Currently, sharing a screen or a window on Wayland opens unnecessary
preview dialog on Chromium side, which is then followed by a similar
dialog on xdg-desktop-portal side. The Chromium dialog is useless on
Wayland, as it doesn't show anything. This is because Chromium doesn't
have access to screen content as in case of X11 session. To fix this, we
want to avoid showing the preview dialog in case we find that we run on
Wayland and only pick a screen or a window from the dialog that comes
from xdg-desktop-portal.
This patch splits BaseCapturerPipeWire class, moving portal related code
into XdgPortalBase, which does all the DBus communication and which is
supposed to be reused by BaseCapturerPipeWire when the user confirms
the dialog from xdg-desktop-portal. The XdgPortalBase is extended to
support multiple calls at once, where each call is identified by Id.
Relevant change on Chromium side will be in a different review.
Bug: chromium:682122
Change-Id: If8afd36da66231eb154cdc00114908ac897ee4cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160649
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32342}
Original description
Move reporting of target bitrate to just after the encoder has been
updated. Originall submitted as refs/heads/master@{#32275}
Patch 1 contains the original cl
,patch 2 the fix to send rtcp even if BWE does not change.
Bug: webrtc:12000
Change-Id: I16766e08229fe1f6f65f449e0e074bed03338693
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186948
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32340}
This reverts commit 39a31afb77e3ce5c4ff53b8bab06364712cae7ce.
Reason for revert: Will cause RTCP Target bitrate messages to not be sent unless BWE changes.
Original change's description:
> Refactor reporting of VideoBitrateAllocation
>
> Move reporting of target bitrate to just after the encoder has been
> updated.
>
> Bug: webrtc:12000
> Change-Id: I3e7c5bd44c2f64e5f7e32d6451861b80e0b779ca
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186041
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32275}
TBR=sprang@webrtc.org,perkj@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12000
Change-Id: Icf21e6ae28dc17c61b9243c037ffef9b623894ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186945
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32337}
This reverts commit f5e261aaf65cdf2eb903cdf40d651846be44f447.
Reason for revert: Breaks downstream projects.
Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11495
Change-Id: Ice318d1b11ca3dff09c190187a0b0a32ca945fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32335}
The previous tests ran in real-time making them flaky, so they were
disabled on a number of platforms.
This CL ports the tests 1:1 (sort of) to use the scenario test
framework which runs with simulated time and much less risk of
flakiness.
Bug: webrtc:10155
Change-Id: I6281f57d73883c8aaa91964e9cfa58d9b47779fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186941
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32333}
This is a reland of f8e62fcb14e37a5be4f1e4f599d34c8483fea8e9
Original change's description:
> Refactor NetEq delay manager logic.
>
> - Removes dependence on sequence number for calculating target delay.
> - Changes target delay unit to milliseconds instead of number of
> packets.
> - Moves acceleration/preemptive expand thresholds to decision logic.
> Tests for this will be added in a follow up cl.
>
> Bug: webrtc:10333
> Change-Id: If690aae4abf41ef1d9353f0ff01fb7d121cf8a26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186265
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32326}
Bug: webrtc:10333
Change-Id: Iad5e7063f63b84762959ee5b412f5f14a7b2cd06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186943
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32332}
During muted state NetEq shortcircuits a large part of the internals to
quickly return a buffer filled with zeros. It can be beneficial for the
controller to be aware that it is in muted state.
Bug: webrtc:11005
Change-Id: I5fe24b4a3704d953cbd68b5a24bbb7ef58b30be0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186760
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32330}
This reverts commit f8e62fcb14e37a5be4f1e4f599d34c8483fea8e9.
Reason for revert: breaks downstream test.
Original change's description:
> Refactor NetEq delay manager logic.
>
> - Removes dependence on sequence number for calculating target delay.
> - Changes target delay unit to milliseconds instead of number of
> packets.
> - Moves acceleration/preemptive expand thresholds to decision logic.
> Tests for this will be added in a follow up cl.
>
> Bug: webrtc:10333
> Change-Id: If690aae4abf41ef1d9353f0ff01fb7d121cf8a26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186265
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32326}
TBR=ivoc@webrtc.org,jakobi@webrtc.org
Change-Id: I1bdeacce61b902a0003a40c740f6acccf1443e3e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186942
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32329}
- Removes dependence on sequence number for calculating target delay.
- Changes target delay unit to milliseconds instead of number of
packets.
- Moves acceleration/preemptive expand thresholds to decision logic.
Tests for this will be added in a follow up cl.
Bug: webrtc:10333
Change-Id: If690aae4abf41ef1d9353f0ff01fb7d121cf8a26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186265
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32326}
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}
This CL adds delay headroom when an external delay estimator is used.
Tested: audioproc_f --aec=1 is bitexact on a large number of aecdumps
Bug: b/158455753
Change-Id: I56de44e841bb8162e302181c6c386ad7fbb00dee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186703
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32323}
- 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}
The extension is suggested to be used for signaling per target bitrate, resolution
and frame rate to a SFU to allow a SFU to know what video layers a client is currently targeting.
It is hoped to replace the current Target bitrate RTCP XR message currently used only for screen share.
Bug: webrtc:12000
Change-Id: Id7b55e7ddaf6304e31839fd0482b096e1dbe8925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185980
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32313}
This prevents having to have sdp_offer_answer depend on peer_connection
for the messaging functions.
Bug: webrtc:11995
Change-Id: Icad7c9c0e6149bd1b8d78e37eff5f9786b74692e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186662
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32310}
The PseudoTcp test class is being used outside of WebRTC in ways
that WebRTC itself doesn't, which caused this revert:
https://webrtc-review.googlesource.com/c/src/+/186564
As it happens though, PseudoTcp doesn't actually use the
StreamInterface part of FifoBuffer, so this CL cuts the dependency
from PseudoTcp on FifoBuffer.
Moving forward, we could just remove this class from WebRTC.
Bug: webrtc:11988
Change-Id: Id34a2a6305e8fe37d705ba5e8876dd6398515125
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186665
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32309}
Currently, when only max bitrate available and min bitratea & target
bitrate are missing from encoding config, the target bitrate is decided
by the calculation from GetSimulcastConfig() according to width/height/qp.
The max bitrate doesn't play a role here other than ensure target < max.
This will make the target bitrate cap at some calculated number even
when control message gives much larger allocation through max bitrate.
In our cases, the L0 (at 180p) is capped at 80-90kbps even control
message gives L0's max bitrate over 300kbps. This under-use of bandwidth
happens to all layer other than top layer. Top layer will be compensated
with all the left bandwidth up to max at last.
Since in web api, we cannot pass down either min bitrate or target bitrate
(https://developer.mozilla.org/en-US/docs/Web/API/RTCRtpEncodingParameters).
We propose a new logic to take max bitrate into consideration in this case,
use 3/4 max bitrate or calculated target bitrate whichever is larger.
Bug: None
Change-Id: I2234b4636daa379fd47d4bbe764cf8307b9a1ea4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186161
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32308}
This needs to be done still for kNative frames, but all other frame types
can be passed in.
I have checked all VideoEncoder implementations in Chromium and confirmed they either convert the frame to their preferred pixel format, or just
forward the frame to a delegate encoder.
Tested:
- video_loopback with NV12 generated frames for VP9, the only
codec supporting NV12, as well as VP8 which only accepts I420 frames.
- internal_tests tryrun
Bug: webrtc:11976,webrtc:11635
Change-Id: If39a815fb0c5636fceb1040c8946c3db2fb350a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185803
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32306}
This is a reland of 2978abb88c49362e296bdce3cb662f6255b17083
Original change's description:
> Reduce the amount of howling reduction in AEC3
>
> This CL backs off the howling protection functionality in AEC3.
> The effect is increased transparency in some cases. No negative effects
> have been identified in the hands-on testing.
>
>
> A kill-switch is added that can be used to turn off the functionality.
>
> Bug: b/150764764
> Change-Id: I604c569c76f911799556a60bc8fd2fb43bbfe196
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186082
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32258}
Bug: b/150764764,chromium:1134939
Change-Id: I5eea60b35e6d09003ec2fee3865513df8bdd5823
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186260
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32304}
This is a reland of ad148272b89394978915cb00e1c1be552d908a42
Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}
Bug: webrtc:11663, chromium:1134234
Change-Id: I0cb34cf08d4d14bc3aee055254493c9c9ee8faa0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186401
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32303}
After this CL, sdp_offer_answer is bigger than peer_connection.
Bug: webrtc:11995
Change-Id: Ie923fabf836de46fa939fe6fd7b3d936bbc85dab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186380
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32301}
This adds a config parameter for reverb modelling in nonlinear mode.
The killswitch is preserved, and will have the same effect before and after this CL.
Tested: Bitexact on a large number of aecdumps both with and without WebRTC-Aec3NonlinearModeReverbKillSwitch.
Bug: webrtc:8671
Change-Id: I7b2fa92307af5316dbe895392367abb8d15f06bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183541
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32298}
The trial name WebRTC-VP9-PerLayerSpeed is used to
a) set encoding speed per spatial layer, based on resolution
b) allow explicitly overriding speed per layer, for testing
Additionally, this CL updates the vp9 wrapper in preparation for
injectable trials.
Bug: webrtc:11551, webrtc:11926
Change-Id: I2bb3a664feaef60483ffc241b71070284d3e0172
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186400
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32294}
