Commit Graph

1655 Commits

Author SHA1 Message Date
fe132e6bd9 Don't expect a transceiver for stopped m-sections
After implementing transceiver.stop and associated logic with regard
to stopped media sections, there might not be a transceiver for every
media section. Allow this case.

There is a test ready for submission in Chrome:
https://chromium-review.googlesource.com/c/chromium/src/+/2410407

Bug: chromium:1127625
Change-Id: I150ea5f0da4a0cbd2bf214bc659ea0df93b607de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184343
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32117}
2020-09-16 13:49:51 +00:00
6da271844c Avoid deallocating the async invoker when clearing the transport.
Deallocating the async invoker is a costly operation
but it's also unnecessary and could cause us to miss signal
events.

The data_channel_transport and data_channel_transport_invoker
are (despite the name) not related, since the latter is
used to signal events on the signaling thread whereas the
former deals with the data.

Bug: webrtc:11908
Change-Id: I37b345476a6381aef5d87807877ec1e05b380137
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184062
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32096}
2020-09-14 12:51:12 +00:00
6b4d962947 Fix standard GetStats to not modify NetEq state.
Add a get_and_clear_legacy_stats flag to AudioReceiveStream::GetStats,
to distinguish calls from standard GetStats and legacy GetStats.

Add const method NetEq::CurrentNetworkStatistics to get current
values of stateless NetEq stats. Standard GetStats will then call this
method instead of NetEq::NetworkStatistics.

Bug: webrtc:11622
Change-Id: I3833a246a9e39b18c99657a738da22c6e2bd5f5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32092}
2020-09-14 09:51:21 +00:00
c8850cbf55 Change gtest name to allow filtering based on the story name.
It is meant for Pinpoint to run only the relevant tests when running a bisection.
The Pinpoint side of this change can be found here:
https://crrev.com/c/2404161

Bug: webrtc:11084
Change-Id: I466f39816b83e2f83a3a49845c99605f4d5a857b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183763
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32082}
2020-09-11 14:11:27 +00:00
dd68063976 rename "sdp" to description in a few places
renames the RTCSessionDescription object from "ѕdp" to "desc" in a few places.
The term SDP should generally refer to the blob of text described in
RFC 4566 while the RTCSessionDescription specified in
  https://w3c.github.io/webrtc-pc/#rtcsessiondescription-class
contains both a type and a sdp.

BUG=None

Change-Id: Iacf332d02b03134e49c2b4147dc5725affa89741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183882
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32080}
2020-09-11 12:36:54 +00:00
fc83cdc819 Avoid proxy thread hops for reading const properties.
This bypasses the proxy for the following properties:
* MediaStream::id()
* AudioTrack::kind() and AudioTrack::id()
* VideoTrack::kind() and VideoTrack::id()
* RtpReceiver::media_type() and RtpReceiver::id()
* RtpSender::media_type() and RtpSender::id()
* VideoTrackSource::remote() and VideoTrackSource::is_screencast()
* RtpTransceiver::media_type()

Bug: webrtc:11923
Change-Id: If7edea1781f778af3775515fc4af9a9e151c8103
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183767
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32071}
2020-09-10 13:11:44 +00:00
abdb470d00 Make MessageHandler cleanup optional.
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}
2020-09-07 12:57:15 +00:00
c03a187391 Default streams: don't block media even if on different transceiver.
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}
2020-09-02 22:28:55 +00:00
2bca008914 Reland "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.

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}
2020-09-01 12:17:00 +00:00
9e02f4716a Fix destruction order of PortAllocator and PacketSocketFactory.
PortAllocator depends on PacketSocketFactory, so it should be deleted
afterwords in case its created sockets depend on the resources owned
by the factory.

Bug: None
Change-Id: I7716c552d371b78360db656cc2f4fd03415d0e00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182881
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32020}
2020-08-31 21:52:27 +00:00
6b381c7456 Call SetVideoCodecSwitchingEnabled on every video media channel.
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}
2020-08-27 21:08:28 +00:00
1f580a97e5 Revert "Make cricket::SctpTransportInternalFactory injectable through PeerConnectionFactory Deps"
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}
2020-08-27 13:59:57 +00:00
4c0a381137 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}
2020-08-27 13:19:14 +00:00
c75c428076 Fix current_direction() when stopping_ but not stopped_
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}
2020-08-26 14:02:03 +00:00
e574a31c50 [Perfect Negotiation] Fire onnegotiationneeded when chain is empty.
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}
2020-08-25 09:56:39 +00:00
bedb605c82 Transition ICE gathering state to "new" once all transports go away
Bug: chromium:1115080
Change-Id: I524ed48ffc2520ce21ad4bdc25fa3b86d9e41af5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182081
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31976}
2020-08-20 18:55:52 +00:00
fcf5e7b131 Make Objective-C interface use SetDirectionWithError
Also moves implementation of legacy setDirection() without error to the
api/ directory.

This is one step in the plan for changing the API
to return RTCError.

Bug: chromium:980879
Change-Id: Ibce8edf8e3c6d41de7ce49d2ffc33f5b282a0e9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181520
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31943}
2020-08-17 10:01:49 +00:00
c2cfd18ab8 Reland "peerconnection: prefer spec names for signaling state"
This is a reland of f79bfc65e52a35d27cf0db2d212e94043fb44da3
the tests that have blocked the roll have been marked as allowed to fail.

Original change's description:
> peerconnection: prefer spec names for signaling state
>
> Map the internal state names to the spec ones defined in
>   https://w3c.github.io/webrtc-pc/#rtcsignalingstate-enum
> instead of exposing them. This only affects the (not specified)
> error strings.
>
> Bug: None
> Change-Id: Ib0b35bb3106b1688e8386f6fdd0b8c7fdebaf1dc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178390
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#31591}

Bug: chromium:1101699
Change-Id: Ia21cec9e76fbaa4df2fa5a80409a7c80fedc4faa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178562
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31914}
2020-08-11 15:44:00 +00:00
6060df5948 Reland "Implement transceiver.stop()"
This is a reland of 11dc6571cb4ff3e71dee1557dfff8d9076e108d3

One fix that makes Web Platform Tests pass in debug mode is applied.

Original change's description:
> Implement transceiver.stop()
>
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
>
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
>
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}

Bug: chromium:980879
Change-Id: Ide31d929ac5ea118d83fdf6a35a592af23f7dfa7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181263
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31907}
2020-08-11 10:46:23 +00:00
a88c9776de Revert "Implement transceiver.stop()"
This reverts commit 11dc6571cb4ff3e71dee1557dfff8d9076e108d3.

Reason for revert: Breaks Chromium WPT tests

Original change's description:
> Implement transceiver.stop()
> 
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
> 
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
> 
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
> 
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}

TBR=sakal@webrtc.org,kthelgason@webrtc.org,hta@webrtc.org,guidou@webrtc.org,marinaciocea@webrtc.org

Change-Id: Ibdc24f7d41e481293ca74ba6d1572de64f7e4654
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:980879
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181262
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31897}
2020-08-10 18:06:30 +00:00
11dc6571cb Implement transceiver.stop()
This adds RtpTransceiver.StopStandard(), which behaves according to
the specification at
https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop

It modifies RTCPeerConnection.getTransceivers() to return only
transceivers that have not been stopped.

Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762

Bug: chromium:980879
Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31893}
2020-08-10 13:29:15 +00:00
c88fe70a8d Make Android/iOS local/remote description accessors thread safe.
Since the descriptions can be modified on the signaling thread,
ToString can only be safely called on that thread.

Bug: webrtc:11791
Change-Id: Icf6aada8aa66d00be94c6bda7b22e41b5d3bbc17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180541
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31862}
2020-08-05 22:34:46 +00:00
fbbfc02698 sdp: reject sdp with malformed b= lines
BUG=webrtc:3782

Change-Id: I3d137b0b74565f7e85bbc6b453e73731a94c2b04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179360
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31838}
2020-08-03 18:21:14 +00:00
239ac8a4e2 Reland "Pass NetworkMonitorFactory through PeerConnectionFactory."
This is a reland of 003c9be817817ed0e3aef3f50c78ae5cb31bc0ff

Found some downstream code that relies on
NetworkMonitorFactory::SetFactory, so I'm adding those methods back
temporarily. BasicNetworkManager will fall back to the static factory
if the one passed into PeerConnectionFactory is null.

Original change's description:
> Pass NetworkMonitorFactory through PeerConnectionFactory.
>
> Previously the instance was set through a static method, which was
> really only done because it was difficult to add new
> PeerConnectionFactory construction arguments at the time.
>
> Now that we have PeerConnectionFactoryDependencies it's easy to clean
> this up.
>
> I'm doing this because I plan to add a NetworkMonitor implementation
> for iOS, and don't want to inherit this ugliness.
>
> Bug: webrtc:9883
> Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31815}

TBR=hta@webrtc.org, sakal@webrtc.org

Bug: webrtc:9883
Change-Id: I2e817c423f21936f87532a9694eb9a0a1b70c212
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180722
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31824}
2020-08-01 00:36:27 +00:00
cfba4ffe31 Revert "Reland "Pass NetworkMonitorFactory through PeerConnectionFactory.""
This reverts commit 7ded73351870bfb45160fa6b9db71a94fe49397b.

Reason for revert: Found more code calling NetworkMonitorFactory::SetFactory...

Original change's description:
> Reland "Pass NetworkMonitorFactory through PeerConnectionFactory."
> 
> This is a reland of 003c9be817817ed0e3aef3f50c78ae5cb31bc0ff
> 
> Original change's description:
> > Pass NetworkMonitorFactory through PeerConnectionFactory.
> >
> > Previously the instance was set through a static method, which was
> > really only done because it was difficult to add new
> > PeerConnectionFactory construction arguments at the time.
> >
> > Now that we have PeerConnectionFactoryDependencies it's easy to clean
> > this up.
> >
> > I'm doing this because I plan to add a NetworkMonitor implementation
> > for iOS, and don't want to inherit this ugliness.
> >
> > Bug: webrtc:9883
> > Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Commit-Queue: Taylor <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31815}
> 
> TBR=hta@webrtc.org, sakal@webrtc.org
> 
> Bug: webrtc:9883
> Change-Id: Ibf69a22e8f94226908636c7d50ff9eda65bd4129
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180720
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31822}

TBR=deadbeef@webrtc.org,sakal@webrtc.org,hta@webrtc.org

Change-Id: Iae51b94072cec9abc021eed4e51d1fbeee998adc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180721
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31823}
2020-07-31 22:44:39 +00:00
7ded733518 Reland "Pass NetworkMonitorFactory through PeerConnectionFactory."
This is a reland of 003c9be817817ed0e3aef3f50c78ae5cb31bc0ff

Original change's description:
> Pass NetworkMonitorFactory through PeerConnectionFactory.
>
> Previously the instance was set through a static method, which was
> really only done because it was difficult to add new
> PeerConnectionFactory construction arguments at the time.
>
> Now that we have PeerConnectionFactoryDependencies it's easy to clean
> this up.
>
> I'm doing this because I plan to add a NetworkMonitor implementation
> for iOS, and don't want to inherit this ugliness.
>
> Bug: webrtc:9883
> Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31815}

TBR=hta@webrtc.org, sakal@webrtc.org

Bug: webrtc:9883
Change-Id: Ibf69a22e8f94226908636c7d50ff9eda65bd4129
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180720
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31822}
2020-07-31 22:05:02 +00:00
7d627545cb Revert "Pass NetworkMonitorFactory through PeerConnectionFactory."
This reverts commit 003c9be817817ed0e3aef3f50c78ae5cb31bc0ff.

Reason for revert: Breaks downstream build which is still using
SetFactory/ReleaseFactory. Probably will need to update this in lockstep.

Original change's description:
> Pass NetworkMonitorFactory through PeerConnectionFactory.
> 
> Previously the instance was set through a static method, which was
> really only done because it was difficult to add new
> PeerConnectionFactory construction arguments at the time.
> 
> Now that we have PeerConnectionFactoryDependencies it's easy to clean
> this up.
> 
> I'm doing this because I plan to add a NetworkMonitor implementation
> for iOS, and don't want to inherit this ugliness.
> 
> Bug: webrtc:9883
> Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31815}

TBR=deadbeef@webrtc.org,sakal@webrtc.org,hta@webrtc.org

Change-Id: I1f09df7be9c860017d515e5a87488340afa6eda6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180640
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31818}
2020-07-31 07:34:42 +00:00
ee8c246be7 Reland "sdp: parse and serialize b=TIAS"
This reverts commit 20b701f3d79c499b0981f03fbf3a9b0fe531ac5d.

Reason for reland: Reverting did not affect the test regression.

Original change's description:
> Revert "sdp: parse and serialize b=TIAS"
>
> This reverts commit c6801d4522ab94f965e258e68259fde312023654.
>
> Reason for revert: Speculatively reverting since it possibly breaks downstream performance test.
>
> One issue I noticed is that the correct SDP won't be produced if set_bandwidth_type hasn't been called. Probably should default to b=AS in that case.
>
> Original change's description:
> > sdp: parse and serialize b=TIAS
> >
> > BUG=webrtc:5788
> >
> > Change-Id: I063c756004e4c224fffa36d2800603c7b7e50dce
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179223
> > Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> > Reviewed-by: Taylor <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31729}
>
> TBR=deadbeef@webrtc.org,hta@webrtc.org,minyue@webrtc.org,philipp.hancke@googlemail.com,jleconte@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:5788
> Change-Id: I2a3f676b4359834e511dffd5adedc9388e0ea0f8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179620
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31762}

TBR=nisse@webrtc.org

Bug: webrtc:5788
Change-Id: I5c0ef29d275bb2264d9b706b085f7933d59e2801
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179760
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31816}
2020-07-30 21:16:08 +00:00
003c9be817 Pass NetworkMonitorFactory through PeerConnectionFactory.
Previously the instance was set through a static method, which was
really only done because it was difficult to add new
PeerConnectionFactory construction arguments at the time.

Now that we have PeerConnectionFactoryDependencies it's easy to clean
this up.

I'm doing this because I plan to add a NetworkMonitor implementation
for iOS, and don't want to inherit this ugliness.

Bug: webrtc:9883
Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31815}
2020-07-30 20:52:28 +00:00
c6cf902034 Improves logging in MediaChannel
This CL changes the style of logging for an API which is essential when
WebRTC is used in Chrome. By changing the format, we can more easily
tie in (search for tags etc.) logs from WebRTC with logs in Chrome.
See e.g.
https://chromium-review.googlesource.com/c/chromium/src/+/2093443
for more details.

I decided to use a new private method to avoid using rtc::StringBuilder.
The idea was to make the log statements less complex and more condensed.

Tbr: mbonadei
Bug: webrtc:11493
Change-Id: I46b4a933ad62ac1db376743b4a41b62c5f8c6ac6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172841
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31808}
2020-07-30 08:10:03 +00:00
397cd82eaf Create port allocator on signaling thread and init on network
Port allocator can be created on one thread and then initialized and
used on another. So we can avoid sync invoke to network thread to create
port allocator.

Bug: webrtc:11799
Change-Id: I5020093a41acbf7e372f2e4970e016ce14a7f406
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180122
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31805}
2020-07-29 11:31:43 +00:00
831ae4ef65 Reland "[Perfect Negotiation] Implement non-racy version of SetLocalDescription."
This is a reland of d4089cae47334a4228b69d6bb23f2e49ebb7496e
with the following fix:

Invoke MaybeStartGathering as the last step of DoSetLocalDescription.
This ensures that candidates and onicegatheringstatechange does not
happen before SLD is resolved. This is important for passing
external/wpt/webrtc/RTCPeerConnection-iceGatheringState.html.

Original change's description:
> [Perfect Negotiation] Implement non-racy version of SetLocalDescription.
>
> BACKGROUND
>
> When SLD is invoked with SetSessionDescriptionObserver, the observer is
> called by posting a message back to the execution thread, delaying the
> call. This delay is "artificial" - it's not necessary; the operation is
> already complete. It's a post from the signaling thread to the signaling
> thread. The rationale for the post was to avoid the observer making
> recursive calls back into the PeerConnection. The problem with this is
> that by the time the observer is called, the PeerConnection could
> already have executed other operations and modified its states.
>
> This causes the referenced bug: one can have a race where SLD is
> resolved "too late" (after a pending SRD is executed) and the signaling
> state observed when SLD resolves doesn't make sense.
>
> When implementing Unified Plan, we fixed similar issues for SRD by
> adding a version that takes SetRemoteDescriptionObserverInterface as
> argument instead of SetSessionDescriptionObserver. The new version did
> not have the delay. The old version had to be kept around not to break
> downstream projects that had dependencies both on he delay and on
> allowing the PC to be destroyed midst-operation without informing its
> observers.
>
> THIS CL
>
> This does the old SRD fix for SLD as well: A new observer interface is
> added, SetLocalDescriptionObserverInterface, and
> PeerConnection::SetLocalDescription() is overloaded. If you call it with
> the old observer, you get the delay, but if you call it with the new
> observer, you don't get a delay.
>
> - SetLocalDescriptionObserverInterface is added.
> - SetLocalDescription is overloaded.
> - The adapter for SetSessionDescriptionObserver that causes the delay
>   previously only used for SRD is updated to handle both SLD and SRD.
> - FakeSetLocalDescriptionObserver is added and
>   MockSetRemoteDescriptionObserver is renamed "Fake...".
>
> Bug: chromium:1071733
> Change-Id: I920368e648bede481058ac22f5b8794752a220b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179100
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31798}

TBR=hta@webrtc.org

Bug: chromium:1071733
Change-Id: Ic6e8d96afa1c19604762f373716c08dbfa9d178c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180481
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31804}
2020-07-29 11:27:43 +00:00
3872873889 Reduce log level in BaseChannel::SendPacket.
This log line is causing test failures due to excessive logging (see
referenced bug); reducing log level.

Bug: chromium:984879
Change-Id: Ic94ba0a39b91b4253a58ad54de0cba1ca49882e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175913
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31803}
2020-07-29 09:57:24 +00:00
4c9c75a2a6 Revert "[Perfect Negotiation] Implement non-racy version of SetLocalDescription."
This reverts commit d4089cae47334a4228b69d6bb23f2e49ebb7496e.

Reason for revert: Breaks chromium WPT that is timing sensitive to onicegatheringstatechanges.
This CL accidentally moved the MaybeStartGatheringIceCandidates to after completing the SLD call. The fix is to move it back. I'll do that in a re-land.

Original change's description:
> [Perfect Negotiation] Implement non-racy version of SetLocalDescription.
> 
> BACKGROUND
> 
> When SLD is invoked with SetSessionDescriptionObserver, the observer is
> called by posting a message back to the execution thread, delaying the
> call. This delay is "artificial" - it's not necessary; the operation is
> already complete. It's a post from the signaling thread to the signaling
> thread. The rationale for the post was to avoid the observer making
> recursive calls back into the PeerConnection. The problem with this is
> that by the time the observer is called, the PeerConnection could
> already have executed other operations and modified its states.
> 
> This causes the referenced bug: one can have a race where SLD is
> resolved "too late" (after a pending SRD is executed) and the signaling
> state observed when SLD resolves doesn't make sense.
> 
> When implementing Unified Plan, we fixed similar issues for SRD by
> adding a version that takes SetRemoteDescriptionObserverInterface as
> argument instead of SetSessionDescriptionObserver. The new version did
> not have the delay. The old version had to be kept around not to break
> downstream projects that had dependencies both on he delay and on
> allowing the PC to be destroyed midst-operation without informing its
> observers.
> 
> THIS CL
> 
> This does the old SRD fix for SLD as well: A new observer interface is
> added, SetLocalDescriptionObserverInterface, and
> PeerConnection::SetLocalDescription() is overloaded. If you call it with
> the old observer, you get the delay, but if you call it with the new
> observer, you don't get a delay.
> 
> - SetLocalDescriptionObserverInterface is added.
> - SetLocalDescription is overloaded.
> - The adapter for SetSessionDescriptionObserver that causes the delay
>   previously only used for SRD is updated to handle both SLD and SRD.
> - FakeSetLocalDescriptionObserver is added and
>   MockSetRemoteDescriptionObserver is renamed "Fake...".
> 
> Bug: chromium:1071733
> Change-Id: I920368e648bede481058ac22f5b8794752a220b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179100
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31798}

TBR=hbos@webrtc.org,hta@webrtc.org

Change-Id: Ie1e1ecc49f3b1d7a7e230db6d36decbc4cbe8c86
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1071733
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180480
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31802}
2020-07-29 09:46:56 +00:00
d4089cae47 [Perfect Negotiation] Implement non-racy version of SetLocalDescription.
BACKGROUND

When SLD is invoked with SetSessionDescriptionObserver, the observer is
called by posting a message back to the execution thread, delaying the
call. This delay is "artificial" - it's not necessary; the operation is
already complete. It's a post from the signaling thread to the signaling
thread. The rationale for the post was to avoid the observer making
recursive calls back into the PeerConnection. The problem with this is
that by the time the observer is called, the PeerConnection could
already have executed other operations and modified its states.

This causes the referenced bug: one can have a race where SLD is
resolved "too late" (after a pending SRD is executed) and the signaling
state observed when SLD resolves doesn't make sense.

When implementing Unified Plan, we fixed similar issues for SRD by
adding a version that takes SetRemoteDescriptionObserverInterface as
argument instead of SetSessionDescriptionObserver. The new version did
not have the delay. The old version had to be kept around not to break
downstream projects that had dependencies both on he delay and on
allowing the PC to be destroyed midst-operation without informing its
observers.

THIS CL

This does the old SRD fix for SLD as well: A new observer interface is
added, SetLocalDescriptionObserverInterface, and
PeerConnection::SetLocalDescription() is overloaded. If you call it with
the old observer, you get the delay, but if you call it with the new
observer, you don't get a delay.

- SetLocalDescriptionObserverInterface is added.
- SetLocalDescription is overloaded.
- The adapter for SetSessionDescriptionObserver that causes the delay
  previously only used for SRD is updated to handle both SLD and SRD.
- FakeSetLocalDescriptionObserver is added and
  MockSetRemoteDescriptionObserver is renamed "Fake...".

Bug: chromium:1071733
Change-Id: I920368e648bede481058ac22f5b8794752a220b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179100
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31798}
2020-07-28 10:05:57 +00:00
20b701f3d7 Revert "sdp: parse and serialize b=TIAS"
This reverts commit c6801d4522ab94f965e258e68259fde312023654.

Reason for revert: Speculatively reverting since it possibly breaks downstream performance test.

One issue I noticed is that the correct SDP won't be produced if set_bandwidth_type hasn't been called. Probably should default to b=AS in that case.

Original change's description:
> sdp: parse and serialize b=TIAS
> 
> BUG=webrtc:5788
> 
> Change-Id: I063c756004e4c224fffa36d2800603c7b7e50dce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179223
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31729}

TBR=deadbeef@webrtc.org,hta@webrtc.org,minyue@webrtc.org,philipp.hancke@googlemail.com,jleconte@webrtc.org

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

Bug: webrtc:5788
Change-Id: I2a3f676b4359834e511dffd5adedc9388e0ea0f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179620
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31762}
2020-07-17 21:12:17 +00:00
3cb525b378 Rename CriticalSection to RecursiveCriticalSection.
This name change communicates that the recursive critical section
should not be used for new code.
The relevant files are renamed rtc_base/critical_section* ->
rtc_base/deprecated/recursive_critical_section*

Bug: webrtc:11567
Change-Id: I73483a1c5e59c389407a981efbfc2cfe76ccdb43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179483
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31754}
2020-07-17 09:19:50 +00:00
1a09faed62 Delete SignalDataChannelTransportNegotiated
This negotiation no longer takes place.

Bug: webrtc:9719
Change-Id: I33bd985105076fabf3200c31ea06b84b413794e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179363
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31753}
2020-07-17 08:36:00 +00:00
6b8271638b Delete unused enum values for DataChannelType
Bug: webrtc:9719
Change-Id: I2281636e3beaa2b0e59ac874b609e70e54d61cb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179365
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31752}
2020-07-17 08:28:20 +00:00
3d2210876e Remove unused critical section includes.
Bug: webrtc:11567
Change-Id: Ic5e43c51ce06c0619adc265d12ad4bef73a9df76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179521
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31745}
2020-07-16 13:52:28 +00:00
4c7bb27a10 Remove rtc::GlobalLock.
This change migrates a last stray consumer of GlobalLock
(SrtpSession) and removes all traces of GlobalLock/GlobalLockScope
from WebRTC.

Bug: webrtc:11567
Change-Id: I28059f2a10075815a4bdee8c357b9d3b6e50f18b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179361
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31736}
2020-07-15 20:45:13 +00:00
c6801d4522 sdp: parse and serialize b=TIAS
BUG=webrtc:5788

Change-Id: I063c756004e4c224fffa36d2800603c7b7e50dce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179223
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31729}
2020-07-15 08:01:06 +00:00
9ad1f6feca Reland "Delete PeerConnectionInterface::BitrateParameters"
This is a reland of e2dfe74b0e29558ddea6473d0272bc38c838370c
Downstream breakage has been fixed.

Original change's description:
> Delete PeerConnectionInterface::BitrateParameters
>
> Replaced by the api struct BitrateSettings, added in
> https://webrtc-review.googlesource.com/74020
>
> Bug: None
> Change-Id: I8b50b32f5c7a8918fad675904d913a21fd905274
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177665
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31714}

Tbr: kwiberg@webrtc.org
Bug: None
Change-Id: Ic039e51f9f842329525887a28d1cb9819addc74b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179282
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31728}
2020-07-15 07:35:16 +00:00
c888ffa57f Delete CompositeDataChannelTransport
And delete the always null members data_channel_transport_ and
composite_data_channel_transport_ from the JsepTransport class.

Bug: webrtc:9719
Change-Id: Ibfd92b74708d63a75521f6f1d5fbc3830bd67e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179280
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31727}
2020-07-15 06:54:06 +00:00
84bb634238 Delete legacy cricket::RtpHeaderExtension struct as unused
Bug: None
Change-Id: I8529475578a91173ca2e89e0bbbf186fc9d39472
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179222
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31722}
2020-07-14 08:55:02 +00:00
f60d4c2dfe Revert "Delete PeerConnectionInterface::BitrateParameters"
This reverts commit e2dfe74b0e29558ddea6473d0272bc38c838370c.

Reason for revert: Breaks downstream project

Original change's description:
> Delete PeerConnectionInterface::BitrateParameters
> 
> Replaced by the api struct BitrateSettings, added in
> https://webrtc-review.googlesource.com/74020
> 
> Bug: None
> Change-Id: I8b50b32f5c7a8918fad675904d913a21fd905274
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177665
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31714}

TBR=deadbeef@webrtc.org,ilnik@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org

Change-Id: Ia62b3c43996e95668d7972882baf06a186a539d3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179221
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31719}
2020-07-13 15:41:39 +00:00
e2dfe74b0e Delete PeerConnectionInterface::BitrateParameters
Replaced by the api struct BitrateSettings, added in
https://webrtc-review.googlesource.com/74020

Bug: None
Change-Id: I8b50b32f5c7a8918fad675904d913a21fd905274
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177665
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31714}
2020-07-13 10:06:42 +00:00
21621e9d08 Delete obsolete method JsepTransport::NegotiateDatagramTransport
Left-over from https://webrtc-review.googlesource.com/c/src/+/176500.

Bug: webrtc:9719
Change-Id: I9e4c9e149756c0ff194a374c002e7d5ac022cfbd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178202
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31712}
2020-07-13 08:52:58 +00:00
0800010dd6 peerconnection: remove old helper function
the TODO is obsolete, that code is only supported in plan-b mode and is a
one-liner.

BUG=webrtc:7600

Change-Id: I4e6c52c3a5b4cfff1b2d9185dedc786df9f474a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179066
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#31701}
2020-07-10 12:35:59 +00:00
edacbd53de Reland "Implement packets_(sent | received) for RTCTransportStats"
This is a reland of fb6f975401972635a644c0db06c135b4c0aaef4a. Related
issue in chromium is fixed here:
https://chromium-review.googlesource.com/c/chromium/src/+/2287294

Original change's description:
> Implement packets_(sent | received) for RTCTransportStats
>
> Bug: webrtc:11756
> Change-Id: Ic0caad6d4675969ef3ae886f50326e4a2e1cbfe7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178741
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31643}

Bug: webrtc:11756
Change-Id: I1e310e3d23248500eb7dabd23d0ce6c4ec4cb8c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178871
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31700}
2020-07-10 11:50:59 +00:00