This allows to delete non-needed headers from .h files.
Ran this on peer_connection.cc to demonstrate.
Bug: none
Change-Id: I69ce48be25a9778297e521282898c6ac31e7ea6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254600
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36170}
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}
Also apply IWYU to all .cc files in pc/, and correct BUILD file to match.
Note: Some files came out wrong when iwyu was applied. These are not included.
Bug: none
Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36064}
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}
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}
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}
As suggested in another review.
Also add one more guard, and some commentary.
Bug: None
Change-Id: I9b84453ff2533fe01d157fe84f07405d352e1dc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235820
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35245}
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}
Deprecate CreateDataChannel, and make it a simple wrapper function.
Bug: webrtc:12796
Change-Id: I053d75a264596ba87ca734a29df9241de93a80c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219784
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34130}
In this CL, JsepTransportController and MediaSessionDescriptionFactory
are updated not to assume that there only exists at most a single BUNDLE
group but a list of N groups. This makes it possible to create multiple
BUNDLE groups by having multiple "a=group:BUNDLE" lines in the SDP.
This makes it possible to have some m= sections in one group and some
other m= sections in another group. For example, you could group all
audio m= sections in one group and all video m= sections in another
group. This enables "send all audio tracks on one transport and all
video tracks on another transport" in Unified Plan. This is something
that was possible in Plan B because all ssrcs in the same m= section
were implicitly bundled together forming a group of audio m= section and
video m= section (even without use of the BUNDLE tag).
PeerConnection will never create multiple BUNDLE groups by default, but
upon setting SDP with multiple BUNDLE groups the PeerConnection will
accept them if configured to accept BUNDLE. This makes it possible to
accept an SFU's BUNDLE offer without having to SDP munge the answer.
C++ unit tests are added. This fix has also been verified manually on:
https://jsfiddle.net/henbos/to89L6ce/43/
Without fix: 0+2 get bundled, 1+3 don't get bundled.
With fix: 0+2 get bundled in first group, 1+3 get bundled in second
group.
Bug: webrtc:10208
Change-Id: Iaf451fa5459c484730c8018274166ef154b19af8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214487
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33838}
Also add a function for accessing the list as internal transceivers
rather than accessing the proxy objects; this exposes where the
internal objects are accessed and where we need external references.
Used the new list function in sdp_offer_answer wherever possible.
Adds an UnsafeList function that is not thread guarded, so that the
job of rooting out those instances can be done in a later CL.
Bug: webrtc:12692
Change-Id: Ia591f22a1c8f82ec452a1a66a94fbf9ab9debd14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215581
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33781}
Since there is only a single type of DataChannel now, the enum was only used
when data channels were disabled at the PC API. That option has been
deprecated 4 years ago, it's now time to remove it.
Bug: webrtc:6625
Change-Id: I9e4ada1756da186e9639dd0fbf0249c55ea0b6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215661
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33778}
* Adds a OnPacketSent callback to MediaChannel, which matches with
MediaChannel::NetworkInterface::SendPacket.
* Moves the OnPacketSent handling to the media channel implementations
(video/voice) and removes the PeerConnection/SdpOfferAnswerHandler
layer from the call path.
* Call::OnSentPacket is called directly from the channels on the network
thread. This eliminates a PostTask to the worker thread for every
audio/video network packet.
* Remove sigslot dependency from MediaChannel (and derived).
Bug: webrtc:11993
Change-Id: I1f79a7aa60f05d47e1882f9be1c9323ea8fac5f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215403
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33777}
With this change, all production callers of BaseChannel::transport_name()
will be making the call from the right thread and we can safely delegate
the call to the transport itself. Some tests still need to be updated.
This facilitates the main goal of not needing synchronization inside
of the channel classes, being able to apply thread checks and eventually
remove thread hops from the channel classes.
A downside of this particular change is that a blocking call to the
network thread from the signaling thread inside of RTCStatsCollector
needs to be done. This is done once though and fixes a race.
Bug: webrtc:12601, webrtc:11687, webrtc:12644
Change-Id: I85f34f3341a06da9a9efd936b1d36722b10ec487
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213080
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33775}
This reverts commit 5a40b3710545edfd8a634341df3de26f57d79281.
Reason for revert: Fixed the bug and ran layout tests.
Original change's description:
> Revert "Use the new DNS resolver API in PeerConnection"
>
> This reverts commit acf8ccb3c9f001b0ed749aca52b2d436d66f9586.
>
> Reason for revert: Speculative revert for https://ci.chromium.org/ui/p/chromium/builders/try/win10_chromium_x64_rel_ng/b8851745102358680592/overview.
>
> Original change's description:
> > Use the new DNS resolver API in PeerConnection
> >
> > Bug: webrtc:12598
> > Change-Id: I5a14058e7f28c993ed927749df7357c715ba83fb
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212961
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33561}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta@webrtc.org
>
> Bug: webrtc:12598
> Change-Id: Idc9853cb569849c49052f9cbd865614710fff979
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213188
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33591}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12598
Change-Id: Ief7867f2f23de66504877cdab1b23a11df2d5de4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33647}
This stops pending internal callbacks from performing unnecessary
operations when closed.
Also update tests pc tests to call Close().
This will allow PeerConnection to be able to expect the
normal path to be that IsClosed() be true in the dtor
once all 'normal' paths do that
Bug: webrtc:12633
Change-Id: I3882bedf200feda0d04594adeb0fdac85bfef652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213426
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33617}
Going forward, we'll need to read this value from other threads than
signaling, so I've moved the initialization into the constructor.
Bug: none
Change-Id: I56b00d38c86788cbab9a2055719074ea48f4750f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213185
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33613}
and report some metrics only on the first connection state
change to connected
BUG=webrtc:12383
Change-Id: I32908e23c51aa40730be8e534793829268d4e25e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208583
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33333}
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.
Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
SdpOfferAnswerHandler now hands over most of the work of adding a
remote candidate over to PeerConnection where the work will be
carried out asynchronously on the network thread (was
synchronous/blocking).
Once added, reporting (ReportRemoteIceCandidateAdded) continues on the
signaling thread as before. The difference is though that we don't
block the UseCandidate() operation which is a part of applying the
local and remote descriptions.
Besides now being asynchronous, there's one behavioural change:
Before starting the 'add' operation, the validity of the candidate
instance to be added, is checked. Previously if such an error occurred,
the error was silently ignored.
Bug: webrtc:9987
Change-Id: Ic1bfb8e27670fc81038b6ccec95ff36c65d12262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206063
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33230}
This reverts commit 6e4fcac31312f2dda5b60d33874ff0cd62f94321.
Reason for revert: Parent CL issue has been resolved.
Original change's description:
> Revert "Remove thread hops from events provided by JsepTransportController."
>
> This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76.
>
> Reason for revert: Parent CL breaks FYI bots.
> See https://webrtc-review.googlesource.com/c/src/+/206466
>
> Original change's description:
> > Remove thread hops from events provided by JsepTransportController.
> >
> > Events associated with Subscribe* methods in JTC had trampolines that
> > would use an async invoker to fire the events on the signaling thread.
> > This was being done for the purposes of PeerConnection but the concept
> > of a signaling thread is otherwise not applicable to JTC and use of
> > JTC from PC is inconsistent across threads (as has been flagged in
> > webrtc:9987).
> >
> > This change makes all CallbackList members only accessible from the
> > network thread and moves the signaling thread related work over to
> > PeerConnection, which makes hops there more visible as well as making
> > that class easier to refactor for thread efficiency.
> >
> > This CL removes the AsyncInvoker from JTC (webrtc:12339)
> >
> > The signaling_thread_ variable is also removed from JTC and more thread
> > checks added to catch errors.
> >
> > Bug: webrtc:12427, webrtc:11988, webrtc:12339
> > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33195}
>
> TBR=nisse@webrtc.org,tommi@webrtc.org
>
> Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12427
> Bug: webrtc:11988
> Bug: webrtc:12339
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33203}
TBR=nisse@webrtc.org,tommi@webrtc.org,guidou@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12427
Bug: webrtc:11988
Bug: webrtc:12339
Change-Id: I4e2e1490e1f9a87ed6ac4d722fd3c442e3059ae0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206809
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33225}
This reverts commit 6b143c1c0686519bc9d73223c1350cee286c8d78.
Reason for revert:
Relanding with updated expectations for SctpTransport::Information
based on TransceiverStateSurfacer in Chromium.
Original change's description:
> Revert "Fix unsynchronized access to mid_to_transport_ in JsepTransportController"
>
> This reverts commit 6cd405850467683cf10d05028ea0f644a68a91a4.
>
> Reason for revert: Breaks WebRTC Chromium FYI Bots
>
> First failure:
> https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925
>
> Failed tests:
> WebRtcDataBrowserTest.CallWithSctpDataAndMedia
> WebRtcDataBrowserTest.CallWithSctpDataOnly
>
> Original change's description:
> > Fix unsynchronized access to mid_to_transport_ in JsepTransportController
> >
> > * Added several thread checks to JTC to help with programmer errors.
> > * Avoid a few Invokes() to the network thread here and there such
> > as for fetching sctp transport name for getStats(). The transport
> > name is now cached when it changes on the network thread.
> > * JsepTransportController instances now get deleted on the network
> > thread rather than on the signaling thread + issuing an Invoke()
> > in the dtor.
> > * Moved some thread hops from JTC over to PC which is where the problem
> > exists and also (imho) makes it easier to see where hops happen in
> > the PC code.
> > * The sctp transport is now started asynchronously when we push down the
> > media description.
> > * PeerConnection proxy calls GetSctpTransport directly on the network
> > thread instead of to the signaling thread + blocking on the network
> > thread.
> > * The above changes simplified things for webrtc::SctpTransport which
> > allowed for removing locking from that class and delete some code.
> >
> > Bug: webrtc:9987, webrtc:12445
> > Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33191}
>
> TBR=tommi@webrtc.org,hta@webrtc.org
>
> Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9987
> Bug: webrtc:12445
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33204}
TBR=tommi@webrtc.org,hta@webrtc.org,guidou@webrtc.org
# Not skipping CQ checks because this is a reland.
Bug: webrtc:9987
Bug: webrtc:12445
Change-Id: Icb205cbac493ed3b881d71ea3af4fb9018701bf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206560
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33219}
This reverts commit 6cd405850467683cf10d05028ea0f644a68a91a4.
Reason for revert: Breaks WebRTC Chromium FYI Bots
First failure:
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925
Failed tests:
WebRtcDataBrowserTest.CallWithSctpDataAndMedia
WebRtcDataBrowserTest.CallWithSctpDataOnly
Original change's description:
> Fix unsynchronized access to mid_to_transport_ in JsepTransportController
>
> * Added several thread checks to JTC to help with programmer errors.
> * Avoid a few Invokes() to the network thread here and there such
> as for fetching sctp transport name for getStats(). The transport
> name is now cached when it changes on the network thread.
> * JsepTransportController instances now get deleted on the network
> thread rather than on the signaling thread + issuing an Invoke()
> in the dtor.
> * Moved some thread hops from JTC over to PC which is where the problem
> exists and also (imho) makes it easier to see where hops happen in
> the PC code.
> * The sctp transport is now started asynchronously when we push down the
> media description.
> * PeerConnection proxy calls GetSctpTransport directly on the network
> thread instead of to the signaling thread + blocking on the network
> thread.
> * The above changes simplified things for webrtc::SctpTransport which
> allowed for removing locking from that class and delete some code.
>
> Bug: webrtc:9987, webrtc:12445
> Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33191}
TBR=tommi@webrtc.org,hta@webrtc.org
Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9987
Bug: webrtc:12445
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33204}
This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76.
Reason for revert: Parent CL breaks FYI bots.
See https://webrtc-review.googlesource.com/c/src/+/206466
Original change's description:
> Remove thread hops from events provided by JsepTransportController.
>
> Events associated with Subscribe* methods in JTC had trampolines that
> would use an async invoker to fire the events on the signaling thread.
> This was being done for the purposes of PeerConnection but the concept
> of a signaling thread is otherwise not applicable to JTC and use of
> JTC from PC is inconsistent across threads (as has been flagged in
> webrtc:9987).
>
> This change makes all CallbackList members only accessible from the
> network thread and moves the signaling thread related work over to
> PeerConnection, which makes hops there more visible as well as making
> that class easier to refactor for thread efficiency.
>
> This CL removes the AsyncInvoker from JTC (webrtc:12339)
>
> The signaling_thread_ variable is also removed from JTC and more thread
> checks added to catch errors.
>
> Bug: webrtc:12427, webrtc:11988, webrtc:12339
> Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33195}
TBR=nisse@webrtc.org,tommi@webrtc.org
Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12427
Bug: webrtc:11988
Bug: webrtc:12339
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33203}
Events associated with Subscribe* methods in JTC had trampolines that
would use an async invoker to fire the events on the signaling thread.
This was being done for the purposes of PeerConnection but the concept
of a signaling thread is otherwise not applicable to JTC and use of
JTC from PC is inconsistent across threads (as has been flagged in
webrtc:9987).
This change makes all CallbackList members only accessible from the
network thread and moves the signaling thread related work over to
PeerConnection, which makes hops there more visible as well as making
that class easier to refactor for thread efficiency.
This CL removes the AsyncInvoker from JTC (webrtc:12339)
The signaling_thread_ variable is also removed from JTC and more thread
checks added to catch errors.
Bug: webrtc:12427, webrtc:11988, webrtc:12339
Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33195}
* Added several thread checks to JTC to help with programmer errors.
* Avoid a few Invokes() to the network thread here and there such
as for fetching sctp transport name for getStats(). The transport
name is now cached when it changes on the network thread.
* JsepTransportController instances now get deleted on the network
thread rather than on the signaling thread + issuing an Invoke()
in the dtor.
* Moved some thread hops from JTC over to PC which is where the problem
exists and also (imho) makes it easier to see where hops happen in
the PC code.
* The sctp transport is now started asynchronously when we push down the
media description.
* PeerConnection proxy calls GetSctpTransport directly on the network
thread instead of to the signaling thread + blocking on the network
thread.
* The above changes simplified things for webrtc::SctpTransport which
allowed for removing locking from that class and delete some code.
Bug: webrtc:9987, webrtc:12445
Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33191}
adds metrics for bundle usage, differentiating between
* BUNDLE is not negotiated and there is only a datachannel,
* BUNDLE is not negotiated and there is at most one m-line per media type,
for unified-plan
* BUNDLE is not negotiated and there are multiple m-lines per media type,
* BUNDLE is negotiated and there is only a datachannel,
* BUNDLE is negotiated but there is at most one m-line per media type,
* BUNDLE is negotiated and there are multiple m-lines per media type,
and for plan-b
* BUNDLE is negotiated
* BUNDLE is not negotiated
BUG=webrtc:12383
Change-Id: I41afc4b08fd97239f3b16a8638d9753c69b46d22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202254
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33090}
- Remove the last sigslot from JsepTransportController.
- Tested the potential test failure on chromium blink test by importing
this CL to chromium source.
Bug: webrtc:11943
Change-Id: I107d05606350aff655ca73a5cb640dff1a7036ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202741
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33085}
it no longer reports just offers.
BUG=chromium:857004
Change-Id: Idf35b6fa98f3ee6637aeef6b11553947fea3ee25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202249
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33024}
Needed in order to return different codes for different failures
in initialization.
Sideswipe: Check TURN URL hostnames for illegal characters.
Bug: webrtc:12238
Change-Id: I1af3a37b9654b83b268304f7356049f9f3786b7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195541
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32710}
and pass it as an argument to PeerConnection::Create
This makes it obvious that 1) options only affect peerconnections
if they are set on the factory before creating the PeerConnection,
and 2) options are unchangeable after PeerConnection creation.
Bug: webrtc:11967
Change-Id: I052eaa3975ac97dccbedde610110f32bf1a17c98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191487
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32549}
After recent refactorings, PeerConnection.tls_cert_verifier_ is
now both const and only accessed on the network thread, so it is
doubly thread-safe. Marking as such.
Bug: webrtc:9987
Change-Id: I2f924ecf2afe364d1e4b7f740435443bc53e4d0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191486
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32545}
And add a Create() method to the class.
This makes it possible to experiment with subclassing the
SdpOfferAnswer object without modifying the PeerConnection.
Bug: webrtc:11995
Change-Id: I0a7c91a8999858ddcb1ea59ac4eb9a3b0663b0f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190288
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32501}
