After recent refactorings, PeerConnection.tls_cert_verifier_ is
now both const and only accessed on the network thread, so it is
doubly thread-safe. Marking as such.
Bug: webrtc:9987
Change-Id: I2f924ecf2afe364d1e4b7f740435443bc53e4d0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191486
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32545}
And add a Create() method to the class.
This makes it possible to experiment with subclassing the
SdpOfferAnswer object without modifying the PeerConnection.
Bug: webrtc:11995
Change-Id: I0a7c91a8999858ddcb1ea59ac4eb9a3b0663b0f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190288
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32501}
These can now be initialized in the constructor and are not touched
explicitly in the destructor.
Bug: none
Change-Id: I3d294b15463a8d02bbe7e37fb14eefd017d5c1e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190284
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32494}
This allows making more members (including IsUnifiedPlan) const in a future CL.
Also revises the test for ReportUsageHistogram to use a configuration member
variable rather than a hook function in PeerConnectionFactory.
Bug: webrtc:12079
Change-Id: I6f1af7d6164c8a0d8466f76378a925d72d57d685
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190280
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32485}
As discussed on a design review, the name RoboCaller is not clear
enough and switching to CallbackList will provide readability benefits.
Bug: webrtc:11943
Change-Id: I010cf0a91b5323e4e9c96b83703be7af1e67439c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190142
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32478}
Also move ssrc_generator and audio/video options, as well as some
signal handling that's related.
These variables were not referenced in peer_connection.cc any more.
Bug: webrtc:11995
Change-Id: I29f8661afad488380d256220b35330233e8233e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189967
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32471}
This factory is only used by SdpOfferAnswerHandler, so it should not
need to depend on PeerConnection.
Bug: webrtc:11995
Change-Id: Ib27d9d9fdf440be7db8890bf0e7520d0c67bde22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189780
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32460}
The Google C++ style guide says that when both use a declaration, both
the .h file and the .cc file should include the relevant header.
Bug: webrtc:12057
Change-Id: I4c01ce8930d73418cb23c7fe1bb7bcd12c1e2568
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189541
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32435}
Also adds a script that runs iwyu to the tools_webrtc directory.
Bug: webrtc:11995
Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32431}
This enables modules that share the resources to reuse the connection
context object but not take a dependency on PeerConnectionFactory.
Bug: webrtc:11967
Change-Id: Ic68cbf061b3226f02f8638abd79ad881e89951d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188120
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32412}
This makes it easier to see that the tying of tracks
to streams affects only the SDP negotiation, and not
what's sent on the wire.
Bug: webrtc:11995
Change-Id: I8ca5adf0050e4a2be55d164a6d0e4d5811582476
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187359
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32368}
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}
the UMA stats currently do not count services like Hangouts that
have "complex" SDP with multiple tracks only in the answer, not in the
offer. Note that this changes the definition of the existing metric.
BUG=chromium:857004
Change-Id: Ib4520a82f7d94cdd4a307d32846e2d26a5f03b90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186701
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#32355}
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}
This prevents having to have sdp_offer_answer depend on peer_connection
for the messaging functions.
Bug: webrtc:11995
Change-Id: Icad7c9c0e6149bd1b8d78e37eff5f9786b74692e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186662
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32310}
After this CL, sdp_offer_answer is bigger than peer_connection.
Bug: webrtc:11995
Change-Id: Ie923fabf836de46fa939fe6fd7b3d936bbc85dab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186380
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32301}
Also use accessors for the last few member variable references
in PeerConnection.
This completes removing the variable accesses from SdpOfferAnswerHandler
to PeerConnection.
Bug: webrtc:11995
Change-Id: I70c78b43035c15f20559f7a6a5b50c3a613fe907
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186200
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32272}
These are functions that are called only from SdpOfferAnswer,
or that logically belong in the SdpOfferAnswer class.
Bug: webrtc:11995
Change-Id: I92136ee84e20e50957814c21b041ca152a2acca4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186268
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32271}
The eventual goal is to replace sigslot entirely, but we need to
start small, tread carefully, and evaluate how it works out.
Also add a few more RoboCaller unit tests to cover the types we
now use with RoboCaller.
Change-Id: I9a5814d1668a37546ea484ca88ec9c2be1913d25
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184660
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32266}
This component is heavily referenced by both PeerConnection and
SdpOfferAnswerHandler; it's likely that it will end up in
SdpOfferAnswerHandler.
Encapsulation makes it easier to move around.
Bug: webrtc:11995
Change-Id: I5329d9a90159d203510bf3698962cd246eea7324
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32229}
SignalDtlsSrtpSetupFailure is never fired, so the setup code for it,
is dead code. Also removing declarations for methods that have no
implementation.
For other public signals in BaseChannel I've added an accessor which
has revealed a threading problem due to the member variable being public.
Bug: webrtc:11994
Change-Id: Iec6046c6a598066b92c956002ba4160708ae7dcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32211}
The callback that the asyncinvoker was being used for, will now use
a safety flag to check if call_ is valid before issuing calls.
Using the flag is a step towards removing the call_ptr_ variable
but in this CL we're just looking at replacing use of the async invoker.
The safety flag is cleared at the same time as call_ is, which prevents
pending callbacks for that call instance from running.
Also adding TODOs related to this change that will be
followed upon in other CLs.
Bug: webrtc:11988, webrtc:11992, webrtc:11993
Change-Id: If3986758af6d01d39b2db0cce82e57fc48be9d7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185508
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32208}
This is a reland of 6f4de80ddddcc05beaced31146ffb753258bc7be
The blocking issue in Chromium is fixed.
Original change's description:
> Remove stopped transceivers at both local and remote SetDescription
>
> This should ensure that the correct number of senders and receivers
> are shown.
>
> Bug: webtc:11840
> Change-Id: Id57f8f9b1ceb8900abb3f92bcae79e5f0341de15
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184606
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32158}
Bug: webtc:11840
Change-Id: Iae8ca01e3f834694dacb36320858096b26f0996b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32181}
This reverts commit 6f4de80ddddcc05beaced31146ffb753258bc7be.
Reason for revert: Causes breakage in WebRTC roll (WPT tests)
Original change's description:
> Remove stopped transceivers at both local and remote SetDescription
>
> This should ensure that the correct number of senders and receivers
> are shown.
>
> Bug: webtc:11840
> Change-Id: Id57f8f9b1ceb8900abb3f92bcae79e5f0341de15
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184606
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32158}
TBR=hbos@webrtc.org,hta@webrtc.org
Change-Id: Ib91d59f506087dd96c5678262bac7c1580736dcf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webtc:11840
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185053
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32166}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/174261
Patchset 1 contains the old cl (plus a merge conflict fix).
Later patchets are bufixes: A PeerConnection can be created without a
Call instance (in the case of DataChannel only), so we can't always
use that to fetch the current trials.
Old CL descritpion:
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
// Since re-land is otherwise unchanged, setting previous reviewers as TBR
TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Bug: webrtc:11926
Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32163}
This should ensure that the correct number of senders and receivers
are shown.
Bug: webtc:11840
Change-Id: Id57f8f9b1ceb8900abb3f92bcae79e5f0341de15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184606
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32158}
RequestEncoderFallback, RequestEncoderSwitch and
SetVideoCodecSwitchingEnabledRequest are now all called on the
worker thread. Before, the work already happened on that thread but
WebRtcVideoChannel adapted internally when needed.
With this CL, there are thread checks to make sure that these calls are
always made the same way, we don't need the async invoker and there
are fewer calls out from the encoder thread in VideoStreamEncoder
(reducing the chance of unintentional blocking).
Bug: webrtc:11908
Change-Id: If8738bc2a708a0fefc6fe850b32655f049f30bdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184603
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32151}
As documented in webrtc:11908 this cleanup is fairly invasive and
when a part of a frequently executed code path, can be quite costly
in terms of performance overhead. This is currently the case with
synchronous calls between threads (Thread) as well with our proxy
api classes.
With this CL, all code in WebRTC should now either be using MessageHandlerAutoCleanup
or calling MessageHandler(false) explicitly.
Next steps will be to update external code to either depend on the
AutoCleanup variant, or call MessageHandler(false).
Changing the proxy classes to use TaskQueue set of concepts instead of
MessageHandler. This avoids the perf overhead related to the cleanup
above as well as incompatibility with the thread policy checks in
Thread that some current external users of the proxies would otherwise
run into (if we were to use Thread::Send() for synchronous call).
Following this we'll move the cleanup step into the AutoCleanup class
and an RTC_DCHECK that all calls to the MessageHandler are setting
the flag to false, before eventually removing the flag and make
MessageHandler pure virtual.
Bug: webrtc:11908
Change-Id: Idf4ff9bcc8438cb8c583777e282005e0bc511c8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183442
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32049}
This fixes some edge cases where early media could cause default
stream that block the actual signaled media from beind delivered.
Bug: webrtc:11477
Change-Id: I8b26df63a690861bd19f083102d1395e882f8733
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183120
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32030}
This is to allow testing without using the singleton sctp library.
cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure.
Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation.
patch 1 contain the original cl.
patch 2 modifications
Bug: none
Change-Id: Ic088da3eb7d9aada79e6d601dbf2d1aa2be777f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182840
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32024}
It was only being called for the first video media channel; with
unified plan SDP mode, it's possible to have multiple video media
channels, one for each video m= section.
Bug: webrtc:10795
Change-Id: I57fda9383d0f8803df1937ac5103d9ae354c0748
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182404
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32010}
This reverts commit 4c0a381137c04fd80830af8a041e25e3428dd33f.
Reason for revert: Breaks downstream test
Original change's description:
> Make cricket::SctpTransportInternalFactory injectable through PeerConnectionFactory Deps
>
> This is to allow testing without using the singleton sctp library.
> cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure.
> Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation.
>
> Bug: none
> Change-Id: I482241269463595062548870750d33f31238c6b1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182082
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32007}
TBR=deadbeef@webrtc.org,mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I46d5ba89fe723caccd065b0ac41d77ed45373838
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32008}
This is to allow testing without using the singleton sctp library.
cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure.
Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation.
Bug: none
Change-Id: I482241269463595062548870750d33f31238c6b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182082
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32007}
Also add an unit test for RtpTransceiver under Unified Plan, and
refactor so that we no longer use StopInternal() internally.
This will make removing it easier.
Bug: chromium:980879
Change-Id: I46219112e3aba8e7513c08336b10e95b1ea5d68b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182681
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31999}
This CL generates "negotiationneeded" events if negotiation is needed
when the Operations Chain becomes empty. This is only implemented in
Unified Plan to avoid Plan B regressions (the event is pretty useless
in Plan B as it fires repeatedly).
In order to implement the spec-compliant behavior of only firing the
event when the chain is empty, this CL introduces
PeerConnectionObserver::OnNegotiationNeededEvent() and
PeerConnectionInterface::ShouldFireNegotiationNeededEvent() to allow
validating the event before firing it. This is needed because the event
must not be fired until a task has been posted and subsequently chained
operations could invalidate it in the meantime.
Test coverage is added for both legacy and modern "negotiationneeded"
events.
Bug: chromium:1060083
Change-Id: I1dbaa8f6ddb1c6e7c8abd8da3b92efcb64060383
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180620
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31989}
