That brings usage of that queue closer to the production.
In particular that should surface race conditions on destruction.
Those should be fixed rather than avoided.
Bug: webrtc:10933
Change-Id: Iff60cf5a4b87bd848117ef543ffc97f6504dc979
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157898
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29665}
This is a reland of 87a7b82520b83a6cf42da27cdc46142c2eb6248c
Original change's description:
> Refactoring of the noise suppressor and adding true multichannel support
>
> This CL adds proper multichannel support to the noise suppressor.
> To accomplish that in a safe way, a full refactoring of the noise
> suppressor code has been done.
>
> Due to floating point precision, the changes made are not entirely
> bitexact. They are, however, very close to being bitexact.
>
> As a safety measure, the former noise suppressor code is preserved
> and a kill-switch is added to allow revering to the legacy noise
> suppressor in case issues arise.
>
> Bug: webrtc:10895, b/143344262
> Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29646}
Bug: webrtc:10895, b/143344262
Change-Id: I236f1e67bb0baa4e30908a4cf7a8a7bb55fbced3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158747
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29663}
The encoder can drop all frames for extended periods if it has produced over budget. After 2 seconds without any encoded frames, the video send stream times out and deallocates the stream.
Ideally the send stream should keep track if frames are captured instead of encoded, but keeping the stream alive using OnDroppedFrame can work as a proxy for that.
Bug: webrtc:11062
Change-Id: Id7ec1ff333427643453c4a36d1db03ca826cd9ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158700
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29662}
This is to allow advanced features of WebRTC/Chrome e.g., field trials.
More style compliant changes may follow up. Only a minimal (not in terms of line changes) is applied, so that presubmit does not complain. These changes include
1. removing unused headers.
2. eliminating c-style casting.
Bug: b/143582588
Change-Id: I6d0fd926c542ab0afdc38cc4bf03aaf584ec13dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158670
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29657}
This is a reland of e7153012682ccd3d1eacc18f802cab7820e3bad3
Original change's description:
> Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.
>
> Bug: chromium:396091
> Change-Id: Ia1b36c771632c536bb8d15322461b479fabc409e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148768
> Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
> Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#29083}
Bug: chromium:396091
Change-Id: I0d9171ae5f340e0489e4b45ce5d97bc52b0a4904
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156067
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29655}
For the automatic content type detection we need to know if the update
rect is trusted or just not available.
Currently we only care if it's not empty, so in case of no update rect
available, full frame resolution was set as a changed region.
This CL makes the update_rect field optional but should be a no-op in the
current code, as absence of update_rect is treated as a full update via
a new getter method |update_rect_or_full_frame()|.
Bug: webrtc:11058
Change-Id: I913545b71ac2fc845861549ac34eb1b630012109
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158673
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29654}
This reverts commit 87a7b82520b83a6cf42da27cdc46142c2eb6248c.
Reason for revert: Speculative revert. Breaks downstream projects.
Original change's description:
> Refactoring of the noise suppressor and adding true multichannel support
>
> This CL adds proper multichannel support to the noise suppressor.
> To accomplish that in a safe way, a full refactoring of the noise
> suppressor code has been done.
>
> Due to floating point precision, the changes made are not entirely
> bitexact. They are, however, very close to being bitexact.
>
> As a safety measure, the former noise suppressor code is preserved
> and a kill-switch is added to allow revering to the legacy noise
> suppressor in case issues arise.
>
> Bug: webrtc:10895, b/143344262
> Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29646}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: I4d4025bda01f484979961fe57380a705e4d78397
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10895, b/143344262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158701
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29651}
Removes all unused features, reducing the exposed interface surface.
This makes refactoring and maintenance simpler as we can change
TransportFeedbackAdapter without making corresponding changes
to RtpVideoSender.
Bug: webrtc:9883
Change-Id: If372a868e0765e94df52b4de52d3bb619ce11471
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156943
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29649}
This CL adds proper multichannel support to the noise suppressor.
To accomplish that in a safe way, a full refactoring of the noise
suppressor code has been done.
Due to floating point precision, the changes made are not entirely
bitexact. They are, however, very close to being bitexact.
As a safety measure, the former noise suppressor code is preserved
and a kill-switch is added to allow revering to the legacy noise
suppressor in case issues arise.
Bug: webrtc:10895, b/143344262
Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29646}
If a stun server is already there, the benefit of adding turn servers as stun servers is small,
and it may create unnecessary stun candidates.
Bug: webrtc:11059
Change-Id: Ia37b43b787180af4d91c1c07c866ccbf1db80262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158680
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29644}
Due to changes in how the transport feedback is processed, the late
feedback results plot doesn't get any entries anymore.
Bug: webrtc:9883
Change-Id: I9df8e86a35bedddf78407128f0ab0b6b321a6f28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158668
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29643}
This change removes the deprecated ChannelLayout versions of ProcessStream
and AnalyzeReverseStream.
Bug: webrtc:5298
Change-Id: I8a7e33e89cffac5eceecd00dfd3c96000643f51b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158529
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29641}
There is still an TSAN issues, but they are inside
usrsctp library, that's why tests are still disabled
by default.
Bug: None
Change-Id: I55f7c66b4d9a5feccd2121e2dd3b131cf1564804
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158522
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29640}
The code path it calls is no longer actually used and will be cleaned
up in a follow-up CL.
This prepares for simplifying the transport feedback adapter and moving
it to run on a task queue.
Bug: webrtc:9883
Change-Id: I750398069414ffa782067d021c0a3837049d98eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158621
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29638}
The extra call doesn't appear to have any side effects, but should be
removed nonetheless.
Bug: None
Change-Id: I45a98261828211b7bc7954acb779913249dc2b01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158620
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29637}
This field-trial allows us to provide multipliers for the opus target
bitrate.
Bug: webrtc:11055
Change-Id: I79c4c6389c6908daadda355e5ce0668413d0aaa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158530
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29634}
This is a reland of 1dddaa1a84330091ca083c950ef2e24a85a48fc8
The regression that caused the original CL to be reverted was the fact that
invoking SetLocalDescription() inside of the CreateOffer() callback was no
longer executing synchronously and immediately.
In this CL, the original CL is patched so that the CreateOffer() operation
is marked as completed just before invoking the CreateOffer() callback
(versus doing it just afterwards). This ensures that the OperationsChain is
popped before the callback runs. The same applies for CreateAnswer().
See diff between Patch Set 1 (Original CL) and the latest Patch Set.
Original change's description:
> [PeerConnection] Use an OperationsChain in PeerConnection for async ops.
>
> For background, motivation, requirements and implementation notes, see
> https://docs.google.com/document/d/1XLwNN2kUIGGTwz9LQ0NwJNkcybi9oKnynUEZB1jGA14/edit?usp=sharing
>
> Using the OperationsChain will unblock future CLs from chaining multiple
> operations together such as implementing parameterless
> setLocalDescription().
>
> In this CL, the OperationsChain is used in existing signaling operations
> with little intended side-effects. An operation that is chained onto an
> empty OperationsChain will for instance execute immediately, and
> SetLocalDescription() and SetRemoteDescription() are implemented as
> "synchronous operations".
>
> The lifetime of the PeerConnection is not indended to change as a result
> of this CL: All chained operations use a WeakPtr to the PC to ensure
> use-after-free does not happen.
>
> There is one notable change though: CreateOffer() and CreateAnswer() will
> asynchronously delay other signaling methods from executing until they
> have completed.
>
> Drive-by fix: This CL also ensures that early failing
> CreateOffer/CreateAnswer operation's observers are invoked if the
> PeerConnection is destroyed while a PostCreateSessionDescriptionFailure
> is pending.
>
> Bug: webrtc:11019
> Change-Id: I521333e41d20d9bbfb1e721609f2c9db2a5f93a9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157305
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29605}
TBR=steveanton@webrtc.org
Bug: webrtc:11019
Change-Id: I57b4496e63378c91c24679ee496e21f5cb6a0e59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158524
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29630}
The feature is added as part a new field trial WebRTC-Bwe-IgnoreProbesLowerThanNetworkStateEstimate
Bug: webrtc:10498
Change-Id: I72b3c73256a35e0583f4d595edef45848f8bbb22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158260
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29624}
The existing ICE stack will choose *the first* connection
that becomes writable.
It is possible that waiting a fixed time will choose a
better connection, avoiding a switch, and making the experience better
in total.
This patch is add two field trials to *explore*
that dimension. I.e the code will be rolled back once
experiments has been performed.
- initial_select_dampening, delays selection by X ms.
- initial_select_dampening_ping_received, delays selection for
candidate that has received ping by X ms.
BUG=webrtc:11054
Change-Id: Ifcdde5183f318815e0f5db5802fbf6b542a95f5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158410
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29623}
This reverts commit 29db239fed94bbe40dfddefa9f02409b951efc18.
Reason for revert: The test is on its way on https://chromium-review.googlesource.com/c/chromium/src/+/1874722.
Original change's description:
> Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
>
> This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed
>
> Original change's description:
> > Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
> >
> > In order to land the component build support in Chromium, it is
> > easier to turn on symbols export every time that is_component_build=true
> > instead of setting rtc_enable_symbol_export=is_component_build in
> > Chromium (since is_component_build is not available in .gn).
> >
> > rtc_enable_symbol_export is still kept in the mix in order to turn
> > on symbol exports in any case a shared library will be added to the
> > WebRTC build.
> >
> > Bug: webrtc:9419
> > Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29540}
>
> TBR: kwiberg@webrtc.org
> Bug: webrtc:9419
> Change-Id: I6a184290f3f1a60c1ed6cbc86aafe3e7df248ce1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158327
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29618}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: Ie3bac3393d6ce0f147f3d79d08d20cc9798b464a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158328
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29619}
This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed
Original change's description:
> Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
>
> In order to land the component build support in Chromium, it is
> easier to turn on symbols export every time that is_component_build=true
> instead of setting rtc_enable_symbol_export=is_component_build in
> Chromium (since is_component_build is not available in .gn).
>
> rtc_enable_symbol_export is still kept in the mix in order to turn
> on symbol exports in any case a shared library will be added to the
> WebRTC build.
>
> Bug: webrtc:9419
> Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29540}
TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: I6a184290f3f1a60c1ed6cbc86aafe3e7df248ce1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158327
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29618}
The post-pacing part of the RTP sender has been moved from RTPSender
into the new RtpSenderEgress class. However, that class is not directly
used and instead a subset of method calls are passed through RTPSender.
This CL prepares for removing dependencies between RTPSender and
RtpSenderEgress. All current behavior is preserved, and unit tests are
unchanged to verify this.
For more context, see patch set 2.
Change-Id: If795f2603aeb6302ac1565d9efaea514af240dc7
Bug: webrtc:11036
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158020
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29616}
