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}
This patch adds send_ping_on_selected_ice_controlling that is
the same like send_ping_on_switch_ice_controlling except that it
also sends the ping on initial selection (i.e not a switch(?)).
I.e it sends "extra" pings when selecting a connection.
This lets the peer know that this candidate pair has been selected,
which causes some peers to unblock sending on the candidate pair.
TODO deprecate send_ping_on_switch_ice_controlling in favor of this,
i.e I should have done it like this on first attempt :(
Bug: webrtc:10273
Change-Id: I0e02cd5fd6d677720c1fa863af5d7c2334d02d4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182780
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32006}
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 is a reland of ba8abbb630cdd9d05e22c830d0845e920762850d
This can be relanded as the queuing issues that were causing a
crash in the WebRTC roll in Chromium have been resolved. I have
added the Chromium failing targets to the CQ for this commit and
they have succeeded.
Original change's description:
> [Adaptation] Remove QualityScalerResource when disabled.
>
> Bug: webrtc:11843
> Change-Id: I2d3e40356c266f189db0242f3c7590e6d83e4456
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181369
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31924}
Bug: webrtc:11843
Change-Id: I228331293060ef996f1dd7f8e18d52b0818f526b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182080
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31996}
This test was added in
https://webrtc-review.googlesource.com/c/src/+/180620. On my machine it
passes about 98% of the time.
The test is meant to count the number of times the callback that the
operations chain is empty has been invoked, and does this by ensuring
the last operation to make the chain empty has completed before
expecting that the counter has increased.
The race happns when the operation has completed but the callback that
the chain is empty has not happened yet. This CL fixes that by using
EXPECT_EQ_WAIT instead.
TBR=hta@webrtc.org
Bug: chromium:1060083
Change-Id: I2ebfac3e635ef895d6602f7360e5ec6006fc1d0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182541
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31992}
Resource adaptation needs refactoring for async adaptations. For now
the resource adaptation processor can work on the encoder thread, until
it is refactored to support async adaptation.
Bug: webrtc:11867
Change-Id: I9c46da356db19c0fd52748c999ccb216f2ca923b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182040
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31991}
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}
Notably, this should detect whether an interface is "available" or not,
which should prevent the failure is with dual SIM card setups.
This is gated behind a field trial for now, to ensure this doesn't cause
any regressions due to false negatives (interfaces that are usable
but not listed as available by NWPathMonitor).
Bug: webrtc:10966
Change-Id: Ia3942c4c57b525d08d8b340e2325f3705cfd0304
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180923
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31977}
Potential deadlock fixed by acquiring lock before calling encoder.
This is a reland of a135557b3c7ffa4fb1710d2d907c3cb86c5d5913
Original change's description:
> Call OnReceivedOverhead after audio network adaptor is created.
>
> This prevents ending up in a state where audio network adaptor never
> receives the current packet overhead and therefore doesn't work.
>
> Bug: chromium:1086942
> Change-Id: I8ee2ffbb7741b342b3ec93fc89f2859a146f4ba7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181583
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31951}
Bug: chromium:1086942
Change-Id: I514e523c6607cee0099b87919f0f77ebec966ddd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181888
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31971}
This change fixes a few cases of mutex recursions resulting in CHECK failures.
Bug: webrtc:11864
Change-Id: I392152e0aed88e100089a09e85504dd0abef62a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182083
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31970}
This patch adds a computed estimate on how long the ice stack
was disconnected before switching to a new connection.
The metric is currently computed as now - max(connection->last_data_recevied())
and has resonably good precision.
Bug: webrtc:11862
Change-Id: I8950d55f0eadcf164de089cdb715b4f7eed0a4c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182002
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31969}
This change fixes recursive locking going on in the Linux Pulse and ALSA audio device managers.
Bug: webrtc:11866
Change-Id: Ia7b7b82e7f1f2a92c2f99e07a7079632499354ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182020
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31968}
Some libraries hooking into WebRTC still manage to have the conference mode
flag enabled on non screenshare sources resulting in a bad rate allocation.
Bug: webrtc:11310
Change-Id: Id5205affb562511eda40c460e380c105d8589c51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182003
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31965}
