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}
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
Bug: webrtc:11926
Change-Id: I70f28c4fbabf6d9e55052342000e38612b46682c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174261
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32129}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
Done in preparation for some threading changes that would be quite
messy if implemented with the class as-is.
This results in some code duplication, but is preferable to
one class having two completely different modes of operation.
RTP data channels are in the process of being removed anyway,
so the duplicated code won't last forever.
Bug: webrtc:9883
Change-Id: Idfd41a669b56a4bb4819572e4a264a4ffaaba9c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178940
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31691}
This ensures with DCHECK-crashes that we don't accidentally do more
blocking invokes than we think.
Remaining blocking invokes FYI:
- PrepareTransceiverStatsInfos_s_w() does 1 blocking invoke (regardless
of the number of transceivers or channels) to the worker thread. This
is because VoiceMediaChannel, VideoMediaChannel and GetParameters()
execute on the worker thread, and the result of these operations are
needed on the signalling thread.
- pc_->GetCallStats() does 1 blocking invoke to the worker thread.
These two blocking invokes can be merged, reducing the total number of
blocking invokes from 2 to 1, but this CL does not attempt to do that.
I filed https://crbug.com/webrtc/11767 for that.
Bug: webrtc:11716
Change-Id: Iebc2ab350d253fd037211cdd283825b4e5b2d446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178867
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31670}
This reverts commit f79bfc65e52a35d27cf0db2d212e94043fb44da3.
Reason for revert: Potentially affects Chromium tests, see
failures on https://chromium-review.googlesource.com/c/chromium/src/+/2276338.
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}
TBR=hbos@webrtc.org,philipp.hancke@googlemail.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: None
Change-Id: I6df20c93f6944b819eb11f22ba30c6221de61d79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178560
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31610}
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}
This removes code from DataChannelController that exposes
an internal vector of data channels and puts the onus of
returning stats for a data channel, on the data channel
object itself. This will come in handy as we make threading
changes to the data channel object.
Change-Id: Ie164cc5823cd5f9782fc5c9a63aa4c76b8229639
Bug: webrtc:11547, webrtc:11687
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177244
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31533}
Add a few DCHECKs and comments about upcoming work.
Bug: webrtc:11547
Change-Id: I2d42f48cb93f31e70cf9fe4b3b62241c38bc9d8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177106
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31530}
This reverts commit 71db9acc4019b8c9c13b14e6a022cbb3b4255b09.
Reason for revert: breaks downstream project.
Reason for force push: win bot broken.
Original change's description:
> RtpTransceiverInterface: introduce SetOfferedRtpHeaderExtensions.
>
> This change adds exposure of a new transceiver method for
> modifying the extensions offered in the next SDP negotiation,
> following spec details in https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface.
>
> Features:
> - The interface allows to control the negotiated direction as
> per https://tools.ietf.org/html/rfc5285#page-7.
> - The interface allows to remove an extension from SDP
> negotiation by modifying the direction to
> RtpTransceiverDirection::kStopped.
>
> Note: support for signalling directionality of header extensions
> in the SDP isn't implemented yet.
>
> https://chromestatus.com/feature/5680189201711104.
> Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk
>
> Bug: chromium:1051821
> Change-Id: Iaabc34446f038c46d93c442e90c2a77f77d542d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176408
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31487}
TBR=hta@webrtc.org,handellm@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: true
Bug: chromium:1051821
Change-Id: I70e1a07225d7eeec7480fa5577d8ff647eba6902
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177103
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31516}
This unblocks injecting platform-specific resources, such as power
usage signals in Chrome.
This CL adds AddAdaptationResource to PeerConnectionInterface and
integration tests verifying that if an injected resource is overusing,
resolution will soon be reduced.
To aid testing, some testing-only classes have been updated.
Bug: webrtc:11525
Change-Id: I820099e79f18d910fd641ee1412ad064b99ebce9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177003
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31505}
This patch fixes a minor bug in the implementation of
surface_ice_candidates_on_ice_transport_type_changed. The existing
implementation correctly handles the surfacing, but accidentally also
set the SetNeedsIceRestartFlag, which made _next_ offer contain
a ice restart.
Modified existing testcase to verify this.
Bug: webrtc:8939
Change-Id: If566e3249296467668627e5941495f6036cbd903
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176127
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31363}
This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d.
Reason for revert: The original CL is updated in PS #2 to
fix the googRtt issue which was that when the legacy sender
stats were put in "aggregated_senders" we forgot to update
rtt_ms the same way that we do it for "senders".
Original change's description:
> Revert "Improve outbound-rtp statistics for simulcast"
>
> This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.
>
> Reason for revert: Breaks googRtt in legacy getStats API
>
> Original change's description:
> > Improve outbound-rtp statistics for simulcast
> >
> > Bug: webrtc:9547
> > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Eldar Rello <elrello@microsoft.com>
> > Cr-Commit-Position: refs/heads/master@{#31097}
>
> TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:9547
> Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31165}
TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
# Not skipping CQ checks because this is a reland.
Bug: webrtc:9547
Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31169}
