No customers have been identified.
Bug: chromium:1197965
Change-Id: Ia3063d0909c718ffb8e824225c8c60180551115a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214963
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33700}
This reverts commit 5a40b3710545edfd8a634341df3de26f57d79281.
Reason for revert: Fixed the bug and ran layout tests.
Original change's description:
> Revert "Use the new DNS resolver API in PeerConnection"
>
> This reverts commit acf8ccb3c9f001b0ed749aca52b2d436d66f9586.
>
> Reason for revert: Speculative revert for https://ci.chromium.org/ui/p/chromium/builders/try/win10_chromium_x64_rel_ng/b8851745102358680592/overview.
>
> Original change's description:
> > Use the new DNS resolver API in PeerConnection
> >
> > Bug: webrtc:12598
> > Change-Id: I5a14058e7f28c993ed927749df7357c715ba83fb
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212961
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33561}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta@webrtc.org
>
> Bug: webrtc:12598
> Change-Id: Idc9853cb569849c49052f9cbd865614710fff979
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213188
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33591}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12598
Change-Id: Ief7867f2f23de66504877cdab1b23a11df2d5de4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33647}
- Signal is transferred from jsep_transport_controller to dtls_transport,
jsep_transport_controller is already using Callbacklist and this
modified the dtls_transport to use callback_list.
Bug: webrtc:11943
Change-Id: I4a7ed08e6dab21b8eb515d4d8971f9b084fb8c86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203722
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33137}
This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a
Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33001}
This reverts commit 69241a93fb14f6527a26d5c94dde879013012d2a.
Reason for revert: Breaks WebRTC roll into Chromium.
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
TBR=mbonadei@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: True
Bug: webrtc:9987
Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32986}
The "//rtc_base:rtc_base" build target has historically been one of the
biggest targets in the WebRTC build. Big targets are the main source of
circular dependencies and non-API types leakage.
This CL is a step forward into splitting "//rtc_base:rtc_base" into
smaller targets (as originally started in 2018).
The only non-automated changes are (like re-wiring the build system):
* The creation of //rtc_base/async_resolver.{h,cc} which allows to
break a circular dependency (is has been extracted from
//rtc_base/net_helpers.{h,cc}).
* The creation of //rtc_base/internal/default_socket_server.{h,cc} to
break another circular dependency.
Bug: webrtc:9987
Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32941}
This is a reland of f5e261aaf65cdf2eb903cdf40d651846be44f447
This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since
there have been some issues.
Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}
Bug: webrtc:11495, webrtc:12218
Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32668}
This is a reland of 40261c3663fe316cfe40262c59cee993165ccf63
Note: Instead of changing the type of JsepTransportController->SignalSSLHandshakeError
added a new member with a different name and used it in webrtc code.
After this change do two more follow up CLs to completely remove the old code
from google3.
Original change's description:
> Replace sigslot usages with robocaller library.
>
> - Replace all the top level signals from jsep_transport_controller.
> - There are still sigslot usages in this file so keep the inheritance
> and that is the reason for not having a binary size gain in this CL.
>
> Bug: webrtc:11943
> Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
> Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32321}
Bug: webrtc:11943
Change-Id: Ia07394ee395f94836f6b576c3a97d119a7678e1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186946
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32359}
This reverts commit f5e261aaf65cdf2eb903cdf40d651846be44f447.
Reason for revert: Breaks downstream projects.
Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11495
Change-Id: Ice318d1b11ca3dff09c190187a0b0a32ca945fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32335}
This reverts commit 40261c3663fe316cfe40262c59cee993165ccf63.
Reason for revert: Breaks downstream project
Original change's description:
> Replace sigslot usages with robocaller library.
>
> - Replace all the top level signals from jsep_transport_controller.
> - There are still sigslot usages in this file so keep the inheritance
> and that is the reason for not having a binary size gain in this CL.
>
> Bug: webrtc:11943
> Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
> Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32321}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,glahiru@webrtc.org
Change-Id: Icf438f87c3d95940d858db3cc5848b23abb82fc4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186844
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32324}
- Replace all the top level signals from jsep_transport_controller.
- There are still sigslot usages in this file so keep the inheritance
and that is the reason for not having a binary size gain in this CL.
Bug: webrtc:11943
Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32321}
The PseudoTcp test class is being used outside of WebRTC in ways
that WebRTC itself doesn't, which caused this revert:
https://webrtc-review.googlesource.com/c/src/+/186564
As it happens though, PseudoTcp doesn't actually use the
StreamInterface part of FifoBuffer, so this CL cuts the dependency
from PseudoTcp on FifoBuffer.
Moving forward, we could just remove this class from WebRTC.
Bug: webrtc:11988
Change-Id: Id34a2a6305e8fe37d705ba5e8876dd6398515125
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186665
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32309}
This macro introduces the possibility to suggest the compiler that a
data member doesn't need an address different from other non static
data members.
The usage of a macro is to maintain portability since at the moment
the attribute [[no_unique_address]] is only supported by clang
with at least -std=c++11 but it should be supported by all the
compilers starting from C++20.
Bug: webrtc:11495
Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32246}
Also remove test code that can cause leaks into production.
Add sequence checkers.
Bug: webrtc:11988
Change-Id: I67b4cec6ee77d73ccffbbc88c9081ebb3c3cc423
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185503
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32228}
Setting gtest_enable_absl_printers to false in .gn uncovers some missing
dependencies that were pulled in by gtest.
Bug: None
Change-Id: Ibd7772f6e2af9c798c97161c24f70b1658e3723c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177843
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31551}
This change fixes a problem where the eventual destruction of a
completed resolver sometimes doesn't happen. This is because the
destruction is posted to the network thread, and if it's destroyed
before the closure is executed, the resolver is leaked.
The fix is in three parts:
1. The resolver->Destroy call is performed on closure destruction
to make sure it will always run.
2. The closure is executed with task queue. This because the
RTC_DCHECK on thread:140 fires with the invoker_.
3. It's not possible to guarantee the context Destroy is called on due
to TaskQueue semantics. Therefore SignalThread::Destroy was changed
to accept any calling context and only requiring it's the last
public call to the object.
For unknown reasons, this leak doesn't trigger the leak checker, see
referred bugs for further investigation.
Bug: webrtc:7723, webrtc:11605, chromium:905542
Change-Id: I2681ff1d2416ccbc564974a65ac84781a9ed7aee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176125
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31359}
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.
Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.
CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn
Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
This file only defined an unused alias.
Bug: None
Change-Id: I0c731401295814e8f5dd91f41350973021efd5d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155173
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30138}
This patch estimates the connection RTT using
EventBasedExponentialMovingAverage. The half time is
set to 500 but can be modified using field trials.
This new metric is currently unused, but will
be used for exploration of of whether it can be used
instead of the existing metric.
Bug: webrtc:11140
Change-Id: I9db93e9b9eb932e3cd18935cd4ce0d90fc1cb293
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161000
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29944}
MediaTransport is deprecated and the code is unused.
No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
Several patches for webrtc:10647 has split the
P2PTransportChannel class/file. This has had the
side effect of it being hard to share the IceFieldTrials-struct.
This patch moves that struct into own file so that can be included
from other components. This patch is a behavioral NOP.
BUG=webrtc:10647
Change-Id: If49cd4d919684a48dde3188a26baf20e4ff2cd8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160301
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29876}
All downstream users have been moved to the new one.
Bug: webrtc:11091
Change-Id: Ia18d0df94a7b95b1a58b4a53cfb195c61ef59ffd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160201
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29873}
This patch moves the logic for
- selection of connection to ping
- selection of connection to use
- selection of connection to prune
into own file and puts it behind a new interface called 'IceControllerInterface'.
BUG=webrtc:10647
Change-Id: I10228b3edd361d3200fa4a734d74a319560966c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158205
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29850}
We now have two downstream users of stun.h, so it appears to be
generally usable. I put this in a new dir networking/, but I'm open to
suggestions here (maybe some things in api/ should move in there).
I checked what our downstream users are actually using, and it's
cricket::ComputeStunCredentialHash
cricket::<constants>
cricket::TurnMessage
cricket::GetStunErrorResponseType
cricket::StunAttribute::CreateAddress
cricket::StunErrorCodeAttribute
cricket::StunByteStringAttribute
StunAttribute::CreateUnknownAttributes
cricket::TurnErrorType
cricket::StunMessage
I reckoned that was pretty much everything in stun.h, so I didn't
bother splitting it up. They don't use every function and constant
in there, but all _types_ of functions and constants, so for the
sake of coherence I don't think it makes sense to split it.
There's some old stuff in there like GTURN which could arguably
be split out, but it should likely go away soon anyway, so I don't
think it's worth the effort.
Steps:
1) land this
2) update downstream to point to the new header and target
3) remove p2p/base:stun_types.
Bug: webrtc:11091
Change-Id: I1f05bf06055475d25601197ec6fefb8d3b55e8e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159923
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29822}
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
This patch adds a feature enabled using webrtc field trial
that remove connections between RELAY and non-RELAY candidates.
Bug: webrtc:11021
Change-Id: I924076277a843bffc1d25f6de14d2165f7012c4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156083
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29464}
This is a reland of 5b4fcb5bf69218c2f42ca2b0cada6c15f2f638e9
Original change's description:
> New build target p2p:stun_types
>
> The media:rtc_media_base target needs definitions of various
> stun-related types and constant. With this new smaller target, it no
> longer needs to depend on all of p2p.
>
> Bug: webrtc:8733
> Change-Id: I05910b6915f6d2c96e8f52a017adbc7eb693dca8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150945
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29036}
Tbr: steveanton@webrtc.org
Bug: webrtc:8733
Change-Id: I1847007ecf29e0e6a27f559b92df632a1cd69280
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151880
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@{#29092}
I gave up on removing proxy_info, user_agent and tcp_options. I don't
think it's feasible to remove them without removing all the proxy code.
The assumption that you can set the proxy and user agent long after
you have created the factory is entrenched in unit tests and the code
itself. So is the ability to set tcp opts depending on protocol or
endpoint properties.
It may be easier to untangle proxy stuff from the factory later,
when it becomes a more first-class citizen and isn't passed via
the allocator.
Requires https://chromium-review.googlesource.com/c/chromium/src/+/1778870
to land first.
Bug: webrtc:7447
Change-Id: Ib496e2bb689ea415e9f8ec1dfedff13a83fa4a8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150799
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29091}
This reverts commit 5b4fcb5bf69218c2f42ca2b0cada6c15f2f638e9.
Reason for revert: Breaks build
Original change's description:
> New build target p2p:stun_types
>
> The media:rtc_media_base target needs definitions of various
> stun-related types and constant. With this new smaller target, it no
> longer needs to depend on all of p2p.
>
> Bug: webrtc:8733
> Change-Id: I05910b6915f6d2c96e8f52a017adbc7eb693dca8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150945
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29036}
TBR=steveanton@webrtc.org,mbonadei@webrtc.org,nisse@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8733
Change-Id: I6e00657a6137ff773325f37ec02ee1014b6fe96b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151740
Reviewed-by: Hannes Landeholm <hnsl@webrtc.org>
Commit-Queue: Hannes Landeholm <hnsl@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29085}
The media:rtc_media_base target needs definitions of various
stun-related types and constant. With this new smaller target, it no
longer needs to depend on all of p2p.
Bug: webrtc:8733
Change-Id: I05910b6915f6d2c96e8f52a017adbc7eb693dca8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150945
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29036}
