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}
After the refactoring, the test fixture is only used for creating the
object under test and dependencies. This leads to more readable code and
allows more flexibility when creating the object under test.
Bug: none
Change-Id: I643330290da17efe0a02fe5dc6b884136705de0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248140
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35770}
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}
This reverts commit 3b18208f13e85b356e61a95c0a261e9781403743
and is the third attempt at removing stun origin support
Bug: webrtc:12132
Change-Id: Ic41a6d011fb6239907a257cc4c81ec4d2923dc4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236260
Reviewed-by: Taylor Brandstetter <deadbeef@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35294}
This is a reland of ba29ce320fe1f9ac69b0ff8eb50fbe402c2912a6
readding the origin to the CreateRelayPortArgs structure to not break
downstream tests yet:
https://webrtc-review.googlesource.com/c/src/+/235300/1..2
Original change's description:
> remove stun origin support
>
> Bug: webrtc:12132
> Change-Id: I0f32e6af77e0c553b0c3b0d047ff03e14c492b31
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234384
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35202}
Bug: webrtc:12132
Change-Id: Ied840b59bb7c9497e98f9b80eb0a54d30008a40f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35220}
The explicitly defined constructor suppresses the assignment operator,
which blocks the chromium roll.
Bug: b/198565646
Change-Id: I35917d4b99ad86dcf8b9863e798f5a63d9073824
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231123
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34904}
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}
GICE was removed around M42
BUG=webrtc:4299
Change-Id: I4e83a888c3ecc1681799c07b47b75c9f31b40baa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227348
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34680}
When the transport is terminated, if an error has occured, it will
be propagated to the channels.
When such errors can happen at the SCTP level (e.g. out of resources),
RTCError may contain an error code matching the definition at
https://www.iana.org/assignments/sctp-parameters/sctp-parameters.xhtml#sctp-parameters-24
If the m= line is rejected or removed from SDP, an error will again be sent
to the data channels, signaling their unexpected transition to closed.
Bug: webrtc:12904
Change-Id: Iea3d8aba0a57bbedb5d03f0fb6f7aba292e92fe8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223541
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34386}
It doesn't make sense to use negative values or 0 to disable the
feature, so we use an optional int value.
Values bigger than 65535 are clamped down.
Bug: webrtc:12730
Change-Id: I6bd9cd92f7d0a70a78cf5a7c91dca52c28d08ba1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33954}
The new verification makes verification a function on a message.
It also stores the password used in the request message, so that
it is easily accessible when verifying the response.
Bug: chromium:1177125
Change-Id: I505df4b54214643a28a6b292c4e2262b9d97b097
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209060
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33366}
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.
Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a
Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33001}
This reverts commit 69241a93fb14f6527a26d5c94dde879013012d2a.
Reason for revert: Breaks WebRTC roll into Chromium.
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
TBR=mbonadei@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: True
Bug: webrtc:9987
Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32986}
The "//rtc_base:rtc_base" build target has historically been one of the
biggest targets in the WebRTC build. Big targets are the main source of
circular dependencies and non-API types leakage.
This CL is a step forward into splitting "//rtc_base:rtc_base" into
smaller targets (as originally started in 2018).
The only non-automated changes are (like re-wiring the build system):
* The creation of //rtc_base/async_resolver.{h,cc} which allows to
break a circular dependency (is has been extracted from
//rtc_base/net_helpers.{h,cc}).
* The creation of //rtc_base/internal/default_socket_server.{h,cc} to
break another circular dependency.
Bug: webrtc:9987
Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32941}
This will cause encoding of a STUN message with an over-long
byte string attribute to fail.
Bug: chromium:1144646
Change-Id: I265174577376ce01439835c03f2d46700842d211
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191322
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32603}
The former was unused, the latter is replaced with the explicit C++11
deletions. The related RTC_DISALLOW_COPY_AND_ASSIGN is left for now,
it is used in a lot more places.
Bug: None
Change-Id: I49503e7f2b9ff43c6285f8695833479bbc18c380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32224}
This is to allow testing without using the singleton sctp library.
cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure.
Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation.
patch 1 contain the original cl.
patch 2 modifications
Bug: none
Change-Id: Ic088da3eb7d9aada79e6d601dbf2d1aa2be777f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182840
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32024}
This reverts commit 4c0a381137c04fd80830af8a041e25e3428dd33f.
Reason for revert: Breaks downstream test
Original change's description:
> Make cricket::SctpTransportInternalFactory injectable through PeerConnectionFactory Deps
>
> This is to allow testing without using the singleton sctp library.
> cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure.
> Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation.
>
> Bug: none
> Change-Id: I482241269463595062548870750d33f31238c6b1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182082
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32007}
TBR=deadbeef@webrtc.org,mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I46d5ba89fe723caccd065b0ac41d77ed45373838
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32008}
This is to allow testing without using the singleton sctp library.
cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure.
Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation.
Bug: none
Change-Id: I482241269463595062548870750d33f31238c6b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182082
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32007}
i.e. when chain are used,
require each decode target to be protected by some chain.
where previously it was allowed to mark decode target as unprotected.
See https://github.com/AOMediaCodec/av1-rtp-spec/pull/125
Bug: webrtc:10342
Change-Id: Ia2800036e890db44bb1162abfa1a497ff68f3b24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178807
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31772}
This interface has a couple of issues. Primarily for me, it makes it
difficult work with the paced sender as we need to either temporarily
release a lock or force a thread-handover in order to avoid a cyclic
lock order.
For video in particular, its behavior is also falky since header sizes
can vary not only form frame to frame, but from packet to packet within
a frame (e.g. TimingInfo extension is only on the last packet, if set).
On bitrate allocation, the last reported value is picked, leading to
timing issues affecting the bitrate set.
This CL removes the callback interface and instead we simply poll the
RTP module for a packet overhead. This consists of an expected overhead
based on which non-volatile header extensions are registered (so for
instance AbsoluteCaptureTime is disregarded since it's only populated
once per second). The overhead estimation is a little less accurate but
instead simpler and deterministic.
Bug: webrtc:10809
Change-Id: I2c3d3fcca6ad35704c4c1b6b9e0a39227aada1ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173704
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31185}
If a STUN attribute is in the "comprehension-required" range
(0x0000-0x7FFF), and the implementation does not recognize it, this
should be treated as an error (as per RFC5389), with different behavior
depending on the type of the message received.
Bug: webrtc:9063
Change-Id: Ic31b0cdd3c26772c21d770b44fe4ee4a1b47030a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/64500
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30925}
The decision to route audio packets to a separate overuse detector
is off by default and requires the field trial
WebRTC-Bwe-SeparateAudioPackets/enabled,packet_threshold:10,time_threshold:1000ms/
The parameters control the threshold for switching over to the
audio overuse detector if we stop receiving feedback for video.
Bug: webrtc:10932
Change-Id: Icdde35bc7a98b18b1a344bd2d620a890fd9421d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168342
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30694}
With current congestion window pushback, when congestion window is filling up, it will reduce bitrate directly and encoder may reduce encode quality, resolution, or framerate to adapt to the allocated bitrate, the behavior is depending on the degradation preference.
This change enable congestion window to only drop frames to reduce bitrate (when needed) instead of reduce general bitrate allocation.
Bug: webrtc:11334
Change-Id: I9cf5c20a0858c4d07d006942abe72aa5e1f7cb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168059
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30483}
Currently, datagram transports must report identical transport
parameters in order to negotiate use of the datagram transport. This is
not strictly necessary, they just need parameters that fit some notion
of "compatability" (eg. both ends share some mutually-supported version
of the datagram protocol).
This change allows datagram transports to implement their own notion of
compatible transport parameters, by adding a
SetRemoteTransportParameters method to DatagramTransportInterface which
checks if the remote parameters are compatible with the local endpoint
and returns an error if they are not.
Bug: webrtc:9719
Change-Id: I166c787b468b89d9082d7e3c9995a6ed50a1650a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167741
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30412}
