Commit Graph

279 Commits

Author SHA1 Message Date
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
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
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
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
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
3a034e15b4 Split DataChannel into two separate classes for RTP and SCTP.
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}
2020-07-10 00:03:21 +00:00
e88c95e516 [Stats] Add more rtc::Thread::ScopedDisallowBlockingCalls to getStats().
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}
2020-07-08 10:55:30 +00:00
58e64bbf3b Revert "peerconnection: prefer spec names for signaling state"
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}
2020-07-02 09:10:37 +00:00
f79bfc65e5 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}
2020-06-30 13:40:26 +00:00
755c65d8b5 Reland 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

Tested: new unit tests in CL and manual tests with downstream project.
Bug: chromium:1051821
Change-Id: I7a4c2f979a5e50e88d49598eacb76d24e81c7c7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177348
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31554}
2020-06-24 10:38:30 +00:00
2a70703eb8 Delete MediaTransportInterface and DatagramTransportInterface
Bug: webrtc:9719
Change-Id: Ic9936a78ab42f4a9bb4cc3265f0a2cf36946558f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176500
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31536}
2020-06-17 08:41:14 +00:00
6476d0bf02 Consolidate creation of DataChannel proxy to a single place
Change-Id: I707733f521a4fda1536741b204a559dd511d0c00
Bug: webrtc:11547
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177344
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31535}
2020-06-17 07:06:34 +00:00
2e94de596e Add GetSctpStats to PeerConnectionInternal, remove sctp_data_channels()
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}
2020-06-16 16:36:42 +00:00
7d3cfbf90d Inject signaling and network threads to DataChannel.
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}
2020-06-16 10:22:19 +00:00
6f727da62b Revert "RtpTransceiverInterface: introduce SetOfferedRtpHeaderExtensions."
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}
2020-06-12 16:26:49 +00:00
4c1e7cc19b [Adaptation] Add ability to inject resources on the PeerConnection.
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}
2020-06-11 14:17:01 +00:00
9276e2c39b Remove enable_simulcast_stats config flag as not needed anymore
Bug: webrtc:9547
Change-Id: Ie50453aa3496d16bfadfc9fdd3e7e6982278cfba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176841
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31492}
2020-06-10 15:59:32 +00:00
71db9acc40 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}
2020-06-10 13:02:44 +00:00
e309651f33 Don't SetNeedsIceRestartFlag if widening candidate filter when surface_ice_candidates_on_ice_transport_type_changed
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}
2020-05-27 08:42:10 +00:00
a0ff50c031 Reland "Improve outbound-rtp statistics for simulcast"
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}
2020-05-05 20:22:19 +00:00
9a925c9ce3 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}
2020-05-05 13:38:51 +00:00
da6cda839d 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}
2020-04-17 11:28:00 +00:00
57cabed0b0 Replace std::string::find() == 0 with absl::StartsWith.
Bug: None
Change-Id: I070c4a5d19455f3a5c5d3ccc05f418545c351987
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172584
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30960}
2020-04-01 11:15:00 +00:00
00b46f7f2a PeerConnection owns the PacketSocketFactory dependency.
The PacketSocketFactory dependency (if present on the object passed to
CreatePeerConnection(...)) is given as a raw pointer to the
PortAllocator, but the unique_ptr remains in the dependencies object
which is destroyed at the end of the Initialize call.

Bug: webrtc:11467
Change-Id: I2ccb22b6313fc6b2887bb581704f73a703092af3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172043
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Jorge Moreira Broche <jemoreira@google.com>
Cr-Commit-Position: refs/heads/master@{#30953}
2020-03-31 22:11:37 +00:00
d9ebe01540 Improve rollback for rtp data channel
Bug: chromium:1057333
Change-Id: I4df21bc183a8df398033ebf29a8407bacf873fac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170621
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@{#30824}
2020-03-18 21:03:20 +00:00
0357b3e7b6 RtpTransceiverInterface: add header_extensions_to_offer()
This change adds exposure of a new transceiver method for getting
the total set of supported extensions stored as an attribute,
and their direction. If the direction is kStopped, the extension
is not signalled in Unified Plan SDP negotiation.

Note: SDP negotiation is not modified by this change.

Changes:
- RtpHeaderExtensionCapability gets a new RtpTransceiverDirection,
  indicating either kStopped (extension available but not signalled),
  or other (extension signalled).
- RtpTransceiver gets the new method as described above. The
  default value of the attribute comes from the voice and video
  engines as before.

https://chromestatus.com/feature/5680189201711104.
go/rtp-header-extension-ip
Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk

Bug: chromium:1051821
Change-Id: I440443b474db5b1cfe8c6b25b6c10a3ff9c21a8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170235
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30800}
2020-03-16 13:16:42 +00:00
61f74d91f8 Reland "Expose can_trickle_ice_candidates on PeerConnection"
This reverts commit cb8c40138ca170f841bc45fa6771cdfc4b966e5f.

Reason for revert: Added missing default.

Original change's description:
> Revert "Expose can_trickle_ice_candidates on PeerConnection"
>
> This reverts commit c6a65c8866487c6adc0a7bb472d3bad9389501f9.
>
> Reason for revert: Breaks downstream due to missing default
>
> Original change's description:
> > Expose can_trickle_ice_candidates on PeerConnection
> >
> > Bug: chromium:708484
> > Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Taylor <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30653}
>
> TBR=deadbeef@webrtc.org,hta@webrtc.org
>
> Change-Id: Iaa5b977c4237715a8a5127cf167cf6512a3f7059
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:708484
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169540
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30655}

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

Change-Id: I608da7781f158b4b02dd226d4dcd5615c4935fa8
Bug: chromium:708484
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169541
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30656}
2020-03-02 11:35:53 +00:00
cb8c40138c Revert "Expose can_trickle_ice_candidates on PeerConnection"
This reverts commit c6a65c8866487c6adc0a7bb472d3bad9389501f9.

Reason for revert: Breaks downstream due to missing default

Original change's description:
> Expose can_trickle_ice_candidates on PeerConnection
> 
> Bug: chromium:708484
> Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30653}

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

Change-Id: Iaa5b977c4237715a8a5127cf167cf6512a3f7059
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:708484
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169540
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30655}
2020-03-02 10:14:14 +00:00
c6a65c8866 Expose can_trickle_ice_candidates on PeerConnection
Bug: chromium:708484
Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30653}
2020-03-02 05:19:16 +00:00
d85ea75cbd Rollback transport created by data channel
No-Try: True
Bug: chromium:1032987
Change-Id: I2c0dbd6a19e71a391dc2e0d30676d4efa26a9525
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168306
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30561}
2020-02-20 01:24:55 +00:00
7a829a8563 Sort threading for sctp_mid_ variable
Split the sctp_mid_ variable into two variables,
sctp_mid_n_ and sctp_mid_s_, each of which is only accessed
by one thread.

Bug: webrtc:9987
Change-Id: I4dce944b920f4698e2606a7b85776791cbf55c28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168243
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30503}
2020-02-12 08:34:12 +00:00
c8ff1600d3 Don't crash when renegotiating after the peer rejects data channels
Bug: webrtc:11320
Change-Id: I5a58d550574a4e0702fc6f05b7fb663fbc23d0b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168200
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30463}
2020-02-05 23:33:29 +00:00
ec47b57f14 Do not transition ICE gathering state to 'complete' when closing
Bug: webrtc:4728
Change-Id: I6bcb3dd0eb47dc945d96555f9481146f22ceb4fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167440
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30433}
2020-01-30 23:17:59 +00:00
f417238217 Remove iceRegatherIntervalRange
This was an ICE configuration experiment added a couple years ago that did not end up being used.

Bug: webrtc:11316
Change-Id: Iafb7e1c4f7b4598815f045808dbf6e470172f119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167680
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30395}
2020-01-28 19:16:18 +00:00
64f1f3f04e Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED
Bug: None
Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 15:20:35 +00:00
b2b2031457 Concatenate string literals at compile time.
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format

After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.

This primary benefit of this change is a small reduction in binary size.

Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
2020-01-14 14:47:48 +00:00
977b265702 Reduce some logging at INFO level by moving log statements
from LS_INFO to LS_VERBOSE.

By default, unit tests run with logging at info level.
A random run today produced more than 70.000 lines of
output. This CL would reduce that by approximately 15.000.

Bug: none
Change-Id: Ie62708cebf109510a2443aa5ab5c4e645ffc6707
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161950
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30077}
2019-12-12 21:54:06 +00:00
0095d37137 Replace hostCandidate with address and port in RTCPeerConnectionIceErrorEvent
Bug: chromium:1013564
Change-Id: Ie1bb86ed6a2a7d73fe6ee666f973d809ed05a7ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161084
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#30004}
2019-12-04 13:18:22 +00:00
246724b0fe Move messaging -> PostTask for freeing datachannels
I could find no reason for the extra complexity of doing messaging
in order to schedule a task to be done after the current cycle.
It also simplifies the peerconnection/datachannelcontroller coupling.

Bug: webrtc:11146
Change-Id: I68f45059b9f4a6869fb44b856e05a480f4652365
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161232
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29997}
2019-12-03 22:57:17 +00:00
05e4d08e35 Refactoring DataChannelController from PeerConnection part 4
This CL:
- Moved HasDataChannel and data_channel_type_
- Moved rtp_data_channels_
- Moved sctp_data_channels_
- Moved data_channel_controller to its own .h file
- Various changes to reduce the coupling between the classes
- Removed friendship between DataChannelController and PeerConnection

Bug: webrtc:11146
Change-Id: Ib8c395e4c90ce34baf40812d1dade0ffa79f2438
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161094
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29987}
2019-12-03 15:35:09 +00:00
00cf34c5e8 Refactor DataChannel control out of PeerConnection
This is step 1-3 of the refactoring process outlined in comment #1 of bugs.webrtc.org/11146

Bug: webrtc:11146
Change-Id: Iccad009bc0585f99d207a6ddb42fd8e71312fc0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161003
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29970}
2019-12-02 10:00:34 +00:00
7a9a092708 Delete media transport integration.
MediaTransport is deprecated and the code is unused.

No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
2019-11-26 19:19:36 +00:00
353a718dfd Address failing wpt test cases for the rollback feature
Also fix https://crbug.com/1025542.

Bug: chromium:1025557, chromium:1025542
Change-Id: I614ca6282f1f1d4d1e2cd507c0efd6bc6a898408
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159932
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29909}
2019-11-25 21:54:30 +00:00
408cb4bf30 Make SCTPtransport enter "closed" state when DTLStransport does.
Bug: webrtc:11090
Change-Id: I30e0b70387746d6c544ed1818f276569d4258cf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159888
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29810}
2019-11-16 14:56:01 +00:00
25ec8882f7 Make ICE transports injectable.
Bug: chromium:1024965
Change-Id: I4961f50aee34c82701299f59a95cb90d231db6f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158820
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29807}
2019-11-15 21:31:19 +00:00
01294f0e29 Don't configure video codec switching if no video stream has been created.
Bug: none
Change-Id: I8e74fefed1e902c35064700f826b8f565e18c704
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159800
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29798}
2019-11-14 13:12:50 +00:00
ee6f4f67ef [PeerConnection] Implement asynchronous version of AddIceCandidate().
This is the same as the existing version, except it uses the Operations
Chain. As such, if an asynchronous operation that uses the chain is
currently pending, such as CreateOffer() or CreateAnswer(),
AddIceCandidate() will not happen until the previous operation
completes.

Bug: chromium:1019222
Change-Id: Ie6e5fc386fa9c29b5e2f8e3f65bfbaf9837d351c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158741
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29704}
2019-11-06 12:16:00 +00:00
4e19670d3a [PeerConnection] Implement parameterless SetLocalDescription().
For background, motivation, requirements and implementation notes, see
https://docs.google.com/document/d/1XLwNN2kUIGGTwz9LQ0NwJNkcybi9oKnynUEZB1jGA14/edit?usp=sharing

The parameterless SetLocalDescription() will implicitly create an
offer or answer to be set by chaining create offer or answer with
setting the session description, as per spec:
https://w3c.github.io/webrtc-pc/#dom-peerconnection-setlocaldescription

Bug: chromium:980885
Change-Id: Ia430160869df18fd47b756b9adf9e7e23ba8e969
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157444
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29653}
2019-10-30 10:24:44 +00:00