A bug in the id being searched for inside OpenSSLStreamAdapter::SslCipherSuiteToName prevented the lookup from ever succeeding.
This resulted in this stat being unavailable when calling PeerConnection::GetStats(). To fix the problem, look for (0x03000000L | cipher_suite) which matches what the BoringSSL codepath is doing.
Bug: webrtc:14596
Change-Id: Ic36d77dbc4c2378fbde1e2f21a9f5bd735b36741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38460}
For now, the run-time check will only be enabled if the
rtc_strict_field_trials GN arg is set.
In order to allow testing with imaginary field trial keys, two test
helpers have been added. It's a bit awkward to test these since the
field trial string is already global, hence the helpers are also
modifying global state. Tests must make sure this global state is reset
between runs. Things won't be an issue anymore when [1] has removed the
global string.
[1] https://crbug.com/webrtc/10335
Bug: webrtc:14154
Change-Id: Ida44cc817079d7177325e2228cf1f1d242b799e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276269
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38447}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
This cl move VideoEncoderConfig from api/ to video/config.
VideoStreamEncoderInterface and VideoStreamEncoderObserver
are moved as collateral.
brandt@ think that the reason these were in api/ in the
first place had to downstream project.
Functionality wise, this is a NOP, but it makes it easier
to modify the encoder (config).
Bug: webrtc:14451
Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38242}
ExpectationToString is used to explain why RTC_DCHECK_RUN_ON is
triggered.
Unfortunately, the current implementation only generates verbose strings
when SequenceCheckerImpl is passed as an argument.
Modify ExpectationToString to generate detailed messages even for
derived classes of SequenceCheckerImpl.
Bug: None
Change-Id: I55f76d44ad59dbe6f21cee7d7d8e19188e0f3088
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276061
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38211}
Unlike the cache of the entire stats report which is time limited, this
certificate cache is valid for an unlimited amount of time, but is
cleared at ClearCachedStatsReport() which is already called on each
SLD/SRD call. Since certificates can only change by negotiation, this
cache is ensured to always be invalidated when certificates change.
Since ClearCachedStatsReport() can happen for other reasons than
certificates changing we may clear the cache more often then is
necessary, but arguably this is seldom enough that we don't have to
create a separate "ClearCertificateStats()" method. Keep it simple?
The cache specifically avoids rtc::SSLCertChain::GetStats which
trigger rtc::SSLCertificate::GetStats and rtc::Base64::EncodeFromArray.
Bug: webrtc:14458
Change-Id: I5f95a4a5eb51cc4462147270fdae7bb9fb7bc822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276602
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38205}
With rtc::Thread::Clear removed, there are no longer calls to external code while holding the mutex and thus it doesn't need to be recursive.
Bug: webrtc:11567
Change-Id: If350684be0bfcbc33806019bd986138905aec66d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276542
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38179}
"ccmni" is the cellular interface name in MTK platform.
The accurate name may be ccmni0, ccmni1 ...etc. We would like
to add it in order to provide correct adapter type when user
invokes webrtc function GetAdapterTypeFromName.
Bug: b/246680543, webrtc:14438
Change-Id: I30074a442b3c02b93bc72a044675718955bd96f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275860
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38113}
Instead protect pending tasks with a shared object.
Some tests destroy VirtualSocket on a different thread than it is used on,
Some tests destroy VirtualSocket together with VirtualSocketServer after
associated thread is deleted, thus complicated check is used to ensure
VirtualSockets are safe to use.
Bug: webrtc:9702
Change-Id: I1a19cd24ac6a598a1cde64434104cad0b750096e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274460
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38103}
The `TimestampExtrapolator` is only used by the `VCMTiming`
class, despite there being references to it from both
`modules/rtp_rtcp/BUILD.gn` and `modules/video_coding/BUILD.gn`.
Bug: webrtc:14111
Change-Id: If1a02a56a0c83b13d619ca08dc76c884fa829369
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275482
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38093}
* Make `percentile` configurable and rename class.
* Introduce convenience type `MovingMedianFilter` that
maintains the behaviour of the old class with that name.
* Move home grown moving 95th percentile filter in
`JitterEstimator` to this new utility class.
Bug: webrtc:14151
Change-Id: I17d525b6e0bc98c28568c7dfe94b72eeab4a1ca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275311
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38082}
Most of the changes are trivial.
Bug: webrtc:14432
Change-Id: I0444527bf57c72c8d65f69754b4a4a1c1d7b2e92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38074}
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
To reduce usage of rtc::MessageHandler, hide rtc::Thread::Send into private section with intention to deprecate it in favor of the new Invoke function.
Bug: webrtc:9702, webrtc:11318
Change-Id: Ib4c26f9abc361e05a45b2a91929af58ab160b3f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274166
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38036}
Post task in a less safer way assuming test won't try to destroy the manager right after starting the updates.
Bug: webrtc:11988
Change-Id: Idf5d001b9c09d8df793112e4ad95690f9660ac3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273961
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38007}
This reverts commit 83db78e854.
Reason for revert: Some tests in Chromium's blink no longer compile because of the change in the signature of the CreateDefaultTaskQueueFactory() function.
Original change's description:
> Add TaskQueueStdlib experiment.
>
> Bug: webrtc:14389
> Change-Id: I23c6e0ae675748ec35a99c334104dd2654995a33
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265802
> Commit-Queue: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37888}
Bug: webrtc:14389
Change-Id: If3e63d6b4ab9e838dc5020b88076a73fd29916e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272920
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37902}
rtc::Thread::Dispose is only used in test code,
but complicates the main thread loop.
Bug: webrtc:8324
Change-Id: I2dccdadcdc932b9992958d1e70fb93d1879b7618
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272821
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37894}
Remote Peek function as unused
Move Get and Dispatch into private section to ensure they are not used
from outside.
Bug: webrtc:9702
Change-Id: Ibd0b236fe43543d60f97f988524526493bbeaaa7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272804
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37889}
This CL adds TimeDelta support to the rtc::Event, and updates
the Wait implementations to work with the improved precision.
Bug: webrtc:14366
Change-Id: Iefeb638b18176a34f4ed2a5131754a7b7e6c9e99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272002
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37831}
While [1] partially fixed the problem, UB was still possible when
AppendData was called with size=0 or in one of the constructors.
[1] - https://webrtc-review.googlesource.com/c/src/+/271502
Bug: webrtc:14292
Change-Id: I9196e23687ee82b7bfbe1ed43460d9f99adcd1ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37814}
While transitioning to TimeDelta, WebRTC and Chromium has a
different idea about what type rtc::Event::kForever is. Code
can't assume rtc::Event::kForever is the same type as timed
wait arguments.
Bug: webrtc:14366
Change-Id: I4783c7de6d567c70b211de9aa9889417f6fafba1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37810}
As per https://en.cppreference.com/w/cpp/string/byte/memcpy, calling
std::memcpy with dest or src as nullptr is UB.
This CL prevents this from happening and is also looking into
why UBSan didn't catch the issue.
Bug: webrtc:14292
Change-Id: I99833f28ac865719d0dcb02c4de00f33a48c3992
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271502
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37783}
This combines the below IPv6 fixes into the field trial
WebRTC-IPv6NetworkResolutionFixes:
1. Prefer global IPv6 address over link local
2. Use address family when resolving STUN hostname
WebRTC-PreferGlobalIPv6ToLinkLocal is currently in Dev but will be
rolled back temporarily.
Bug: webrtc:14334, webrtc:14131
Change-Id: I1fb3f55c4c5f3c5c0b441ece30e72cf393e074d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271340
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37754}
