Notably, this should detect whether an interface is "available" or not,
which should prevent the failure is with dual SIM card setups.
This is gated behind a field trial for now, to ensure this doesn't cause
any regressions due to false negatives (interfaces that are usable
but not listed as available by NWPathMonitor).
Bug: webrtc:10966
Change-Id: Ia3942c4c57b525d08d8b340e2325f3705cfd0304
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180923
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31977}
Also moves implementation of legacy setDirection() without error to the
api/ directory.
This is one step in the plan for changing the API
to return RTCError.
Bug: chromium:980879
Change-Id: Ibce8edf8e3c6d41de7ce49d2ffc33f5b282a0e9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181520
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31943}
This is a reland of 11dc6571cb
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 11dc6571cb.
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}
Since the descriptions can be modified on the signaling thread,
ToString can only be safely called on that thread.
Bug: webrtc:11791
Change-Id: Icf6aada8aa66d00be94c6bda7b22e41b5d3bbc17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180541
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31862}
This member of the CodecInfo struct was set in several places, but not
used for anything. To aid deletion, this cl defines a default implementation
of VideoEncoderFactory::QueryVideoEncoder.
The next step is to delete almost all downstream implementations of that method,
since the only classes that have to implement it are the few factories that
produce "internal source" encoders, e.g., for Chromium remoting.
Bug: None
Change-Id: I1f0dbf0d302933004ebdc779460cb2cb3a894e02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179520
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31844}
This name change communicates that the recursive critical section
should not be used for new code.
The relevant files are renamed rtc_base/critical_section* ->
rtc_base/deprecated/recursive_critical_section*
Bug: webrtc:11567
Change-Id: I73483a1c5e59c389407a981efbfc2cfe76ccdb43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179483
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31754}
These two types need to be exported in order to access the stats report
from an ObjC / Swift codebase.
Bug: webrtc:11158
Change-Id: Ibb2f81f289b56f824f02df70971c28accd5a1350
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174900
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31257}
This CL introduced 2 new macros that affect the WebRTC OBJC API symbols:
- RTC_OBJC_TYPE_PREFIX:
Macro used to prepend a prefix to the API types that are exported with
RTC_OBJC_EXPORT.
Clients can patch the definition of this macro locally and build
WebRTC.framework with their own prefix in case symbol clashing is a
problem.
This macro must only be defined by changing the value in
sdk/objc/base/RTCMacros.h and not on via compiler flag to ensure
it has a unique value.
- RCT_OBJC_TYPE:
Macro used internally to reference API types. Declaring an API type
without using this macro will not include the declared type in the
set of types that will be affected by the configurable
RTC_OBJC_TYPE_PREFIX.
Manual changes:
https://webrtc-review.googlesource.com/c/src/+/173781/5..10
The auto-generated changes in PS#5 have been done with:
https://webrtc-review.googlesource.com/c/src/+/174061.
Bug: None
Change-Id: I0d54ca94db764fb3b6cb4365873f79e14cd879b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31153}
network_priority was already exposed, but without the ability to set
enable_dscp, it wasn't actually doing anything.
Bug: webrtc:5658
Change-Id: I092bc3dd46e3e7be363313203428bccfccccf3c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171641
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30951}
This reverts commit 86e0ea5711.
Reason for revert: The reasons for removing bitratePriority are unclear. Aside from the fact that you can't yet use it for the relative bitrate of simulcast streams, only the relative bitrate of entire tracks, it's working as intended. It differs from the standard, but only in that it's more flexible; the web standard only allows values of 0.5, 1.0, 2.0, and 4.0 while for the native API we allow any ratio.
Original change's description:
> Remove bitratePriority from the Obj-C RTCRtpEncodingParameters wrapper.
>
> This was added in CL 135122, but the bitratePriority parameter is not
> standard and not implemented in a way users would expect. So it should
> actually not be exposed in the Obj-C SDK.
>
> Bug: webrtc:10438
> Change-Id: I801ce940a32701d2703e951ef2b601c606aa2111
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135287
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27861}
TBR=andersc@webrtc.org,kthelgason@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10438
Change-Id: Ibc16b6054a1583de43a868d98683ea114bd89435
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171140
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30863}
This is a reland of de86381161
Original change's description:
> Leverage dispatch_queue_create_with_target when possible.
>
> Replacing dispatch_queue_create followed by
> dispatch_set_target_queue with dispatch_queue_create_with_target
> is claimed to be source of GCD performance improvement:
> https://developer.apple.com/videos/play/wwdc2017/706/
> Video since 40 min. Slides since 199.
>
> Bug: webrtc:9055
> Change-Id: I0136f7faaef0951a7ad243bc8772f3ee952d5470
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168491
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
> Cr-Commit-Position: refs/heads/master@{#30781}
Bug: webrtc:9055
Change-Id: I36b0b6423c81c0497f66f7c993741c33ff6ec5ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170443
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30821}
Removes support for parsing and serializing
a=ssrc:1 mslabel:stream
a=ssrc:1 label:track
which have been superceeded by
a=ssrc:1 msid:stream track
a long time ago.
Bug: webrtc:7110
Change-Id: I3aca47728098b6e7e049b82ed34c59426d411c41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168244
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30801}
Replacing dispatch_queue_create followed by
dispatch_set_target_queue with dispatch_queue_create_with_target
is claimed to be source of GCD performance improvement:
https://developer.apple.com/videos/play/wwdc2017/706/
Video since 40 min. Slides since 199.
Bug: webrtc:9055
Change-Id: I0136f7faaef0951a7ad243bc8772f3ee952d5470
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168491
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/master@{#30781}
The degradation preference is now based on the content hint of the track
if it's unspecified.
Bug: webrtc:11164
Change-Id: Iaa0dbf1c1bf68a46fc5131e534d423c30c5439c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161233
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30691}
It can only be one of four possible values, so it never made sense
for it to be a double. Other than the fact that its neighbor
bitrate_priority is a double, and they're both defined as the same enum
in the web spec. However, while bitrate_priority being a double
offers more flexibility than the web spec, network_priority being a
double is only confusing.
Bug: webrtc:5658
Change-Id: I0784c116f3260c4b3a8b99a3cd85c8d66017e46f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168840
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30685}
This patch exposes webrtc::PeerConnectionDependencies c++-object
and makes it possible to supply one when creating a PeerConnection.
This makes it possible to e.g inject a VideoBitrateAllocatorFactory.
Bug: webrtc:10547
Change-Id: Ib7431bdcec1380e7903dc5f66f3583501aeab0a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168307
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30480}
This was an ICE configuration experiment added a couple years ago that did not end up being used.
Bug: webrtc:11316
Change-Id: Iafb7e1c4f7b4598815f045808dbf6e470172f119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167680
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30395}
This will avoid getting into an inconsistent state where isInterrupted==YES while isActive==YES.
Bug: webrtc:11112
Change-Id: Ia4db85483e1e7a339f520d52a2feb475a73c262e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160140
Commit-Queue: Joe Chen <jsphchn@google.com>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30306}
Since rtc::Thread is the only class inheriting from rtc::MessageQueue
and most members of MessageQueue are public or protected the split is
not adding much value. In preparation for future cleanup, this cl merges
the two classes.
Bug: webrtc:9883
Change-Id: Ia0efb4349f66f653aa34fa4d244998f187e3ce36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165340
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30235}
1. On iOS13 the implementation of methods begin- and endGeneratingDeviceOrientationNotifications changed and now are looks like "not threadsafe" (in specific sence) - they should be called only on the main thread. This fact is not documented. And may be a mistake.
2. By the Apple official documentation methods begin- and endGeneratingDeviceOrientationNotifications should be balanced. (Each begin- method should be balanced with end- method.)
By the reason two above facts they consequences merged and produced the "floating" NSInternalInconsistencyException crash.
Bug: webrtc:11216
Change-Id: Ibedd5bba7476cc687de3b9b04be49e3cceac1d27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162205
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30121}
This reverts commit f22af3cca7.
Reason for revert: Downstream tests have been updated.
Original change's description:
> Revert "Distinguish between send and receive video codecs"
>
> This reverts commit 18314bd8d2.
>
> Reason for revert: Breaks downstream test.
>
> Original change's description:
> > Distinguish between send and receive video codecs
> >
> > Even though send and receive codecs are the same,
> > they might have different support in HW.
> > Distinguish between send and receive codecs to be able to keep
> > track of which codecs have HW support.
> >
> > Bug: chromium:1029737
> > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30041}
>
> TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
>
> Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30042}
TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1029737
Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30078}
This reverts commit 18314bd8d2.
Reason for revert: Breaks downstream test.
Original change's description:
> Distinguish between send and receive video codecs
>
> Even though send and receive codecs are the same,
> they might have different support in HW.
> Distinguish between send and receive codecs to be able to keep
> track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30041}
TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30042}
Even though send and receive codecs are the same,
they might have different support in HW.
Distinguish between send and receive codecs to be able to keep
track of which codecs have HW support.
Bug: chromium:1029737
Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30041}
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.
Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
This method is called on every GetStats call and fills up log output on iOS
with three log lines per cycle at INFO+ (the not-supported one is LS_ERROR):
[181:040] [82471] (audio_device_module_ios.mm:646): GetPlayoutUnderrunCount
[181:040] [82471] (audio_device_generic.cc:48): GetPlayoutUnderrunCount: Not supported on this platform
[181:040] [82471] (audio_device_module_ios.mm:649): output: -1
Alternatively, we could remove the error logging in the base class, or (better) log it once the first time it is called, but this is the simpler change.
Bug: None
Change-Id: Ibaa1d176f10cdc92f2ba1a6bf15aaa580da6edb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159672
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29797}
This happend because sdk_unittests is not built on arm/arm64 iOS build.
Bug: webrtc:11022
Change-Id: I8f9adfd48e11c8512c27992804cc9b69dff15ded
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156100
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29407}
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.
Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
Some devices have issues decoding the resolutions that result when using 4
as a factor.
Bug: webrtc:9381
Change-Id: I5055923ca318a1bde62bcefb452cae8f33165e43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150102
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28945}
This silences a warning that appeared with iOS 13, and is more efficient
in general.
Bug: webrtc:10866
Change-Id: I23db6b78af36e59b1d825d3f0cccc6008f9b626a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149808
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28911}
Bug: webrtc:10419
Change-Id: I18528bf2526e933568bf052de76a434f012161da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148320
Commit-Queue: Alex Drake <alexdrake@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28838}
This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
plus a ton of now-dead code.
Bug: webrtc:10556
Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28523}
This is a partial reland of
https://webrtc-review.googlesource.com/c/src/+/39680,
including only the (hopefully) non-problematic parts of it, but
postponing actual deletion of automatic thread wrapping.
Bug: webrtc:9714
Change-Id: I9b79dd073f0e945cbb62f3b54cff05eaaea9b06c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141664
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28265}
This is a reland of 11dfff0878
Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org
Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
This reverts commit 11dfff0878.
Reason for revert: Downstream import failure.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org
Change-Id: I7f833055c67f1f879b01dd8c156ba7b8840e8747
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141411
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28225}
After this CL lands, an announcement will be made to
discuss-webrtc about the deprecation of one version
of InitEncode().
Bug: webrtc:10720
Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28224}
from a field trial to RTCConfiguration.
The test coverage is also expanded for the underlying feature.
Bug: None
Change-Id: Ic9c1362867e4a956c5453be7a9355083b6a442f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138980
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28143}
This CL adds the callback on changes of the ICE connection state
following the standardized transitions
(https://www.w3.org/TR/webrtc/#dom-rtciceconnectionstate) to the
Android and the iOS SDKs.
Bug: None
Change-Id: I6133391fa54dd4e09016f29dddb85e4a0e270878
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138181
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28127}
When the LossNotifications field trial is in effect, LNTF should
be offered/accepted in the SDP message, not assumed to be configured
on both sides equally.
Bug: webrtc:10662
Change-Id: Ibd827779bd301821cbb4196857f6baebfc9e7dc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138079
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28056}
Since many tests rely on rtc::Thread::Current(), add an
explicit rtc::AutoThread in the main() function used by tests.
Bug: webrtc:9714
Change-Id: Id82121967c9621fe1c2945846009c48139fa57da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/39680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28000}
This fixes a potential crash if interrupted before the audio unit has been initialized.
Bug: None
Change-Id: Ib9f5ea305c98a172f8df52af5767c8543e59701c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136800
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27937}
This will allow call audio to function when audio session attributes like `preferredInputNumberOfChannels` cannot be set due to intermittent OS errors.
Bug: webrtc:10602
Change-Id: Ie9f3e58a6ab54a26a9bd795575d16c3a9fe5c65f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135440
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27871}
