While Artem is OOO there is no owner over PC framework changes that can appove CLs.
Bug: webrtc:12247
Change-Id: I70aa5e1263efa9c0971a077ecbb247a7c41991cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196091
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32743}
This enables send side bandwidth estimation for audio and removes field
trial "WebRTC-Audio-SendSideBwe" which this was controlled through.
Transport-cc extension still needs to be negotiated.
Bug: webrtc:12222
Change-Id: Ie2268fad13703eeb0f0d38fcf484baaa29715b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194142
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32681}
This can be used to test ICE behavior.
Bug: chromium:1024965
Change-Id: Ie4ba9cd5c3cf3c2f71bab3637f925263dbc6296e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193701
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32625}
Calculate quality metrics for dropped frames by comparing original
frame against last decoded one.
This feature makes comparison of encoders which do/don't drop frames
more fair.
The feature is controlled by analyze_quality_of_dropped_frames flag
and is disabled by default.
Bug: none
Change-Id: Ifab8df92d0b76e743ff3193c05d7c8dbd14921c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190660
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32518}
In the old jitter buffer the two VCMVideoProtection modes |kProtectionNone| and |kProtectionFEC| could be set on the jitter buffer for it to not wait for NACK and instead generate incomplete frames. This has not been possible for a long time.
Bug: webrtc:9378, webrtc:7408
Change-Id: I0a2d3ec34d721126c1128306d5fad88314f8d59f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32513}
This changes the default behavior to use pacing factor of 1.1x instead
of 2.5x, it also sets libvpx rate controler as trusted, turns on the
encoder pushback mechanism and sets spatial hysteresis to 1.2.
The unused "dynamic rate" settings in libvpx is removed.
The new settings matches what has been used in chromium since 2019.
If needed, the legacy behavior can be enabled using the field trial
WebRTC-VideoRateControl.
Bug: webrtc:10155
Change-Id: I8186b491aa5bef61e8f568e96c980ca68f0c208f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186661
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32477}
This is expected to yield slightly higher bandwidth estimates when
probing is used, since it reduces a bias in how packet sizes are counted.
Bug: webrtc:11780
Change-Id: I6a4a3af0c50670d248dbe043a4d9da60915e3699
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187491
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32394}
(Reland with no changes after the fix to the downstream project)
This can be overriden for kNative frame types to perform scaling efficiently.
Default implementations for existing buffer types require actual
buffer implementation, thus this CL also merges "video_frame"
with "video_frame_I420" build targets.
Originally Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303
(Landing with TBR as it's unchaged reland of already approved CL)
TBR=nisse@webrtc.org,sakal@webrtc.org
Bug: webrtc:11976, chromium:1132299
Change-Id: Ia23f7d3e474bd9cdc177104cc5c6d772f04b210f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187345
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32362}
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}
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 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}
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 change lets the fuzzer modify the first few bytes of the RTP
payload. One of the benefits is that it can cover the RED header
splitter functionality.
The CL also fixes an issue found while running the fuzzer locally.
Bug: webrtc:11640
Change-Id: I7ca73676440897a14a0aaca796f70d381e016575
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185819
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32242}
Currently isolated output directory is created in flags_compatibility.py script.
This doesn't work for android swarming tasks because this script isn't called.
Bug: webrtc:11895
Change-Id: I8b8f01850d6e5970292b524d104314eef7ab17be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185883
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32236}
This reduces the degree of interdependency among modules related
to the PeerConnection class, and makes it easier to isolate inappropriate
external dependencies.
Bug: webrtc:11967
Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32235}
This can be used in the future to test NV12 video frames with encoders, both
from unittests and from tools like video_loopback.
Tested using video_loopback with generator NV12.
Bug: webrtc:11978
Change-Id: I0d24ae3ebab2267f076703cbda81e99cec465ec8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185045
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32206}
I ran into this when using repeating_task, which depends on clock (in
system_wrappers) which in turn added a dependency on rtc_base on Windows
due to win32 files. That's a problem since rtc_base depends on
repeating_task:
//rtc_base:rtc_base ->
//rtc_base/task_utils:repeating_task ->
//system_wrappers:system_wrappers ->
//rtc_base:rtc_base
We could additionally consider moving Clock out of system_wrappers.
Bug: webrtc:9987
Change-Id: I54ed715ad5eb9e3f5dd6c322233c18c05d895dff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185506
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32203}
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}
Make timestamps on the charts for metrics reported from
DefaultVideoQualityAnalyzer more precise.
Bug: webrtc:11959
Change-Id: I805fdac0d499b7326d6bc2240154c1c31ca81a62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184602
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32149}
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}
Add a get_and_clear_legacy_stats flag to AudioReceiveStream::GetStats,
to distinguish calls from standard GetStats and legacy GetStats.
Add const method NetEq::CurrentNetworkStatistics to get current
values of stateless NetEq stats. Standard GetStats will then call this
method instead of NetEq::NetworkStatistics.
Bug: webrtc:11622
Change-Id: I3833a246a9e39b18c99657a738da22c6e2bd5f5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32092}
Currently is_linux is set to true on Chrome OS build,
but it is planned to be set false. This CL is the preparation
to keep the compatibility.
Bug: chromium:1110266
Test: Build locally.
Change-Id: Ic79a202b0b3baeff157955cd03a07556bfb958a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Hidehiko Abe <hidehiko@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32073}
It is required to properly support real and simulated time.
Bug: webrtc:11743
Change-Id: If6dd59691d966378f8ff897c82dee05c1899e9e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183602
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32057}
