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 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 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}
Document with a comment the suspected place that could cause a bug.
Also fix an error in previous role observation code.
Bug: webrtc:13668
Change-Id: Id7f6af6905d90f7974b5570145c201c8339aaf72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251388
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35973}
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 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}
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 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}
So applications don't need to create and inject their own instance of
BasicPortAllocator, just to change these settings.
Bug: webrtc:13145
Change-Id: I08ac8658b4c0ef87019fa579be9195a8a6b50feb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239643
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35476}
This is a partial revert of commit f9e502d935fa9ca8d9cc7540680021b93d285f28.
Reason for revert: Functionality turns out to be needed by some partners for some months more.
Original change's description:
> Remove enable_dtls_srtp option
>
> This is part of the removal of support for SDES.
>
> Bug: webrtc:11066
> Change-Id: I448d0e0032672c04c87b00550ab4b9d792071a0b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234864
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35262}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11066, chromium:1271469
Change-Id: I79a90f025e53816789b391bc52a0e896b9be87e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238170
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35378}
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}
Removes the ability to accept nonencrypted answers to encrypted offers.
Fixes some logic around bundled sessions and requirement for
transport parameters.
Bug: webrtc:11066
Change-Id: I56d8628d223614918a1e5260fdb8a117c8c02dbd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236344
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35298}
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 patch adds support for manually setting subnets that
should be handled as VPN, i.e be subject to VpnPreference,
in case webrtc fails to auto-detect VPNs.
Bug: webrtc:13097
Change-Id: I42514f0677a35cfe30ad053570fa9c2a5b4a856b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230122
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34852}
This patch adds a vp preference field to RTCConfig.
DEFAULT, // No VPN preference.
ONLY_USE_VPN, // only use VPN connections.
NEVER_USE_VPN, // never use VPN connections
PREFER_VPN, // use a VPN connection if possible, i.e VPN connections sorts higher than all other connections.
AVOID_VPN, // only use VPN if there is no other connections, i.e VPN connections sorts last.
Bug: webrtc:13097
Change-Id: I3f95bdfa9134e082c7d389f803bd08facfb70262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229591
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34842}
Uppercase constants are more likely to conflict with macros (for
example rtc::SRTP_AES128_CM_SHA1_80 and OpenSSL SRTP_AES128_CM_SHA1_80).
This CL renames some constants and follows the C++ style guide.
Bug: webrtc:12997
Change-Id: I2398232568b352f88afed571a9b698040bb81c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226564
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34553}
This is part of moving the thread hops from the network thread to
worker (required by Call) into Call itself so that we can eventually
remove them.
Bug: webrtc:11993
Change-Id: Ib3ccdd6c75a3848daae2e3ce6c9a55d9617c2f50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220604
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34160}
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}
We don't support audio simulcast, so we should reject the layers
early during an addTransceiver() call.
Bug: webrtc:12719
Change-Id: Ieeb92c66de741e9b11943e0173a6f2e052926f13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216685
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33886}
This changes the notification to a single std::function pointer
instead of being a sigslot::signal1<> collection.
Summary:
* Remove SignalFirstPacketReceived_, the last sigslot member variable.
(still inherits from sigslot::has_slots<>)
* BaseChannel doesn't post to the signaling thread anymore. The only
reason that remains for the signaling_thread_ variable, is for
thread checking.
* Remove BaseChannel's reliance on MessageHandlerAutoCleanup
(still inherits from MessageHandler)
RtpTransceiver is the consumer of this event. That class is also the
class that sits between the PC classes and the channel object, holding
a pointer to the channel and managing calls that come in on the
signaling thread, such as SetChannel. The responsibility of delivering
the first packet received on the signaling thread is now with
RtpTransceiver:
* RtpTransceiver always requires a ChannelManager instance. Previously
this variable was sometimes set, but it's now required.
* Updated tests in rtp_transceiver_unittest.cc to include a
ChannelManager as well as fix them to include call expectations for
mock sender and receivers.
Bug: webrtc:11993, webrtc:11988
Change-Id: If49d6be157cd7599fa6fe3a42cd0a363464e3a74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215979
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33853}
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.
Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
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}
