On a 32bit system, this reduces the allocation size of the flag
down from 12 bytes to 8, and removes the need for a vtable (the extra
4 bytes are the vtable pointer).
The downside is that this change makes the binary layout of the
flag, less compatible with RefCountedObject<> based reference counting
objects and thus we don't immediately get the benefits of identical
COMDAT folding and subsequently there's a slight binary size increase.
With wider use, the binary size benefits will come.
Bug: none
Change-Id: I04129771790a3258d6accaf0ab1258b7a798a55e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215681
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33793}
This is a refactor to simplify a follow-up CL of adding
SdpVideoFormat::IsSameCodec.
The original files media/base/h264_profile_level_id.* and
media/base/vp9_profile.h must be kept until downstream projects
stop using them.
Bug: chroimium:1187565
Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33782}
This denies the ability to request RTP data channels to callers.
Later CLs will rip out the actual code for creating these channels.
Bug: chromium:928706
Change-Id: Ibb54197f192f567984a348f1539c26be120903f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177901
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33740}
"enableImplicitRollback" is necessary for perfect negotiation algorithm
"offerExtmapAllowMixed" is necessary for backward compatibility with
legacy clients.
Bug: webrtc:12609
Change-Id: I30a5a01c519ca9080a346e2d36b58f7bab28f15a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212741
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33639}
PC.restartIce() is part of perfect negotiation algorithm.
Bug: webrtc:12609
Change-Id: I21a0f8637e92e13ee2653ef477d0cd22a32bf9c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212645
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33589}
Instead of using SetAlive on the old flag (which might allow old
tasks in the queue to run).
Bug: webrtc:12339
Change-Id: Ia1a3eb6932f62881f013fd62b0e008d97d8713cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211863
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33475}
This reverts commit 1e60490ddb11afecc796058802fbc80867d851d8.
Reason for revert: Downstream project has been fixed (127.0.0.1 is not PII)
Original change's description:
> Revert "Fix problem with ipv4 over ipv6 on Android"
>
> This reverts commit da2fd2a2b25ee4bd7b383424cb26d51fb6cc7716,
> as well as follow-up b7227a5a10f233cec04642f15a0233e7355bd340,
> "Fix handling of partial match for GetVpnUnderlyingAdapterType".
>
> Reason for revert: Breaks downstream test.
>
> First change's description:
> > Fix problem with ipv4 over ipv6 on Android
> >
> > This patch fixes a problem with using ipv4 over ipv6
> > addresses on Android. These addresses are discovered
> > using 'getifaddr' with interfaces called 'v4-wlan0' or
> > 'v4-rmnet' but the Android API does not report them.
> >
> > This leads to failure when BasicPortAllocator tries
> > to bind a socket to the ip-address, making the ipv4
> > address unusable.
> >
> > This solution does the following
> > 1) Insert BasicNetworkManager as NetworkBinderInterface
> > rather than AndroidNetworkManager.
> >
> > 2) When SocketServer calls BindSocketToNetwork,
> > BasicNetworkManager first lookup the interface name,
> > and then calls AndroidNetworkManager.
> >
> > 3) AndroidNetworkManager will then first try to bind
> > using the known ip-addresses, and if it can't find the network
> > it will instead match the interface names.
> >
> > The patch has been tested on real android devices, and works fine.
> > And everything is disabled by default, and is enabled by field trial.
> >
> > My plan is to rollout the feature, checking that it does not introduce
> > any problems, and if so, enabled for all.
> >
> > Bug: webrtc:10707
> > Change-Id: I7081ba43d4ce17077acfa5fbab44eda127ac3971
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211003
> > Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33422}
>
> Second change's description:
> > Fix handling of partial match for GetVpnUnderlyingAdapterType
> >
> > This is a followup to https://webrtc-review.googlesource.com/c/src/+/211003
> > and fixes the problem pointed out by deadbeef@, thanks!
> >
> > Bug: webrtc:10707
> > Change-Id: I8dea842b25ba15416353ce4002356183087873c7
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211344
> > Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33436}
>
> TBR=hta@webrtc.org,jonaso@webrtc.org
> NOTRY=True
>
> Bug: webrtc:10707
> Change-Id: Ib13127fbf087c7f34ca0ccc6ce1805706f01d19d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211740
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33453}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10707
Change-Id: I0a11025c366c3127e2f57cd2cd2c33cc3877d1e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211862
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33460}
This reverts commit da2fd2a2b25ee4bd7b383424cb26d51fb6cc7716,
as well as follow-up b7227a5a10f233cec04642f15a0233e7355bd340,
"Fix handling of partial match for GetVpnUnderlyingAdapterType".
Reason for revert: Breaks downstream test.
First change's description:
> Fix problem with ipv4 over ipv6 on Android
>
> This patch fixes a problem with using ipv4 over ipv6
> addresses on Android. These addresses are discovered
> using 'getifaddr' with interfaces called 'v4-wlan0' or
> 'v4-rmnet' but the Android API does not report them.
>
> This leads to failure when BasicPortAllocator tries
> to bind a socket to the ip-address, making the ipv4
> address unusable.
>
> This solution does the following
> 1) Insert BasicNetworkManager as NetworkBinderInterface
> rather than AndroidNetworkManager.
>
> 2) When SocketServer calls BindSocketToNetwork,
> BasicNetworkManager first lookup the interface name,
> and then calls AndroidNetworkManager.
>
> 3) AndroidNetworkManager will then first try to bind
> using the known ip-addresses, and if it can't find the network
> it will instead match the interface names.
>
> The patch has been tested on real android devices, and works fine.
> And everything is disabled by default, and is enabled by field trial.
>
> My plan is to rollout the feature, checking that it does not introduce
> any problems, and if so, enabled for all.
>
> Bug: webrtc:10707
> Change-Id: I7081ba43d4ce17077acfa5fbab44eda127ac3971
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211003
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33422}
Second change's description:
> Fix handling of partial match for GetVpnUnderlyingAdapterType
>
> This is a followup to https://webrtc-review.googlesource.com/c/src/+/211003
> and fixes the problem pointed out by deadbeef@, thanks!
>
> Bug: webrtc:10707
> Change-Id: I8dea842b25ba15416353ce4002356183087873c7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211344
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33436}
TBR=hta@webrtc.org,jonaso@webrtc.org
NOTRY=True
Bug: webrtc:10707
Change-Id: Ib13127fbf087c7f34ca0ccc6ce1805706f01d19d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211740
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33453}
Use PendingTaskSafetyFlag for safe Stop. Followup to
https://webrtc-review.googlesource.com/c/src/+/209181.
Also fix rtc::scoped_refptr to work with RTC_PT_GUARDED_BY.
Bug: webrtc:12339
Change-Id: Ic0e3ecb17049f1a0e6af887ba5f97a5b48a32d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211351
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33447}
This patch fixes a problem with using ipv4 over ipv6
addresses on Android. These addresses are discovered
using 'getifaddr' with interfaces called 'v4-wlan0' or
'v4-rmnet' but the Android API does not report them.
This leads to failure when BasicPortAllocator tries
to bind a socket to the ip-address, making the ipv4
address unusable.
This solution does the following
1) Insert BasicNetworkManager as NetworkBinderInterface
rather than AndroidNetworkManager.
2) When SocketServer calls BindSocketToNetwork,
BasicNetworkManager first lookup the interface name,
and then calls AndroidNetworkManager.
3) AndroidNetworkManager will then first try to bind
using the known ip-addresses, and if it can't find the network
it will instead match the interface names.
The patch has been tested on real android devices, and works fine.
And everything is disabled by default, and is enabled by field trial.
My plan is to rollout the feature, checking that it does not introduce
any problems, and if so, enabled for all.
Bug: webrtc:10707
Change-Id: I7081ba43d4ce17077acfa5fbab44eda127ac3971
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211003
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33422}
This is consistent with other Notify methods in this class, which
handle callbacks from java using blocking invokes to the network
thread.
This eliminates the use of the deprecated AsyncInvoker class.
Bug: webrtc:12339
Change-Id: Ib2d19b37b8f669df5b97e89d720f6eb6fc9e5517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209181
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33398}
Some locations in the WebRTC codebase RTC_LOG the value of the
__FUNCTION__ macro which probably is useful in debug mode. Moving
these instances to RTC_DLOG saves ~10 KiB on arm64.
Bug: webrtc:11986
Change-Id: I5d81cc459d2850657a712b9aed80c187edf49a3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203981
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33086}
This is just general cleanup.
The assumed behavior is late decoding, and this function is not used to make any decision (except in the deprecated jitter buffer).
Bug: webrtc:12271
Change-Id: Ifb48186d55903f068f25e44c5f73e7a724f6f456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200804
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32940}
This CL makes it possible to use a low-latency mode on Android O and later. This should help to reduce the audio latency. The feature is disabled by default and needs to be enabled when creating the audio device module.
Bug: webrtc:12284
Change-Id: Idf41146aa0bc1206e9a2e28e4101d85c3e4eaefc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196741
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32854}
In particular move end_of_picture flag out of vp9 specific information
since VP9 is not the only codec that can use spatial scalability and
thus need to distinguish layer frame and picture (aka temporal unit).
Bug: webrtc:12167
Change-Id: I0d046d8785fbea55281209ad099738c03ea7db96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192542
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32588}
And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.
Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
In the old jitter buffer the two VCMVideoProtection modes |kProtectionNone| and |kProtectionFEC| could be set on the jitter buffer for it to not wait for NACK and instead generate incomplete frames. This has not been possible for a long time.
Bug: webrtc:9378, webrtc:7408
Change-Id: I0a2d3ec34d721126c1128306d5fad88314f8d59f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32513}
(Reland with no changes after the fix to the downstream project)
This can be overriden for kNative frame types to perform scaling efficiently.
Default implementations for existing buffer types require actual
buffer implementation, thus this CL also merges "video_frame"
with "video_frame_I420" build targets.
Originally Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303
(Landing with TBR as it's unchaged reland of already approved CL)
TBR=nisse@webrtc.org,sakal@webrtc.org
Bug: webrtc:11976, chromium:1132299
Change-Id: Ia23f7d3e474bd9cdc177104cc5c6d772f04b210f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187345
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32362}
Since https://webrtc-review.googlesource.com/c/src/+/161447,
frame_extra_info_ is modified from both the encoder thread and the
callback thread.
Bug: None
Change-Id: Idece4d19cae6d8363428234721616f6ca6f85832
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184121
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32095}
All it provides is a method to call a signal on the network thread,
so it's not worth the added complexity. Implementations of
NetworkMonitorInterface must hop to the network thread anyway to
guard their members.
Also added some thread annotations to AndroidNetworkMonitor.
Bug: webrtc:9883
Change-Id: I64bb82ea593433f3a52871dbb75eb2ac4f47d69c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181420
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32087}
As documented in webrtc:11908 this cleanup is fairly invasive and
when a part of a frequently executed code path, can be quite costly
in terms of performance overhead. This is currently the case with
synchronous calls between threads (Thread) as well with our proxy
api classes.
With this CL, all code in WebRTC should now either be using MessageHandlerAutoCleanup
or calling MessageHandler(false) explicitly.
Next steps will be to update external code to either depend on the
AutoCleanup variant, or call MessageHandler(false).
Changing the proxy classes to use TaskQueue set of concepts instead of
MessageHandler. This avoids the perf overhead related to the cleanup
above as well as incompatibility with the thread policy checks in
Thread that some current external users of the proxies would otherwise
run into (if we were to use Thread::Send() for synchronous call).
Following this we'll move the cleanup step into the AutoCleanup class
and an RTC_DCHECK that all calls to the MessageHandler are setting
the flag to false, before eventually removing the flag and make
MessageHandler pure virtual.
Bug: webrtc:11908
Change-Id: Idf4ff9bcc8438cb8c583777e282005e0bc511c8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183442
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32049}
This patch adds a computed estimate on how long the ice stack
was disconnected before switching to a new connection.
The metric is currently computed as now - max(connection->last_data_recevied())
and has resonably good precision.
Bug: webrtc:11862
Change-Id: I8950d55f0eadcf164de089cdb715b4f7eed0a4c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182002
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31969}
This clears the decks for deprecating and eventually removing
the nonstandard SetDirection method.
Bug: chromium:980879
Change-Id: Ibc291de3db690e9ef4e6cb3550390d7728f02a83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181860
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31948}
This is a reland of 11dc6571cb4ff3e71dee1557dfff8d9076e108d3
One fix that makes Web Platform Tests pass in debug mode is applied.
Original change's description:
> Implement transceiver.stop()
>
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
>
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
>
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}
Bug: chromium:980879
Change-Id: Ide31d929ac5ea118d83fdf6a35a592af23f7dfa7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181263
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31907}
This reverts commit 11dc6571cb4ff3e71dee1557dfff8d9076e108d3.
Reason for revert: Breaks Chromium WPT tests
Original change's description:
> Implement transceiver.stop()
>
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
>
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
>
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}
TBR=sakal@webrtc.org,kthelgason@webrtc.org,hta@webrtc.org,guidou@webrtc.org,marinaciocea@webrtc.org
Change-Id: Ibdc24f7d41e481293ca74ba6d1572de64f7e4654
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:980879
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181262
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31897}
This adds RtpTransceiver.StopStandard(), which behaves according to
the specification at
https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
It modifies RTCPeerConnection.getTransceivers() to return only
transceivers that have not been stopped.
Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
Bug: chromium:980879
Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31893}
