Commit graph

944 commits

Author SHA1 Message Date
Niels Möller
611fba4517 Mark construction time members of PhysicalSocketServer as const
Bug: webrtc:11567
Change-Id: I06d48aa1636ce1dc684e6a1f6332366be9df22d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175007
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31242}
2020-05-13 14:37:35 +00:00
Tommi
a98cea863d Remove the PendingTaskSafetyFlag::Pointer type add ScopedTaskSafety.
ScopedTaskSafety simplifies usage of PendingTaskSafetyFlag,
so this CL also includes ToQueuedTask support for ScopedTaskSafety
and test updates.

This is following up on feedback in the following CL:
https://webrtc-review.googlesource.com/c/src/+/174262

Change-Id: Idd38dfc1914b24a05fdc4ad256b409dcf1795fc0
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174740
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31241}
2020-05-13 14:17:39 +00:00
Mirko Bonadei
ff88a64b67 Revert "Delete unused code to handle posix signals in PhysicalSocketServer"
This reverts commit d2490aef20.

Reason for revert: peerconnection_client fails to link.

Original change's description:
> Delete unused code to handle posix signals in PhysicalSocketServer
> 
> Bug: None
> Change-Id: I3abddef4f1af5499f39a8d3f643c779effe9e01d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175006
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31237}

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

Change-Id: Ia5a44b4f1a54f6b444b8c53e64d1a3972d166728
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175011
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31240}
2020-05-13 13:48:28 +00:00
Niels Möller
d2490aef20 Delete unused code to handle posix signals in PhysicalSocketServer
Bug: None
Change-Id: I3abddef4f1af5499f39a8d3f643c779effe9e01d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175006
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31237}
2020-05-13 12:33:09 +00:00
Jonas Oreland
2105d64a02 Add field trial for using different network cost cellular types
This field trial will be used to rollout the cellular costs added
in https://webrtc-review.googlesource.com/c/src/+/172582 in
a controlled fashion.

Bug: webrtc:11473
Change-Id: I14fd5cada187ba161124325a7ff69d355ef52b25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174880
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31233}
2020-05-13 09:43:11 +00:00
Per Åhgren
b8a9630e9e Add a Release method for file wrapper
This CL adds a Release method for the FileWrapper class that allows it
to release the wrapped FILE* object without closing it.

Bug: b/155316201
Change-Id: If9ef4345724705dc7c66183f17bd8daadbdd00b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174720
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31183}
2020-05-07 14:37:00 +00:00
Sergey Ulanov
9af75432b2 Add RTC_EXPORT for NullSocketServer
NullSocketServer needs to be exported in order to use it in
JingleThreadWrapper in chromium.

Bug: none
Change-Id: I9bce49c764a1ca1c28fc44041d0d5f04f794066e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173920
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#31174}
2020-05-06 20:19:49 +00:00
Tommi
3c5450e693 Add support for PendingTaskSafetyFlag to ToQueuedTask.
This keeps usage of ToQueuedTask consistent and avoids callers having
to add additional boiler plate when using the safety flag.

From this:

tq->PostTask(ToQueuedTask([safety = my_safety_flag_]() {
  if (!safety->alive())
    return;
  Foo();
});

to this:

tq->PostTask(ToQueuedTask(my_safety_flag_, []() {
  Foo();
});


Bug: none
Change-Id: I205af56a64dd9839eb845321083d533140d614ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174262
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31161}
2020-05-04 18:20:10 +00:00
Mirko Bonadei
a81e9c82fc Wrap WebRTC OBJC API types with RTC_OBJC_TYPE.
This CL introduced 2 new macros that affect the WebRTC OBJC API symbols:

- RTC_OBJC_TYPE_PREFIX:
  Macro used to prepend a prefix to the API types that are exported with
  RTC_OBJC_EXPORT.

  Clients can patch the definition of this macro locally and build
  WebRTC.framework with their own prefix in case symbol clashing is a
  problem.

  This macro must only be defined by changing the value in
  sdk/objc/base/RTCMacros.h  and not on via compiler flag to ensure
  it has a unique value.

- RCT_OBJC_TYPE:
  Macro used internally to reference API types. Declaring an API type
  without using this macro will not include the declared type in the
  set of types that will be affected by the configurable
  RTC_OBJC_TYPE_PREFIX.

Manual changes:
https://webrtc-review.googlesource.com/c/src/+/173781/5..10

The auto-generated changes in PS#5 have been done with:
https://webrtc-review.googlesource.com/c/src/+/174061.

Bug: None
Change-Id: I0d54ca94db764fb3b6cb4365873f79e14cd879b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31153}
2020-05-04 15:01:26 +00:00
Marina Ciocea
ce1320cc4d Add WaitForPreviouslyPostedTasks to TaskQueueForTest.
Add an utility function to TaskQueueForTest to execute all already
posted tasks on the queue.

Bug: webrtc:11380
Change-Id: I6cf75bc543cfd2dd1c363935134d3f7bd55eec58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174140
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31152}
2020-05-04 13:47:35 +00:00
Mirko Bonadei
95d9a1a3d7 Update set of known root certificates.
This has been automatically generated by running [1].

See https://codereview.webrtc.org/1503473002 for some background about
the generator script.

[1] - https://cs.chromium.org/chromium/src/third_party/webrtc/tools_webrtc/sslroots/generate_sslroots.py

Bug: chromium:978779
Change-Id: I78cf8947b3363738dd0e21182348253dbad95f02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173821
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31131}
2020-04-24 20:40:45 +00:00
Mirko Bonadei
2c80923230 Remove WebRTC-LibvpxVp{8,9}TrustedRateController.
Bug: webrtc:11503
Change-Id: I58704606a109a9f6a5dbc1bfd59ca76fa8c23d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173479
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31095}
2020-04-17 09:17:57 +00:00
Taylor Brandstetter
4479a822c0 Remove deprecated SSLIdentity methods that return raw pointers.
Bug: webrtc:11410
Change-Id: I40e5549cb7c1082eebd870e0f133a3be0918dcaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173571
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31092}
2020-04-16 20:56:25 +00:00
Mirko Bonadei
6415dcad7a Remove WebRTC-ExperimentalScreenshareSettings.
This field trial is unused.

Bug: webrtc:11503
Change-Id: Id79b0dc64fed3559b9b63ebcf539e5536ddad589
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173339
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31090}
2020-04-16 18:15:08 +00:00
Harald Alvestrand
b33a0ca1ee Remove deprecated ssl_identity methods
This is a followup to
https://webrtc-review.googlesource.com/c/src/+/170637

Bug: webrtc:11450
Change-Id: I69928ed7236c6a8a569c7dc0383f7debb4408179
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171224
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31086}
2020-04-16 14:21:41 +00:00
Tommi
3c9bcc1f7a Reland of the test portion of:
https://webrtc-review.googlesource.com/c/src/+/172847

------------ original description --------------

Preparation for ReceiveStatisticsProxy lock reduction.

Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).

Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.

One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.

Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.

Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.

Bug: webrtc:11489
Change-Id: I491e13344b9fa714de0741dd927d907de7e39e83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173583
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31077}
2020-04-15 16:09:44 +00:00
Mirko Bonadei
f7f6870f1b Mark static const class/struct members as constexpr.
This change fixes declarations that have initial values but are
technically not definitions by marking them constexpr (which counts as a
definition).

Bug: None
Change-Id: Icbecf8d83faffa83b9f7e1ffe4d6ef3a3f0b0c2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173587
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31073}
2020-04-15 09:30:07 +00:00
Mirko Bonadei
dce61741f6 Remove deprecated SSLAdapter::SetIdentity.
Bug: webrtc:10198
Change-Id: I675bc08bffa2774546357fb0b554bd52ca69c095
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173465
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31061}
2020-04-14 11:00:49 +00:00
Mirko Bonadei
9f297b5960 Remove OpenSSLIdentity::GenerateWithExpiration.
These static functions were marked as deprecated and since they
are not used this CL just removes them.

Bug: webrtc:10198
Change-Id: I4872e31701543c988fe71ab4e0b32bd73ff07753
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173467
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31057}
2020-04-14 09:34:04 +00:00
Mirko Bonadei
2d2c2947fd Remove OpenSSLAdapter restartable_ data member.
Bug: webrtc:10198
Change-Id: I5beabba3837b92d600e2d7067954adf334adbdd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173335
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31056}
2020-04-14 09:01:23 +00:00
henrika
f0dc5c52be Adds tiny rtc::StringFormat utility
Bug: webrtc:11493
Change-Id: If11a0362dfa820e4464129d0ea58ff8bc4ce86bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173323
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31043}
2020-04-09 18:13:11 +00:00
mmorrison
25eeda1872 Fix socket not getting registered for epoll events
When epoll is enabled in the PhysicalSocketServer, a socket may
not get registered for its epoll events. If an AsyncSocket is
closed and re-created during one of its signal callbacks, its
old epoll events and new epolls events bitmasks may be the same,
even though the fd has changed. This causes the epoll implementation
to not register the new fd for any events.

Fix this by resetting the saved events bitmask when the socket is
closed. This ensures the new fd, if any, is registered if needed.

Bug: webrtc:11497
Change-Id: Idea499e09aefdf292430d1a774a046f963603b95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173103
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31039}
2020-04-09 10:17:47 +00:00
Artem Titov
16cc9efd54 Revert "Preparation for ReceiveStatisticsProxy lock reduction."
This reverts commit 24eed2735b.

Reason for revert: Speculative revert: breaks downstream project

Original change's description:
> Preparation for ReceiveStatisticsProxy lock reduction.
> 
> Update tests to call VideoReceiveStream::GetStats() in the same or at
> least similar way it gets called in production (construction thread,
> same TQ/thread).
> 
> Mapped out threads and context for ReceiveStatisticsProxy,
> VideoQualityObserver and VideoReceiveStream. Added
> follow-up TODOs for webrtc:11489.
> 
> One functional change in ReceiveStatisticsProxy is that when sender
> side RtcpPacketTypesCounterUpdated calls are made, the counter is
> updated asynchronously since the sender calls the method on a different
> thread than the receiver.
> 
> Make CallClient::SendTask public to allow tests to run tasks in the
> right context. CallClient already does this internally for GetStats.
> 
> Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
> 
> Bug: webrtc:11489
> Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31008}

TBR=mbonadei@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org,tommi@webrtc.org,juberti@webrtc.org,mflodman@webrtc.org

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

Bug: webrtc:11489
Change-Id: I48b8359cdb791bf22b1a2c2c43d46263b01e0d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173082
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31023}
2020-04-07 19:50:20 +00:00
Tommi
24eed2735b Preparation for ReceiveStatisticsProxy lock reduction.
Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).

Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.

One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.

Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.

Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.

Bug: webrtc:11489
Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31008}
2020-04-06 14:34:38 +00:00
Jonas Oreland
c7ea04af91 Don't trigger OnNetworkChange when changing from 3G to 4G
This patch is a follow up to https://webrtc-review.googlesource.com/c/src/+/172582
and change so that a switch from CELLULAR_X to CELLULAR_Y does not
trigger OnNetworkChange.

This is needed as the OnNetworkChange signals triggers
BasicPortAllocator to rescan all networks and generate new candidates.

The actual adapter type change is still possible to react on using
SignalTypeChanged.

BUG: webrtc:11473
Change-Id: Icc1a945b8a4df1714c6ec4b02ec759ecada92d7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172802
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30992}
2020-04-03 09:37:52 +00:00
Mirko Bonadei
06d3559b79 Replace std::string::find() == 0 with absl::StartsWith (part 2).
This CL has been generated using clang-tidy [1] except for changes to
BUILD.gn files.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/abseil-string-find-startswith.html

Bug: None
Change-Id: Ibf75601065a53bde28623b8eef57bec067235640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172586
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30984}
2020-04-02 14:38:30 +00:00
Mirko Bonadei
cfa0e8ffe2 Fix errors C2238, C2248 and C2059 on MSVC bots.
This CL fixes the following errors on MSVC bots:

../../rtc_base/units/unit_base_unittest.cc(42): error C2059:
  syntax error: '<'

../../rtc_base/units/unit_base_unittest.cc(42): error C2238:
  unexpected token(s) preceding ';'

../..\rtc_base/units/unit_base.h(39): error C2248:
  'webrtc::`anonymous-namespace'::TestUnit::TestUnit':
  cannot access protected member declared in class
  'webrtc::`anonymous-namespace'::TestUnit'

No-Try: True
Bug: None
Change-Id: Ic63a75132107381474aca2e1d42ba96d1f6a1c00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172621
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30972}
2020-04-02 09:54:27 +00:00
Jonas Oreland
08d1806e54 Extend rtc::AdapterType with 2g, 3G, 4G & 5G enum values.
This patch adds new enum values for different types of cellular
connections.

The new costs are currently blocked when sending to remote,
(so that arbitrary network switches does not starts occurring).

The end-game for this series to be able to distinguish between
different type of cellular connections in the ice-layer (e.g when
selecting/switching connections).

BUG: webrtc:11473
Change-Id: I587ac8fdff4f6cdd0f8905f327232f58818db4f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172582
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30970}
2020-04-02 07:48:36 +00:00
Mirko Bonadei
6f402f991e Remove unnecessary breaks after return.
Patch author: thakis@chromium.org.

TBR=kwiberg@webrtc.org

No-Try: True
Bug: chromium:1066980
Change-Id: Ifcc7e831337bb2a9bf06b0af0bbd9d1c586db78a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172627
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30968}
2020-04-01 22:20:37 +00:00
Paulina Hensman
b239a2e357 Remove some more instances of IP logging.
Bug: b/152662380
Change-Id: I1f33f470c4dd5458c2d2598e2f17f6691f72df4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172446
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30957}
2020-04-01 08:17:47 +00:00
Christoffer Rodbro
1c7a6589a9 Add test for relay bandwidth capping.
Feature was added in
https://webrtc-review.googlesource.com/c/src/+/171226

Bug: webrtc:11434
Change-Id: Iee1e350976ab4043f15c5932cdc4f53b413bb302
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171861
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30940}
2020-03-30 13:02:46 +00:00
Olga Sharonova
f74d2ce649 Revert "Add interface_id to rtc::Network"
This reverts commit 7e91482fcc.

Reason for revert: Speculative revert, as Android FYI bots are red
starting https://webrtc.googlesource.com/src/+/7e91482fcc496103f36333a569992c81b6dc9e9c
where this CL landed.

See also https://bugs.chromium.org/p/chromium/issues/detail?id=1065805.

Original change's description:
> Add interface_id to rtc::Network
>
> This patch adds an interface_id property
> to rtc::Network. It is an enumeration of the
> interface names that are present.
>
> This enables a local ICE agent to keep track
> of which connections are using which interfaces,
> something that is useful for predicting how
> connections behave.
>
> This is part 1 of https://webrtc-review.googlesource.com/c/src/+/85520
>
> Bug: webrtc:9446
> Change-Id: Ia6ec1f14ac240799fb1be49d67d82e2733e87acf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171061
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30882}

No-Presubmit: True
Bug: webrtc:9446
TBR=hta@webrtc.org, jonaso@webrtc.org

Change-Id: If86e2e0653b53a8eae26a97ce9fa68748b440607
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172092
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30937}
2020-03-30 09:29:51 +00:00
Jonas Oreland
7e91482fcc Add interface_id to rtc::Network
This patch adds an interface_id property
to rtc::Network. It is an enumeration of the
interface names that are present.

This enables a local ICE agent to keep track
of which connections are using which interfaces,
something that is useful for predicting how
connections behave.

This is part 1 of https://webrtc-review.googlesource.com/c/src/+/85520

BUG: webrtc:9446
Change-Id: Ia6ec1f14ac240799fb1be49d67d82e2733e87acf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171061
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30882}
2020-03-25 13:03:09 +00:00
Jonas Oreland
d1a0062db7 remove deprecated fields in rtc::NetworkRoute
this patch is a followup to https://webrtc-review.googlesource.com/c/src/+/170628
and removed the now deprecated fields {local/remote}_network_id that
is now no longer used by downstream.

BUG: webrtc:11434
Change-Id: Ia322609c0b4f07b05b8592cbca7f001a115da109
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171515
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30874}
2020-03-25 06:44:06 +00:00
Jonas Oreland
5b6a4d8908 Only print route if it has changed
This is a follow up change to https://webrtc-review.googlesource.com/c/src/+/170628
and modifies code to only LOG if the route really has changed.

Existing code will LOG like this, which is slightly annoying. Notice that the same route change is LOG:ed twice.
03-23 13:28:49.281 17986 18850 I rtp_transport_controller_send.cc: [1183:590] [18850] (line 253): Network route changed on transport audio: new_route = [ connected: 1 local: [ 2/4 Wifi turn: 0 ] remote: [ 2/3 Wifi turn: 0 ] packet_overhead_bytes: 32 ]
03-23 13:28:49.281 17986 18850 I rtp_transport_controller_send.cc: [1183:590] [18850] (line 278): old_route = [ connected: 1 local: [ 2/4 Wifi turn: 1 ] remote: [ 2/3 Wifi turn: 0 ] packet_overhead_bytes: 28 ]
03-23 13:28:49.281 17986 18850 I rtp_transport_controller_send.cc: [1183:591] [18850] (line 253): Network route changed on transport audio: new_route = [ connected: 1 local: [ 2/4 Wifi turn: 0 ] remote: [ 2/3 Wifi turn: 0 ] packet_overhead_bytes: 32 ]
03-23 13:28:49.282 17986 18850 I rtp_transport_controller_send.cc: [1183:591] [18850] (line 278): old_route = [ connected: 1 local: [ 2/4 Wifi turn: 0 ] remote: [ 2/3 Wifi turn: 0 ] packet_overhead_bytes: 32 ]

The way this method is called twice with same argument is out of scope
for this change.

BUG: webrtc:11434
Change-Id: I052d089c59714513a09cbaed49f24c8f1300af58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171460
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30865}
2020-03-24 11:48:42 +00:00
Mirko Bonadei
f1df04b094 Fix -Wunreachable-code on Linux.
Starting from [1] the toolchain has started to enforce
-Wunreachable-code on Linux, this CL fixes the issues that are preventing
the Chromium roll into WebRTC.

Error example at [2].

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/2093537
[2] - https://ci.chromium.org/p/webrtc/builders/try/linux_rel/34282?

Bug: webrtc:11448
Change-Id: I96e8901ae80c44d69143ed8d972e250b6b926a7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30858}
2020-03-23 20:51:37 +00:00
Harald Alvestrand
8515d5a4ab Refactor ssl_stream_adapter API to show object ownership
Backwards compatible overloads are provided.

Bug: none
Change-Id: I065ad6b269fe074745f9debf68862ff70fd09628
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170637
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30851}
2020-03-21 18:53:46 +00:00
Jonas Oreland
71fda3613c Extend NetworkRoute with more info about local/remote endpoints
This patch extends the NetworkRoute struct with more information
about local/remote endpoints. It adds
- adapter type
- adapter id
- relay

(previously it was "only" network_id)

The patch leaves the {local/remote}_network_id fields
around and populated since downstream projects depend
on them. They will be removed once they have migrated.

OWNER: srte@ call/ test/
OWNER: asapersson@ video/
OWNER: hta@ p2p/ pc/ rtc_base/

BUG: webrtc:11434
Change-Id: I9bcec385b40d707db385fef40b2c7a315dd35dd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170628
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30848}
2020-03-20 16:55:38 +00:00
Yura Yaroshevich
ebf739be7b Reland "Leverage dispatch_queue_create_with_target when possible."
This is a reland of de86381161

Original change's description:
> Leverage dispatch_queue_create_with_target when possible.
> 
> Replacing dispatch_queue_create followed by
> dispatch_set_target_queue with dispatch_queue_create_with_target
> is claimed to be source of GCD performance improvement:
> https://developer.apple.com/videos/play/wwdc2017/706/
> Video since 40 min. Slides since 199.
> 
> Bug: webrtc:9055
> Change-Id: I0136f7faaef0951a7ad243bc8772f3ee952d5470
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168491
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
> Cr-Commit-Position: refs/heads/master@{#30781}

Bug: webrtc:9055
Change-Id: I36b0b6423c81c0497f66f7c993741c33ff6ec5ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170443
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30821}
2020-03-18 16:06:09 +00:00
Alex Loiko
fcafbfdbf0 Revert "Leverage dispatch_queue_create_with_target when possible."
This reverts commit de86381161.

Reason for revert: Fails downstream project, """fatal error: 'rtc_base/system/gcd_helpers.h' file not found"""

Original change's description:
> Leverage dispatch_queue_create_with_target when possible.
> 
> Replacing dispatch_queue_create followed by
> dispatch_set_target_queue with dispatch_queue_create_with_target
> is claimed to be source of GCD performance improvement:
> https://developer.apple.com/videos/play/wwdc2017/706/
> Video since 40 min. Slides since 199.
> 
> Bug: webrtc:9055
> Change-Id: I0136f7faaef0951a7ad243bc8772f3ee952d5470
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168491
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
> Cr-Commit-Position: refs/heads/master@{#30781}

TBR=tommi@webrtc.org,kthelgason@webrtc.org,yura.yaroshevich@gmail.com

Change-Id: I47fafa47afa2c825c8f100253d8a1f035203d9e8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9055
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170361
Reviewed-by: Alex Loiko <aleloi@google.com>
Commit-Queue: Alex Loiko <aleloi@google.com>
Cr-Commit-Position: refs/heads/master@{#30785}
2020-03-13 08:02:34 +00:00
Yura Yaroshevich
de86381161 Leverage dispatch_queue_create_with_target when possible.
Replacing dispatch_queue_create followed by
dispatch_set_target_queue with dispatch_queue_create_with_target
is claimed to be source of GCD performance improvement:
https://developer.apple.com/videos/play/wwdc2017/706/
Video since 40 min. Slides since 199.

Bug: webrtc:9055
Change-Id: I0136f7faaef0951a7ad243bc8772f3ee952d5470
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168491
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/master@{#30781}
2020-03-12 20:33:48 +00:00
Minyue Li
dd14a95596 Allow TimestampAligner to translate timestamp without new observation of system clock.
Bug: chromium:1054403
Change-Id: I32c622851fc0bed2c47ae142c743399acb91ae84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169924
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30744}
2020-03-10 17:22:54 +00:00
Harald Alvestrand
137991396d Make a switch to disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC.
This reverts commit af1f8655b2

Landing the change with default set to
"enabled" (DTLS 1.0 will continue to work by default),
so that flipping the default can be a separate CL.

Original change's description:
> Revert "Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC."
>
> This reverts commit 7276b974b7.
>
> Reason for revert: Changing to a later Chrome release.
>
> Original change's description:
> > Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC.
> >
> > This change disables DTLS 1.0, TLS 1.0 and TLS 1.1 in WebRTC by default. This
> > is part of a larger effort at Google to remove old TLS protocols:
> > https://security.googleblog.com/2018/10/modernizing-transport-security.html
> >
> > For the M74 timeline I have added a disabled by default field trial
> > WebRTC-LegacyTlsProtocols which can be enabled to support these cipher suites
> > as consumers move away from these legacy cipher protocols but it will be off
> > in Chrome.
> >
> > This is compliant with the webrtc-security-arch specification which states:
> >
> >    All Implementations MUST implement DTLS 1.2 with the
> >    TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the P-256
> >    curve [FIPS186].  Earlier drafts of this specification required DTLS
> >    1.0 with the cipher suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, and
> >    at the time of this writing some implementations do not support DTLS
> >    1.2; endpoints which support only DTLS 1.2 might encounter
> >    interoperability issues.  The DTLS-SRTP protection profile
> >    SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
> >    Implementations MUST favor cipher suites which support (Perfect
> >    Forward Secrecy) PFS over non-PFS cipher suites and SHOULD favor AEAD
> >    over non-AEAD cipher suites.
> >
> > Bug: webrtc:10261
> > Change-Id: I847c567592911cc437f095376ad67585b4355fc0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125141
> > Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> > Reviewed-by: David Benjamin <davidben@webrtc.org>
> > Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27006}
>
> TBR=steveanton@webrtc.org,davidben@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:10261
> Change-Id: I34727e65c069e1fb2ad71838828ad0a22b5fe811
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130367
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27403}

Bug: webrtc:10261
Change-Id: I28c6819d37665976e396df280b4abf48fb91d533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169851
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30733}
2020-03-09 19:23:44 +00:00
Minyue Li
37e388ad2d Refactor TimestampAligner for more general use.
This only changes the comments and rename variables.

Bug: chromium:1054403
Change-Id: Ie7419ca23e482361e9f90405587b8c8f839b26d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169101
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30710}
2020-03-06 15:05:00 +00:00
Jiawei Ou
14e5f0b2cb Update RTC_CHECK and RTC_LOG macros so they work when called from xxxxx::rtc namespaces
Adding :: before rtc allow us to use the macro in nested rtc namespace for external components like

namespace xxxxxxx {
namespace rtc {
RTC_CHECK(true);
}
}

Bug: webrtc:11400
Change-Id: I79349b847c3fce8197c82aec31b672a1a16e5388
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169683
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30684}
2020-03-04 22:53:34 +00:00
Sebastian Jansson
da7267a10f Makes Thread::Send execute sent messages after pending posted messages.
Bug: webrtc:11255
Change-Id: I4b9036d22c9db3a5ec0e19fc5f2f5ac0d7e2289a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168058
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30667}
2020-03-03 12:15:55 +00:00
Karl Wiberg
d084ea93b6 BoundedInlineVector: Add resize() method
Bug: webrtc:11391
Change-Id: I34d659d0e295617e9058393d4d1b510111a78b83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169520
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30664}
2020-03-02 20:55:28 +00:00
Karl Wiberg
c126937564 BoundedInlineVector: Vector class of bounded size with inline allocation
Selling point is that it never touches the heap. Intended use case is
cheaply returning a variable, bounded, and small number of things from
a function.

Specifically, there are situations where we'd like to return things like

  ArrayView<ArrayView<float>>

where we currently have to allocate an array of ArrayView<float> for
the outer ArrayView to point to, which is a bother; however, although
the outer ArrayView is of variable size, that size is statically
guaranteed to not exceed some small constant. After this CL, we'll be
able to instead return

  BoundedInlineVector<ArrayView<float>, kSmallConstant>

which is much more convenient. We already had the option of returning e.g.

  std::vector<ArrayView<float>>

but that would bloat our binary with code to handle heap allocations
in places we'd rather be lean and mean.

https://godbolt.org/z/r-vcPj demonstrates that the overhead compared to
a raw C array + a size is ~zero.

Bug: webrtc:11391
Change-Id: Ifb6d937193052588be641aa62cc67ba0ec64ded6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168944
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30663}
2020-03-02 20:45:58 +00:00
Sebastian Jansson
db5d7e470f Cleanup: Use common IP overhead definitions in test and prod code
This avoid duplication. As part of this moving the overhead calculation
to the IP address class so it's easier to find and more natural to use.

Bug: webrtc:9883
Change-Id: If4d865f445bc1a302572896932966ce30294e339
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169445
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30657}
2020-03-02 11:36:58 +00:00
Niels Möller
dbf5416a80 Delete header file rtc_base/memory/aligned_array.h
Move definition of AlignedArray to the only code using it, the
test-only LappedTransform class, and delete unused methods.

Bug: webrtc:6424, webrtc:9577
Change-Id: I1bb5f57400f7217345b7ec7376235ad4c4bae858
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168701
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30576}
2020-02-20 14:55:25 +00:00