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}
This explores the theory that targets that have no files, just
dependencies, are unnecessary.
Bug: webrtc:13805
Change-Id: I1feb50cf3886128031af8970eae361e35fb052c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256974
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36363}
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}
Because rtc::Thread inherits from TaskQueueBase, it already implements
a pair of PostTask/PostDelayedTask methods that we want to keep. But in
addition to those, rtc::Thread defines its own PostTask/PostDelayedTask
using templates. These are the versions that we want to deprecate.
They were originally implemented prior to rtc::Thread inheriting from
TaskQueueBase. We want to deprecate them because...
- We don't want to have multiple code paths that do the same thing.
- We want to move away from rtc::Thread to TaskQueueBase long-term.
- These versions are not overridable in Chromium.
- These versions don't have high/low precision versions of PDT.
Helper methods are added to rtc::Thread so that callers don't have to
wrap every lambda in webrtc::ToQueuedTask() and update dependencies.
Bug: webrtc:13582
Change-Id: I58702c53f4cb3705681bd9f1ea16b7aaa5052c18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35750}
Add TODO for accessing `previous_demuxer_criteria_`, currently accessed
from two threads (unsafe).
Changed RtpDemuxerCriteria to be a class, all members private with
accessor methods instead of direct variable access. Moving forward
this can allow for things like checking for thread/sequence and state
consistency.
Bug: webrtc:12517, webrtc:11993
Change-Id: I21c1b3067e988494ce6f4c6c85c62165801883bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244083
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35616}
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}
Deletes the helper methods SdpSetObserver and SdpCreateObserver,
replaced with observer classes where used, in peer_scenario_client.cc.
Deletes the class webrtc_sdp_obs_impl::SdpSetObserversInterface, which
indirectly inherits rtc::RefCountInterface twice. Migrates this code
to use rtc::make_ref_counted, and migrates away from deprecated
versions of SetLocalDescription and SetRemoteDescription that use raw
pointers and SetSessionDescriptionObserver.
Bug: webrtc:12701, webrtc:11798
Change-Id: I18ea3fb51f533d7454a6dc75292b1827b1c80ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229981
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34843}
As a side effect, break out pc/simulcast_description.
Step 1: Don't move the {h,cc} files; just declare the targets
so that downstream projects can add dependencies on it.
Bug: webtc:11967
Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34254}
We want to turn off PT based demux because SSRC-based endpoints that
send media prematurely (which is a popular non-standard behavior still
heavily in use) can otherwise get incorrect mappings and unsignalled
ssrc issues because of the PT demux path.
This CL disables PT based demuxing when the MID header extension is
present on all m= sections in the SDP for that kind (audio/video), not
caring if it was in the offer or answer. However if PT demuxing has been
used in the past then it is always allowed. This ensures PT is off by
default but that either offer or answer can enable PT and once it has
been on it is also possible to get early media with PT.
- Want PT-based demux? The MID header extension has to be removed in
either the offer or the answer. Follow-up O/As allow PT demuxing if
possible.
- Want to use MID or SSRC demuxing? Great, you don't need PT-based demux
and won't mind that we turned it off for you.
The reason for disabling PT demux at offer time (if MID is present)
instead of waiting for the SDP answer is because by the time the SDP
answer arrives, early media could have triggered PT demux and caused
incorrect mappings. The safe thing is to assume a spec-compliant
endpoint until proven otherwise.
However if PT demux is ever enabled, then from that point on we always
allow PT-based demux in follow-up O/A exchanges. This ensures we don't
drop packets in follow-up exchanges. The fact that PT-based demux is
disabled during the initial offer should not matter because before the
initial O/A exchange we don't have fingerprints.
This change only affects Unified Plan and bundled groups. Existing test
coverage ensuring we do not break legacy endpoints:
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/peer_connection_integrationtest.cc;l=1156
[2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/protocol/rtp-demuxing.html;l=59
UnsignaledStreamTest is also updated to test the interesting setups.
A kill-switch is added in case we want to disable this change.
Bug: webrtc:12814
Change-Id: I807a82a543325753633aaef698e06cb4c9dfebaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221101
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34251}
Applications should use CreatePeerConnectionOrError instead.
Moved fallback implementations of CreatePeerConnection into the
api/peer_connection_interface.h file, so that we do not have to
declare these methods in the proxy.
Bug: webrtc:12238
Change-Id: I70c56336641c2a108b68446ae41f43409277a584
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217762
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33964}
This reverts commit 6e4fcac31312f2dda5b60d33874ff0cd62f94321.
Reason for revert: Parent CL issue has been resolved.
Original change's description:
> Revert "Remove thread hops from events provided by JsepTransportController."
>
> This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76.
>
> Reason for revert: Parent CL breaks FYI bots.
> See https://webrtc-review.googlesource.com/c/src/+/206466
>
> Original change's description:
> > Remove thread hops from events provided by JsepTransportController.
> >
> > Events associated with Subscribe* methods in JTC had trampolines that
> > would use an async invoker to fire the events on the signaling thread.
> > This was being done for the purposes of PeerConnection but the concept
> > of a signaling thread is otherwise not applicable to JTC and use of
> > JTC from PC is inconsistent across threads (as has been flagged in
> > webrtc:9987).
> >
> > This change makes all CallbackList members only accessible from the
> > network thread and moves the signaling thread related work over to
> > PeerConnection, which makes hops there more visible as well as making
> > that class easier to refactor for thread efficiency.
> >
> > This CL removes the AsyncInvoker from JTC (webrtc:12339)
> >
> > The signaling_thread_ variable is also removed from JTC and more thread
> > checks added to catch errors.
> >
> > Bug: webrtc:12427, webrtc:11988, webrtc:12339
> > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33195}
>
> TBR=nisse@webrtc.org,tommi@webrtc.org
>
> Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12427
> Bug: webrtc:11988
> Bug: webrtc:12339
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33203}
TBR=nisse@webrtc.org,tommi@webrtc.org,guidou@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12427
Bug: webrtc:11988
Bug: webrtc:12339
Change-Id: I4e2e1490e1f9a87ed6ac4d722fd3c442e3059ae0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206809
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33225}
This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76.
Reason for revert: Parent CL breaks FYI bots.
See https://webrtc-review.googlesource.com/c/src/+/206466
Original change's description:
> Remove thread hops from events provided by JsepTransportController.
>
> Events associated with Subscribe* methods in JTC had trampolines that
> would use an async invoker to fire the events on the signaling thread.
> This was being done for the purposes of PeerConnection but the concept
> of a signaling thread is otherwise not applicable to JTC and use of
> JTC from PC is inconsistent across threads (as has been flagged in
> webrtc:9987).
>
> This change makes all CallbackList members only accessible from the
> network thread and moves the signaling thread related work over to
> PeerConnection, which makes hops there more visible as well as making
> that class easier to refactor for thread efficiency.
>
> This CL removes the AsyncInvoker from JTC (webrtc:12339)
>
> The signaling_thread_ variable is also removed from JTC and more thread
> checks added to catch errors.
>
> Bug: webrtc:12427, webrtc:11988, webrtc:12339
> Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33195}
TBR=nisse@webrtc.org,tommi@webrtc.org
Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12427
Bug: webrtc:11988
Bug: webrtc:12339
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33203}
Events associated with Subscribe* methods in JTC had trampolines that
would use an async invoker to fire the events on the signaling thread.
This was being done for the purposes of PeerConnection but the concept
of a signaling thread is otherwise not applicable to JTC and use of
JTC from PC is inconsistent across threads (as has been flagged in
webrtc:9987).
This change makes all CallbackList members only accessible from the
network thread and moves the signaling thread related work over to
PeerConnection, which makes hops there more visible as well as making
that class easier to refactor for thread efficiency.
This CL removes the AsyncInvoker from JTC (webrtc:12339)
The signaling_thread_ variable is also removed from JTC and more thread
checks added to catch errors.
Bug: webrtc:12427, webrtc:11988, webrtc:12339
Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33195}
- Replace few sigslot usages in jsep_transport_controller.
- There is still one sigslot usages in this file so keeping the inheritance
and that is the reason for not having a binary size gain in this CL.
- Remaining sigslot will be removed in a separate CL.
Bug: webrtc:11943
Change-Id: Idb8fa1090b037c48eeb62f54cffd3c485cebfcda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190146
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33034}
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 is a reland of 40261c3663fe316cfe40262c59cee993165ccf63
Note: Instead of changing the type of JsepTransportController->SignalSSLHandshakeError
added a new member with a different name and used it in webrtc code.
After this change do two more follow up CLs to completely remove the old code
from google3.
Original change's description:
> Replace sigslot usages with robocaller library.
>
> - Replace all the top level signals from jsep_transport_controller.
> - There are still sigslot usages in this file so keep the inheritance
> and that is the reason for not having a binary size gain in this CL.
>
> Bug: webrtc:11943
> Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
> Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32321}
Bug: webrtc:11943
Change-Id: Ia07394ee395f94836f6b576c3a97d119a7678e1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186946
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32359}
This reverts commit 40261c3663fe316cfe40262c59cee993165ccf63.
Reason for revert: Breaks downstream project
Original change's description:
> Replace sigslot usages with robocaller library.
>
> - Replace all the top level signals from jsep_transport_controller.
> - There are still sigslot usages in this file so keep the inheritance
> and that is the reason for not having a binary size gain in this CL.
>
> Bug: webrtc:11943
> Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
> Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32321}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,glahiru@webrtc.org
Change-Id: Icf438f87c3d95940d858db3cc5848b23abb82fc4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186844
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32324}
- Replace all the top level signals from jsep_transport_controller.
- There are still sigslot usages in this file so keep the inheritance
and that is the reason for not having a binary size gain in this CL.
Bug: webrtc:11943
Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32321}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/174261
Patchset 1 contains the old cl (plus a merge conflict fix).
Later patchets are bufixes: A PeerConnection can be created without a
Call instance (in the case of DataChannel only), so we can't always
use that to fetch the current trials.
Old CL descritpion:
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
// Since re-land is otherwise unchanged, setting previous reviewers as TBR
TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Bug: webrtc:11926
Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32163}
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
Bug: webrtc:11926
Change-Id: I70f28c4fbabf6d9e55052342000e38612b46682c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174261
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32129}
This fixes some edge cases where early media could cause default
stream that block the actual signaled media from beind delivered.
Bug: webrtc:11477
Change-Id: I8b26df63a690861bd19f083102d1395e882f8733
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183120
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32030}
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}
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}
