BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
which may allow caching some relatively persistent statistics
such as codec statistics that only change during renegotiation.
BUG=webrtc:8693
Change-Id: Ifd68c9d666d9f328d0efecb64e4201d003788ca8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273324
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37981}
This simplifies the logic of what simulcast layers to signal, and avoids
situations where the upper layers get confused about which layers exist.
Bug: chromium:1350245
Change-Id: I9edeb93cbb30e872c4d3f3429a85a1fccf17996a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272902
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37905}
We should have done this a long time ago.
Let's do the same for stats_types.h in a separate CL because that file
is part of the api/ folder and needs some special care (typedefs and
temporarily include helper to avoid breaking downstream projects).
Bug: webrtc:14180
Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37426}
This is part of the project to delete the class entirely.
The CL also adds an "use_rtx" parameter to the function for listing
video codecs, rather than filtering those away afterwards.
Bug: webrtc:13931
Change-Id: I96b9b18c694a1c0986ccf22face76ef4c704d372
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262666
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36963}
The killswitch is no longer needed, because the googScreencastMinBitrate
has been successfully removed from the web platform.
The native RTCConfiguration::screencast_min_bitrate is still available
though because there are other downstream users than Chrome.
Bug: chromium:1315155
Change-Id: I2145f9014dbe57bb50e61f1faeacd533d76acb29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261725
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36831}
This reverts commit c48ad732d6eb69f14dd6d44f801d62997cef2c2f.
Reason for revert: breaks downstream project
Original change's description:
> Don't create channel_manager when media_engine is not set
>
> Also remove a bunch of functions in ChannelManager that were just
> forwarding to MediaEngineInterface.
>
> Bug: webrtc:13931
> Change-Id: Ia38591fd22c665cace16d032f5c1e384e413cded
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261304
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36801}
Bug: webrtc:13931
Change-Id: I1e260a2489547bd9483b50e043c28d2805b0fa5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261660
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Owners-Override: Artem Titov <titovartem@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#36811}
Also remove a bunch of functions in ChannelManager that were just
forwarding to MediaEngineInterface.
Bug: webrtc:13931
Change-Id: Ia38591fd22c665cace16d032f5c1e384e413cded
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261304
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36801}
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 breaks the link from sdp_offer_answer.cc to channel.h.
Bug: webrtc:13931
Change-Id: I75608f75713bf4e69013ac5f5b17c19e53d07519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261060
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36757}
Prior to this CL, rollback did not restore FiredDirection and remote
streams were only sometimes restored. This resulted in not firing
ontrack if a track was rolled back and then added again on the same
transceiver.
Rollback also never performed OnTrack, which is incorrect because a
transceiver that goes from sendrecv to inactive will cause OnRemoveTrack
and if this is rolled back (so we become sendrecv again) then we need
OnTrack to fire.
This CL improves rollback's "memory", fires ontrack in Rollback() and
adds test coverage.
Needed to solve similar bugs in the Chromium layers as well:
https://chromium-review.googlesource.com/c/chromium/src/+/3613313
Bug: chromium:1320669
Change-Id: I655dd7d8a6b86080fe0e7c32c9e8c6434062ae91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260330
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36734}
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 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}
See go/deprecating-media-constraints for motivation.
Setting this min bitrate is necessary for BWE to work properly when
sending screencast in low BW scenarios. The value 100 kbps appears to be
a sensible default in practise (this is the value used by Google Meet).
In order for apps not to have to rely on non-standard APIs
(googScreencastMinBitrate) for BWE to work properly, we change the
default to 100 kbps. This will unblock deprecating and removing legacy
mediaConstraints.
A kill switch is added in case this causes any unforeseen issues, but if
all goes well we can remove the kill switch in the next milestone.
Bug: chromium:1315155
Change-Id: I02b4eb0dfb26f934e678760313a0423f412512c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258681
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36523}
This was a side effect of testing out the "gn_check_autofix.py" tool
after running "apply-iwyu -r" on a few files.
Seems worth committing.
Bug: none
Change-Id: I3df446c640d4c4e3d6b15eddbdf66a1a40135f69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258024
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36446}
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 flag was used as a kill switch in case turning off payload type
demuxing in some Unified Plan cases (https://crbug.com/webrtc/12814)
would cause any issues. That landed way back in M93 and no issues were
ever reported, so time to clean up the flag.
Bug: webrtc:12814
Change-Id: I1970936131384dc0be1cd118e6b0ac877b8c289c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253240
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36109}
Also deleted iwyu script that was not maintained, and deleted
some options that made the script more complex.
Bug: none
Change-Id: I39d8eaa37f12c72ddc127ae145e6a3a80f328316
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251384
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35966}
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 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}
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}
This is a followup to [1] that moves parts of the SetRemoteDescription
operation into a subclass of SdpOfferAnswerHandler.
[1] https://webrtc-review.googlesource.com/c/src/+/244980/
Bug: webrtc:13540
Change-Id: Ic5d97f9bfd30763f3988f2f6832703ffb819a51d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245641
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35714}
Remove Init_w and Deinit(), both of which were wrappers around Invoke()
calls from the worker thread to the network thread.
Instead, replace them with Init_n() and Deinit_n() that are currently*
required to be called by external code in order to associate/disassociate
the channels with the transport.
This CL mostly moves things around in order to prepare for upcoming
changes, but it does change channel destruction in the following way:
- When destroying channels, we don't block the worker thread anymore
while uninitialization happens on the network thread. Previously
both signal and worker threads were blocked during the
uninitialization in the ChannelManager.
* In an upcoming CL, Init_n() and Deinit_n() will be called internally
from a different method that's always called on the network thread
when a channel is associated/disassociated with a transceiver. When
we're there, we will have removed several invokes that currently are
a part of constructing/destructing channel objects.
Bug: webrtc:11992
Change-Id: Ibc30447a40749ceb36d37834b0cfc5c5ea60e895
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246502
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35707}
This is an operation specific subclass of SdpOfferAnswerHandler that in
this first step, takes over the implementation details that before this
CL were implemented in SdpOfferAnswerHandler::DoSetRemoteDescription.
This CL does not change the behavior of the implementation but it does
break up DoSetRemoteDescription into smaller methods and moves the state
related to the SRD operation, into a class that in upcoming steps can
be passed around asynchronously as needed, which will allow us to avoid
blocking threads.
Bug: webrtc:13540
Change-Id: Id2002d2390a4a13725f5967df5b82064b37c7490
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244980
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35669}
This changes SetLocalDescription/SetRemoteDescription to just resetting
the internal cache timestamp for the legacy stats handler instead of
performing a full update, which can be costly.
Bug: webrtc:13557
Change-Id: I93971dbd7abf33c0d0f2836f9c17ba4550f41a00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245645
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35661}
This part is specific to unified plan and doesn't need most of
the state related to the remote description (and doesn't return an
error).
Bug: none
Change-Id: I0de66bdb2e925072a6d9010e4444e75d4574ae04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245102
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35642}
This is just a step to reduce the size of ApplyRemoteDescription to make
refactoring it easier (and ultimately support async operations).
Bug: none
Change-Id: Idb950c35f585a887d6640278b6edfdd0c7cec3fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245101
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35641}
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 simplifies the work that happens on the worker thread in
preparation of avoiding having to go to the worker at all.
Bug: webrtc:11993
Change-Id: I13f063bdecce8efdb978ef1976c819019f30e020
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244082
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35610}
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}
which is the maxium length allowed by one-byte header extensions
BUG=webrtc:12517
Change-Id: I003105d3566a34b5b7affb84ffe69b7705973ee3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35333}
