Repeatedly open and close data channels on a peer connection
to check that the channels are properly negotiated and SCTP
stream IDs properly recycled.
Bug: webrtc:13994, chromium:1320194
Change-Id: I244911abb5abaf0a290de07a0d790cd1edffe8cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260984
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36780}
If media_engine is not passed in init parameters, the PC can't handle
media, but can be used for datachannels. This CL adds testing that
datachannels work without media engine, and adds failure returns
to PeerConnection APIs that manipulate media when media engine is
not present.
Bug: webrtc:13931
Change-Id: Iecdf17a0a0bb89e0ad39eb74d6ed077303b875c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261246
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36778}
This ensures that only the compilation units that actually need
ChannelManager details can see it.
Bug: webrtc:13931
Change-Id: Iddd37580c0ceceba5b7095e84b981e6a525b2800
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261200
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36762}
This also hides the existence of the classes VideoChannel and
VoiceChannel from anything that does not include "channel.h".
Bug: webrtc:13931
Change-Id: I080a692b6acfd5d2d0401ec20d59c3a684eddb05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260944
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36746}
This makes the channel manager object into a factory, not a manager.
Bug: webrtc:13931
Change-Id: I59f7d818a739797a7c0a7a32e6583450834df122
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260467
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36718}
This makes it clearer which modules set the channel.
Also remove GetChannel() from PeerConnection public API
This was only used once, internally, and can better be inlined.
Part of reducing the exposure of Channel.
Bug: webrtc:13931
Change-Id: I5f44865230a0d8314d269c85afb91d4b503e8de0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260187
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36695}
This is an implementation API, user classes should in principle
only use the channel_interface.h
Bug: webrtc:13931
Change-Id: I85c285217858dc087c90a50792e980f731f4439f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260185
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36674}
Prior to this CL, calling RtpTransceiver::SetChannel() with null
arguments would cause the receiver's track to end. This is wrong,
because the channel can be nulled for other reasons than the transceiver
being stopped/removed - such as when the transceiver is rolled back but
still in use. Also, stopping a transceiver will end the track, so we
should simply ensure to always stop the transceiver when that is needed.
This CL makes sure that the transceiver is stopped or stopping in all
appropriate places, allowing us to remove the ability to end the source
for any other reason. A side-effect of this is that:
- The track never ends prematurely, fixing https://crbug.com/1315611.
- Removed transceivers are always stopped, fixing
https://crbug.com/webrtc/14005.
This CL fixes the issue of track being ended in the ontrack event when
running https://jsfiddle.net/henbos/nxebusjm/.
- We don't have WPT test coverage for this, so I'll add that separately.
With SetSourceEnded() removed, some stopping/stop in response to
rejecting locally SDP munged content had to be added in order not to
regress the existing test coverage for this:
*PeerConnectionInterfaceTest.RejectMediaContent/1
Bug: chromium:1315611, webrtc:14005.
Change-Id: I21f30a1259e51324066dc84f72a72485b9e0fadc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36669}
This better reflects the ownership passing of AddTrack, and is more
consistent for RemoveTrack.
Bug: webrtc:13980
Change-Id: Ide5baccf15fc687a4e092f8831ce8c0fea46604e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259740
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36603}
This cl/ adds the feature actually injecting a FieldTrialsView into
PeerConnectionFactory, or into a PeerConnection or both.
The field trials used for a PeerConnection is those specified in
PeerConnectionDependencies. Otherwise will those from
PeerConnectionFactoryDependencies be used (and until we're finished with
this conversion, the global string fallback is used as last resort).
Note that it is currently not possible to create 2 FieldTrials
objects concurrently...due to global string,
so this cl/ is mostly (but entirely) for show, i.e one _can_
realistically inject them into a PeerConnectionFactory.
Bug: webrtc:10335
Change-Id: Id2e60525f48a1f8293c1dd0be771e3ed03790963
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258134
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36578}
Anything linking to //third_party/jsoncpp is hiding deprecated usage
warnings, so these were not discovered earlier.
Bug: chromium:983223
Change-Id: Id0ade4ca016f19db16377dbeeb756358a7e94fa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258124
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36463}
Also update API proxy Create() factory functions to accept the inner
reference counted object via scoped_refptr instead of a raw pointer.
This is to avoid accidentally creating and deleting an object when
passing an inner object to a proxy class.
Consider something like:
auto proxy = MyProxy::Create(
signaling_thread(), make_ref_counted<Foo>());
Bug: webrtc:13464, webrtc:12701
Change-Id: I55ccfff43bbc164a5e909b2c9020e306ebb09075
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256010
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36261}
This cl/
1) move WebRtcKeyValueConfig from api/transport to api/ directory.
2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials).
3) removes usage of webrtc::field_trial:: from the pc/ directory.
4) removes a few unused includes of system_wrappers/field_trial.h.
Bug: webrtc:10335
Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36160}
This makes SetChannel() consistently make 2 invokes instead of a
multiple of senders+receivers (previous minimum was 4 but could be
larger).
* Stop() doesn't hop to the worker thread.
* SetMediaChannel(), an already-required step on the worker thread for
senders and *sometimes* for receivers[1], is now consistently required
for both. This simplifies transceiver teardown and enables the next
bullet.
* Transceiver stops all senders and receivers in one go rather than
ping ponging between threads.
[1] When not required, it was done implicitly inside of Stop().
See changes in `RtpTransceiver::SetChannel`
Bug: webrtc:13540
Change-Id: Ied61636c8ef09d782bf519524fff2a31e15219a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249797
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36057}
Also expands integration_test_helpers to deal with multiple
datachannels.
The bug has not yet been triggered.
Bug: webrtc:13668
Change-Id: I82a0fdae0cc32815c250a691b56c614bfd6d606b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251602
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35980}
A pointer to the transport controller is now maintained on
both the network thread and the signaling thread. We use
thread specific accessors to make it explicit which copy we
are accessing at any given time.
We also move the initial offerer value to the SDP offer/answer
class; this is determined on the basis of SDP offer/answer, so
there is no need to hop to the network thread for that.
Work in progress.
Bug: webrtc:9987
Change-Id: Idbe5a7fbf44f667adcd119e486133cf6e43ab1f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251382
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35965}
This reverts commit 8efc914cf353cea138a453c45e970e589bec0834.
Reason for revert: Breaks downstream project.
Original change's description:
> Replace use of sigslot with CallbackList in data_channel_controller
>
> This is a straightforward replacement; simplifications due to the ability
> to inline functions in the lambdas are for a later CL.
>
> Bug: webrtc:11943
> Change-Id: I7274cedde507b954f1d8aa8bc560861102eeb264
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250540
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35936}
TBR=nisse@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I8fd0f32ceec866bfd9a08cac1108b559bf03caac
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251280
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35941}
This is a straightforward replacement; simplifications due to the ability
to inline functions in the lambdas are for a later CL.
Bug: webrtc:11943
Change-Id: I7274cedde507b954f1d8aa8bc560861102eeb264
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35936}
This removes a couple of methods from the PeerConnectionSdpMethods
interface.
Bug: webrtc:11995
Change-Id: I0a68178b1f0a99e779e6d7f94d03b493d811f500
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249794
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35841}
This CL changes the SdpOfferAnswerHandler class to depend on a new class
PerConnectionInternalMethods, which is implemented by PeerConnection.
This means that SdpOfferAnswerHandler no longer depends on
PeerConnectionInterface.
This opens the way for refactoring PeerConnection so that
PeerConnectionInternalMethods is a member object (encapsulation not
inheritance), which will make it possible to break some of the
dependency cycles that make the "peerconnection" target in the BUILD
file so huge.
Bug: webrtc:11995
Change-Id: Ib8413a31c0148b8d8602764b7367dfd3068da58a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249785
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35828}
Just applied a short sed script. See bug description for
the motiviation for this change.
This is the command that was used to generate the changes:
$ find . -type f \( -iname '*.cc' -o -iname '*.h' \) -print0 | \
xargs -0 sed -i -e 's/(const override)/(const, override)/'
Bug: webrtc:13090
Change-Id: Iec7d280f9d55263a972dbb3bd644ebfcd2eb38cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249088
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35801}
Move deletion of channel objects over to the RtpTransceiver instead
of having it done by SdpOfferAnswer.
The deletion is now also done via PostTask rather than Invoke.
Bug: webrtc:11992, webrtc:13540
Change-Id: I5aff14956d5e572ca8816bbfef8739bb609b4484
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248170
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35798}
The interface is implemented by the ChannelManager and contains methods
to create and destroy media channel objects as used by a transceiver.
This will subsequently allow us to delete the channel objects from
the transceiver class where ownership really lies rather than from
the outside - which is currently required by some tests that keep
channel objects on the stack. We'll furthermore be able to do the
destruction asynchronously without additional Invoke()s as we do now
which will remove an Invoke when making sdp changes.
With introducing the interface, the following simplifications were made:
* ChannelManager constructed on the signaling thread.
Before, there was an Invoke in the context class, which existed
for the purposes of calling MediaEngine::Init() (which in turn is
only needed for the VoiceEngine). This Invoke has now been moved
into the CM (more tbd).
* The CM now has a pointer to the signaling thread (since that's the
construction thread). That allows us to remove the signaling thread
parameter from the CreateFooChannel methods.
* The ssrc_generator (UniqueRandomIdGenerator) instance for SSRCs moved
from SdpOfferAnswerHandler to the CM, as it's always used in
combination with the CM. This simplifies the CreateFooChannel methods
as well as a couple of other classes that have a CM dependency.
* Removed DestroyFooChannel related code from SdpOfferAnswerHandler since
the channel type detail can be taken care of by the CM.
Bug: webrtc:11992, webrtc:13540
Change-Id: I04938a803734de8489ba31e6212d9eaecc244126
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247904
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35766}
...and one when destroying a channel object.
This CL removes Init_n() and Deinit_n() from the BaseChannel class.
Channel classes now use SetRtpTransport to do initialization and
uninitialization on the network thread.
Notably if an implementation has called SetRtpTransport() with a valid
transport pointer, it is required that SetRtpTransport be called again
with a nullptr before the channel object can be deleted.
In situations where multiple channels are created, this can mean
a substantial reduction in thread hops. We still hop to the worker
in order to construct the objects - this can probably be avoided
and SetChannel() is still a synchronous operation for the transceivers.
Furthermore, teardown of channel objects also still happens
synchronously and across network/worker/signaling threads.
Bug: webrtc:11992
Change-Id: I68ca7596e181fc82996e3e290733d97381aa5e78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246740
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35738}
In preparation for switching the default from kPlanB to kUnifiedPlan,
which could cause subtle bugs for those not prepared for it, we change
the default to kNotSpecified. The only purpose of kNotSpecified is to
crash, forcing any dependencies to explicitly set their sdp_semantics
value.
Tests are updated to explicitly set sdp_semantics when necessary, and
where the test does not care we update to kUnifiedPlan.
If this change lands without getting reverted we can let it sit for a
few weeks, after which we should change the default to kUnifiedPlan and
delete kNotSpecified.
Bug: webrtc:11121
Change-Id: I19b669b0735d78e269e19eaae86c2d7d95a91141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242968
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35651}
This is a slight refactoring while doing some other changes, so not
strictly necessary, but the error param is always supplied in practice
so it made sense to update the tests to reflect that, test that error
values are reported in (at least) some cases and remove the additional
code that checks for whether or not error information is requested.
Bug: none
Change-Id: Ia5739a18ea2beb6970eabf9d809c24dfa43466b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244097
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35632}
This also removes all internal usage of RemoveTrack, and changes
the replacement function to RemoveTrackOrError rather than RemoveTrackNew.
Bug: webrtc:9534
Change-Id: Idf7bb17495686de77c70428dcbfb12278328ce59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244094
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35624}
Because of this (seemingly simple) change, I had to change the return
type of transport_name from `const std::string&` to `absl::string_view`
to handle the case when there's no transport assigned.
That in turn caused an avalanche of required updates.
Bug: webrtc:12230, webrtc:11993
Change-Id: I16ec6c6a5fc2f5f7c7de572355a3c6ca924bb9d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244084
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35617}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
Also switch to "LT(meaured, limit)" format as this is easier to read.
There have been bot runs that exceeded the old limit.
Bug: webrtc:13354
Change-Id: I1c19c98e1c1777177e7dc14ae4679765c1c40550
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237342
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35317}
This makes it possible to invoke methods on the transceiver object
from any thread.
Also makes a few of the mock observer objects thread-safe, to allow
testing when the main thread is not the signaling thread.
Bug: webrtc:13183
Change-Id: Ic97efef71a21c3075700a028103061032f8d2bcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35010}
This is a reland of commit 704a834f685eb96c9fcf891ca345557bef4d138a,
after it was reverted in order to merge a CL to M93.
Original change's description:
> Fix bug where we assume new m= sections will always be bundled.
>
> A recent change [1] assumes that all new m= sections will share the
> first BUNDLE group (if one already exists), which avoids generating
> ICE candidates that are ultimately unnecessary. This is fine for JSEP
> endpoints, but it breaks the following scenarios for non-JSEP endpoints:
>
> * Remote offer adding a new m= section that's not part of any BUNDLE
> group.
> * Remote offer adding an m= section to the second BUNDLE group.
>
> The latter is specifically problematic for any application that wants
> to bundle all audio streams in one group and all video streams in
> another group when using Unified Plan SDP, to replicate the behavior of
> using Plan B without bundling. It may try to add a video stream only
> for WebRTC to bundle it with audio.
>
> This is fixed by doing some minor re-factoring, having BundleManager
> update the bundle groups at offer time.
>
> Also:
> * Added some additional validation for multiple bundle groups in a
> subsequent offer, since that now becomes relevant.
> * Improved rollback support, because now rolling back an offer may need
> to not only remove mid->transport mappings but alter them.
>
> [1]: https://webrtc-review.googlesource.com/c/src/+/221601
>
> Bug: webrtc:12906, webrtc:12999
> Change-Id: I4c6e7020c0be33a782d3608dee88e4e2fceb1be1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225642
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34544}
Bug: webrtc:12906, webrtc:12999
Change-Id: Id6acab2e2d7430c65f4b6a1d7372388a70cc18ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228465
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34728}
This relands commit I41cae74605fecf454900a958776b95607ccf3745, after
reverting it in order to merge the revert to M93 (the deadline for
which is now exceeded).
Original change description:
> If a bundle is established, then per JSEP, the offerer is required to
> include the new track in the bundle, and per BUNDLE, the answerer has
> to either accept the track as part of the bundle or reject the track;
> it cannot move it out of the group. So we will never need the transport.
>
> Bug: webrtc:12837
> Change-Id: I41cae74605fecf454900a958776b95607ccf3745
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221601
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34290}
Bug: webrtc:12837
Change-Id: I30a8f03165ab797ed766b51c4eb15c2a9cecb5ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228500
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34727}
This reverts commit I41cae74605fecf454900a958776b95607ccf3745
Reason for revert: Needed in order to cherry pick this revert into M93,
in order to fix crbug.com/1236202.
Original change description:
> If a bundle is established, then per JSEP, the offerer is required to
> include the new track in the bundle, and per BUNDLE, the answerer has
> to either accept the track as part of the bundle or reject the track;
> it cannot move it out of the group. So we will never need the transport.
>
> Bug: webrtc:12837
> Change-Id: I41cae74605fecf454900a958776b95607ccf3745
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221601
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34290}
TBR=hta@webrtc.org
Bug: webrtc:12837, chromium:1236202
Change-Id: Ie59e2ad5168e6829eefa67b1031b8f400ed66507
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227822
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34669}
This reverts commit 704a834f685eb96c9fcf891ca345557bef4d138a.
Reason for revert: Reverting this in order to revert
https://webrtc-review.googlesource.com/c/src/+/221601, so we can
merge that revert to M93.
Original change's description:
> Reland "Fix bug where we assume new m= sections will always be bundled."
>
> This is a reland of d2b885fd91909f1b17fb11292a8c989d5d883b22, after
> making sure transports that are just being kept alive in case of
> rollback don't contribute to connection state, which broke a WPT.
>
> Original change's description:
> > Fix bug where we assume new m= sections will always be bundled.
> >
> > A recent change [1] assumes that all new m= sections will share the
> > first BUNDLE group (if one already exists), which avoids generating
> > ICE candidates that are ultimately unnecessary. This is fine for JSEP
> > endpoints, but it breaks the following scenarios for non-JSEP endpoints:
> >
> > * Remote offer adding a new m= section that's not part of any BUNDLE
> > group.
> > * Remote offer adding an m= section to the second BUNDLE group.
> >
> > The latter is specifically problematic for any application that wants
> > to bundle all audio streams in one group and all video streams in
> > another group when using Unified Plan SDP, to replicate the behavior of
> > using Plan B without bundling. It may try to add a video stream only
> > for WebRTC to bundle it with audio.
> >
> > This is fixed by doing some minor re-factoring, having BundleManager
> > update the bundle groups at offer time.
> >
> > Also:
> > * Added some additional validation for multiple bundle groups in a
> > subsequent offer, since that now becomes relevant.
> > * Improved rollback support, because now rolling back an offer may need
> > to not only remove mid->transport mappings but alter them.
> >
> > [1]: https://webrtc-review.googlesource.com/c/src/+/221601
> >
> > Bug: webrtc:12906, webrtc:12999
> > Change-Id: I4c6e7020c0be33a782d3608dee88e4e2fceb1be1
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225642
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34544}
>
> Bug: webrtc:12906, webrtc:12999
> Change-Id: I68bf988b1918dd2d51de76e53e4fd696fea5a09b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227120
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34596}
TBR=hta@webrtc.org
Bug: webrtc:12906, webrtc:12999
Change-Id: I129d9eb3b9831317fa24b0263db191027246cb99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227821
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34666}
This is a reland of d2b885fd91909f1b17fb11292a8c989d5d883b22, after
making sure transports that are just being kept alive in case of
rollback don't contribute to connection state, which broke a WPT.
Original change's description:
> Fix bug where we assume new m= sections will always be bundled.
>
> A recent change [1] assumes that all new m= sections will share the
> first BUNDLE group (if one already exists), which avoids generating
> ICE candidates that are ultimately unnecessary. This is fine for JSEP
> endpoints, but it breaks the following scenarios for non-JSEP endpoints:
>
> * Remote offer adding a new m= section that's not part of any BUNDLE
> group.
> * Remote offer adding an m= section to the second BUNDLE group.
>
> The latter is specifically problematic for any application that wants
> to bundle all audio streams in one group and all video streams in
> another group when using Unified Plan SDP, to replicate the behavior of
> using Plan B without bundling. It may try to add a video stream only
> for WebRTC to bundle it with audio.
>
> This is fixed by doing some minor re-factoring, having BundleManager
> update the bundle groups at offer time.
>
> Also:
> * Added some additional validation for multiple bundle groups in a
> subsequent offer, since that now becomes relevant.
> * Improved rollback support, because now rolling back an offer may need
> to not only remove mid->transport mappings but alter them.
>
> [1]: https://webrtc-review.googlesource.com/c/src/+/221601
>
> Bug: webrtc:12906, webrtc:12999
> Change-Id: I4c6e7020c0be33a782d3608dee88e4e2fceb1be1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225642
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34544}
Bug: webrtc:12906, webrtc:12999
Change-Id: I68bf988b1918dd2d51de76e53e4fd696fea5a09b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227120
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34596}
This reverts commit 37ee0f5e594dd772ec6d620b5e5ea8a751b684f0.
Reason for revert: Revert in order to be able to revert https://webrtc-review.googlesource.com/c/src/+/225642
Original change's description:
> Use backticks not vertical bars to denote variables in comments for /pc
>
> Bug: webrtc:12338
> Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34575}
TBR=hta@webrtc.org,titovartem@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I5eddd3a14e1f664bf831e5c294fbc4de5f6a88af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227082
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34577}
