpad: e3d485f73f..d6a6f52606https://chromium.googlesource.com/catapult.git/+log/35976bfcac..4f82770e78
* src/third_party/depot_tools: 7386a1e04f..d071f2a8a97849316ccf..fedd945c3e1b0cdaae57..07f486922153806e5b83..7686ff854b12825ed8aa..e6379c8802https://aomedia.googlesource.com/aom.git/+log/e72d81ffbc..fb9f22ce8c
* src/third_party/libyuv: 93b1b332cd..a8c181050c7945040041..37518c39e3a6647318b5..6e00fe20dfbd77bcca0d..6cf14f8442e1b9354ff4..74090df66ce1b9354ff4..74090df66cTBR=titovartem@webrtc.org ,
BUG=None
Change-Id: Id2b5745ddc858a9cc4aa0ee7c4009fff62fc7fca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210962
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33407}
2021-03-09 17:09:11 +00:00
dc08aea610
Fix chromium roll into WebRTC
...
Add missing org_robolectric_shadows_multidex dependency as it is done
in chromium DEPS file.
Tried on: https://webrtc-review.googlesource.com/c/src/+/211004
Bug: chromium:1184780
Change-Id: Ib8f9e794b36c8da008672d989271f368df9203df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211040
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33406}
2021-03-09 14:35:39 +00:00
c84a887b8a
Allow port 80 for TURN servers
...
Bug: webrtc:12537
Change-Id: Id6a419b4b435284f743314956945783d4db94a7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210960
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Jonas Oreland <jonaso@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33405}
2021-03-09 09:58:58 +00:00
668dbf66ce
[Stats] Populate "frames" stats for video source.
...
Spec: https://www.w3.org/TR/webrtc-stats/#dom-rtcvideosourcestats-frames
Wiring up the "frames" stats with the cumulative fps counter on the video source.
Tests:
./out/Default/peerconnection_unittests
./out/Default/video_engine_tests
Bug: webrtc:12499
Change-Id: I4103f56ed04cb464f5f7e70fbf2d77c25a867a68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208782
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33404}
2021-03-09 08:54:38 +00:00
5ab9b32ee6
Update WebRTC code version (2021-03-09T04:03:43).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: I8915d8f307b09348e8879027d47a2a546103db54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210920
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#33403}
2021-03-09 06:13:31 +00:00
92d12707e0
Expose PeerConnection.restartIce in iOS SDK.
...
Bug: None
Change-Id: I76b95b3182e6b384fd68aecf4210c23459f76d2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209709
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com >
Cr-Commit-Position: refs/heads/master@{#33402}
2021-03-08 17:38:54 +00:00
d67253532f
Expose parameterless setLocalDescription() in iOS SDK.
...
Parameterless setLocalDescription is used to implement perfect
negotiation algorithm.
Bug: None
Change-Id: I89c39ee175fec5b09d9ca1700ef682e3cf20fe94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209700
Commit-Queue: Kári Helgason <kthelgason@webrtc.org >
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33401}
2021-03-08 16:39:54 +00:00
f00cd533ae
Do more actions on SDP fuzzing.
...
This will wait for completion of the setRemoteDescription() call,
and if the setRemoteDescription() is successful, it will do a
setLocalDescription() (default description).
Bug: none
Change-Id: Id1cb6b1ecbdc90d4f2c5b46a7f8e92b7491ff401
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210682
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33400}
2021-03-08 12:18:10 +00:00
a113d231a6
srtp: use srtp_crypto_policy_set_from_profile_for_* from libsrtp
...
use the helper functions
srtp_crypto_policy_set_from_profile_for_rtp
and
srtp_crypto_policy_set_from_profile_for_rtcp
provided by libsrtp to initialize the rtp and rtcp policies.
BUG=None
Change-Id: Ib1560c0fc1c06d9e79c1f871b028555b3b4d66d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208480
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
Cr-Commit-Position: refs/heads/master@{#33399}
2021-03-08 10:41:29 +00:00
c81665cf9d
Change AndroidNetworkMonitor::NotifyConnectionTypeChanged to use Invoke
...
This is consistent with other Notify methods in this class, which
handle callbacks from java using blocking invokes to the network
thread.
This eliminates the use of the deprecated AsyncInvoker class.
Bug: webrtc:12339
Change-Id: Ib2d19b37b8f669df5b97e89d720f6eb6fc9e5517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209181
Reviewed-by: Taylor <deadbeef@webrtc.org >
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33398}
2021-03-08 09:04:29 +00:00
82a94125a4
Reland "Add a fuzzer test that tries to connect a PeerConnection."
...
This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38.
Reason for revert: Added Chromium compile guards
Original change's description:
> Revert "Add a fuzzer test that tries to connect a PeerConnection."
>
> This reverts commit c67b77eee4b08c05638a219723a9141a65015da4.
>
> Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll.
>
> Original change's description:
> > Add a fuzzer test that tries to connect a PeerConnection.
> >
> > Bug: none
> > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org >
> > Reviewed-by: Henrik Boström <hbos@webrtc.org >
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#33369}
>
> NOPRESUBMIT=true
>
> Bug: none
> Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640
> Commit-Queue: Evan Shrubsole <eshr@google.com >
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#33380}
Bug: none
Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 08:58:09 +00:00
be66d95ab7
srtp: document rationale for srtp overhead calculation
...
documents why it is safe to not follow libsrtp's advice
to ensure additional SRTP_MAX_TRAILER_LEN bytes are available
when calling srtp_protect (and similar srtcp functions).
BUG=None
Change-Id: I504645d21553160f06133fd8bb3ee79e178247da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209064
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
Cr-Commit-Position: refs/heads/master@{#33396}
2021-03-08 08:50:09 +00:00
456a2642d3
Update WebRTC code version (2021-03-08T04:01:44).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: Ia9c79e506989ca773c47399b3f1eca56875b096c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210640
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#33395}
2021-03-08 05:59:19 +00:00
aedb0f0c66
Update WebRTC code version (2021-03-07T04:02:41).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: I1f4e7859d1fcca9154840ad72bac68e152456ef9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210380
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#33394}
2021-03-07 05:50:46 +00:00
eb449a979b
Revert "Reland "Enable quality scaling when allowed""
...
This reverts commit 83be84bb74133343358bba22e4e5106ecc385721.
Reason for revert: Suspect of crbug.com/1185276
Original change's description:
> Reland "Enable quality scaling when allowed"
>
> This reverts commit 609b524dd3ff36719b5c4470b85d37dcdadfb1f8.
>
> Reason for revert: Disable QualityScalingAllowed_QualityScalingEnabled on iOS.
>
> Original change's description:
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020
> Bug: webrtc:12511
> Change-Id: Ia0923e5a62acdfdeb06f9aad5d670be8a0f8d746
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209643
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
> Reviewed-by: Åsa Persson <asapersson@webrtc.org >
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#33385}
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I7004014c5936176f8c125aeb55da91ce095b266e
TBR: ssilkin@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209708
Reviewed-by: Guido Urdaneta <guidou@webrtc.org >
Commit-Queue: Guido Urdaneta <guidou@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33393}
2021-03-06 09:40:50 +00:00
77f97ec066
Update WebRTC code version (2021-03-06T04:02:56).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: Ied0c453f05c0d2d172661b712d63660ce1227988
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210120
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#33392}
2021-03-06 05:40:47 +00:00
964a88fce1
Prevent possible out-of-bounds access
...
It read from simulcast_layers[0] even if vector is empty.
Bug: none
Change-Id: I293890feda70022beae4247ab10cf8182b4cf4a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209706
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33391}
2021-03-05 17:48:48 +00:00
4a4273bf05
VP9 ResolutionBitrateLimits: If bitrates are configured, use intersection.
...
Bug: none
Change-Id: I58ada41d7a196837b35df4cf61f7e7561998cf13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209704
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org >
Commit-Queue: Åsa Persson <asapersson@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33390}
2021-03-05 17:39:04 +00:00
a86b29be01
Add VP9-specific default resolution bitrate limits
...
Bug: none
Change-Id: Ifb6f962f04b1f05d20f80a721b1f41904e0a7e99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209702
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33389}
2021-03-05 14:28:14 +00:00
1a89bc8b62
build: improve rtc_include_tests documentation
...
"including" them in the standalone checkout is ambigious,
this might mean not checking them out from git.
BUG=None
NOTRY=true
Change-Id: I53334a718c5535f63faa87d9e0fe6575012e0d2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209223
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
Cr-Commit-Position: refs/heads/master@{#33388}
2021-03-05 13:54:20 +00:00
1413e2db4c
Update WebRTC code version (2021-03-05T04:03:29).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: I8fbcbf192da90593ad03fc8da0392957d33753ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209861
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#33387}
2021-03-05 05:57:29 +00:00
8a38b1cf68
Revert "Fix race between destroying SctpTransport and receiving notification on timer thread."
...
This reverts commit a88fe7be146b9b85575504d4d5193c007f2e3de4.
Reason for revert: Breaks downstream test, still investigating.
Original change's description:
> Fix race between destroying SctpTransport and receiving notification on timer thread.
>
> This gets rid of the SctpTransportMap::Retrieve method and forces
> everything to go through PostToTransportThread, which behaves safely
> with relation to the transport's destruction.
>
> Bug: webrtc:12467
> Change-Id: Id4a723c2c985be2a368d2cc5c5e62deb04c509ab
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208800
> Reviewed-by: Niels Moller <nisse@webrtc.org >
> Commit-Queue: Taylor <deadbeef@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#33364}
TBR=nisse@webrtc.org
Bug: webrtc:12467
Change-Id: Ib5d815a2cbca4feb25f360bff7ed62c02d1910a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209820
Reviewed-by: Taylor <deadbeef@webrtc.org >
Commit-Queue: Taylor <deadbeef@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33386}
2021-03-05 03:06:38 +00:00
