This is to simplify implementation of new feature flags.
- Move helper functions to anonymous namespace.
- Add members to avoid passing everything by function paramaters.
Bug: webrtc:14508
Change-Id: I0a4958645a4eb76515f28d8ce868a66be6748919
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277720
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38274}
Early return will cause `call` to be destroyed outside the worker thread, which gives confusing error messages when all you did was type the wrong path to the input file :)
Bug: webrtc:14508
Change-Id: I029910d8da4bc7b08dafd02cb5ebf88d9c7afa59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277443
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38254}
This cl/ implements configuring of encode resolution
in the video_stream_encoder (webrtc_video_engine) in
a way that is independent of frame resolution (i.e
not using scale_resolution_down_by).
The cl/ reuses the VideoAdapter as is, and hence
the output resolution will be the same as it is today.
Anticipated further patches
3) Hook up resource adaptation
4) Let VideoSource do adaption if possible
Bug: webrtc:14451
Change-Id: I881b031c5b23be26cacfe138730154f1cb1b66a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276742
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38245}
This cl move VideoEncoderConfig from api/ to video/config.
VideoStreamEncoderInterface and VideoStreamEncoderObserver
are moved as collateral.
brandt@ think that the reason these were in api/ in the
first place had to downstream project.
Functionality wise, this is a NOP, but it makes it easier
to modify the encoder (config).
Bug: webrtc:14451
Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38242}
This cl/ is a NOP refactoring,
moving the EncoderStreamFactory from within webrtc_video_engine.cc
into own file in video/. simulcast.cc is collateral.
Bug: webrtc:14451
Change-Id: Ia69b9241d8cd8a12be6628d887701f2e244c07cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276861
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38224}
Making it clear that the field is used to store the applied input
volume and not the recommended input volume.
Bug: webrtc:7494, b/241923537
Change-Id: Ib91bc1a12348f63e3a4ba6e068ed02e40786a87b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271342
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38051}
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
This reverts commit 83db78e854.
Reason for revert: Some tests in Chromium's blink no longer compile because of the change in the signature of the CreateDefaultTaskQueueFactory() function.
Original change's description:
> Add TaskQueueStdlib experiment.
>
> Bug: webrtc:14389
> Change-Id: I23c6e0ae675748ec35a99c334104dd2654995a33
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265802
> Commit-Queue: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37888}
Bug: webrtc:14389
Change-Id: If3e63d6b4ab9e838dc5020b88076a73fd29916e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272920
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37902}
In migrating rtc::Event to use TimeDelta instead of int,
rtc::Event::kForever will have to become something else.
This change removes dependencies on that kForever is int.
Bug: webrtc:14366
Change-Id: Ic36057dda95513349e7ae60204e7271ff1f58825
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271288
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37795}
Starting from Go 1.17, modules are the default and cannot be disabled.
This requires a change in the GitHub repositury [1].
As a stopgap solution, this CL moves Go back to 1.16 and disables
modules [2].
[1] - https://github.com/webrtc/apprtc/tree/master/src/collider
[2] - https://go.dev/blog/go116-module-changes
No-Presubmit: True
Bug: webrtc:14342
Change-Id: Idd03639588bc03497a78f0cef350daebf3b2f1d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271481
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37760}
This allow to remove the testonly from the rtc_event_log_visualizer
binary and the implicit dependency on the path of the default
conversational speech file.
The binary size of event_log_visualizer passes from 2.1 MB to 4.0 MB.
Bug: b/237526033
Change-Id: I71cf647f039f26f30c792c49c752cff5c5b329a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267663
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37453}
This CL duplicates a few lines of utility code from
//modules/audio_processing:audioproc_test_utils (which contains more
testonly things) and allows the possibility to remove testonly from
the unpack_aecdump tool.
Bug: b/237526033
Change-Id: If2e1dd4cc825429c496091cf8640c67069fb6e6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267701
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37437}
Instead of creating a TaskQueue from packet_sender, create a rtc::Thread
in test_controller so that test_controller instantiates a SocketServer,
eliminating the use of rtc::Thread::socketserver().
Also did various cleanups, such as adding threading annotations, and
ensuring that all network operations are done in dedicated threads.
Bug: webrtc:13145
Test: Unittest, and manually verified using Android clients and Linux servers
Change-Id: I05ebe5e29bd80f14a193c9ee8b0bf63a1b6b94d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263321
Commit-Queue: Daniel.l Lee <daniel.l@hpcnt.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37411}
which is useful for validating flexfec recovers frames correctly.
This can be tested by passing a keyframe covered by flexfec along
with the fec packets and removing one packet from the frame.
BUG=None
Change-Id: Icd73eca138f62b9387bf850a6efbd7db03b4b569
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261956
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37392}
which is already supported in modules/video_coding/utility/ivf_file_reader.cc
BUG=None
Change-Id: I7b00659dc460d372312dff2eb53837a321ab16af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262246
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37390}
and fix a typo
BUG=None
Change-Id: Ie286c52cbdfbc0269d92503c46b805bfdf5bb556
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264151
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37042}
To make it usable in tests without depending on all of CallTest.
Bug: None
Change-Id: Ie3102ab71bcfe3862dd6c35d3285098e961e54df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262807
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36932}
- Filter out very old packets (to ensure that the estimate doesn't drop to zero if sending is paused and later resumed).
- Discard packets older than previously discarded packets (to avoid the estimate dropping after deep reordering.)
- Add tests cases for high loss, deep reordering and paused/resumed streams to unittest.
- Remove some field trial settings that have very minor effect and rename some of the others.
- Change analyzer.cc to only draw data points if the estimators have valid estimates.
Bug: webrtc:13402
Change-Id: I47ead8aa4454cced5134d10895ca061d2c3e32f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236347
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36849}
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.
Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
Create a server using:
./data_channel_benchmark --server --port 12345
Start the flow of data from the server to a client using:
./data_channel_benchmark --port 12345 --transfer_size 100
The throughput is reported on the server console.
The negotiation does not require a 3rd party server and is done over a
gRPC transport. No TURN server is configured, so both peers need to be
reachable using STUN only.
Bug: webrtc:13288
Change-Id: Iac9a96cf390ab465ea45a46bf0b40950c56dfceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235661
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36206}
This CL also removes dependency on the legacy field trial methods.
Bug: webrtc:11926
Change-Id: I53feeee86b92878cf0f2b8ebdce3d101f9e04014
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255381
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36205}
This is a reland of 43fb16921b
Original change's description:
> Refactor AnalyzerConfig to use Timestamps instead of microseconds.
>
> Add optional offset-to-UTC parameter to output. This allows aligning
> the x-axis in the generated charts to other UTC-based logs.
>
> Bug: b/215140373
> Change-Id: I65bcd295718acbb8c94e363907c1abc458067bfd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250203
> Reviewed-by: Kristoffer Erlandsson <kerl@google.com>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35992}
Bug: b/215140373
Change-Id: Id2b88cc4b8078a97275d49a617581cbbd02d2c6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252380
Reviewed-by: Kristoffer Erlandsson <kerl@google.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36066}
Add optional offset-to-UTC parameter to output. This allows aligning
the x-axis in the generated charts to other UTC-based logs.
Bug: b/215140373
Change-Id: I65bcd295718acbb8c94e363907c1abc458067bfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250203
Reviewed-by: Kristoffer Erlandsson <kerl@google.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35992}
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 is part of a set of CLs to fix the Chromium roll.
Bug: None
Change-Id: I7cc1e5b84a2443f311fb4e047954d7d4d9c41189
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236761
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35284}
In callers where it's non-trivial to explicitly pass the right
SocketFactory, pull the call to rtc::Thread::socketserver() into the
caller, with a TODO comment.
Bug: webrtc:13145
Change-Id: I029d3adca385d822180e089f016c3778e0d4fd0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231227
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35063}
CL partially auto-generated with:
git grep -l "\bassert(" | grep "\.[c|h]" | \
xargs sed -i 's/\bassert(/RTC_DCHECK(/g'
And with:
git grep -l "RTC_DCHECK(false)" | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'
With some manual changes to include "rtc_base/checks.h" where
needed.
A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.
The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.
This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).
Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
This disables fast accelerate mode but max buffer size is the same.
Bug: None
Change-Id: Iba883051c42b28ab094075948a43ec288b77ad5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224545
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34403}
This is a reland of e9ae4729e0TBR=philipel@webrtc.org,terelius@webrtc.org
Original change's description:
> Correctly handle retransmissions/padding in early loss detection.
>
> This CL makes sure we don't cull packets from the history based on
> incorrect ack mapping, just like it's predecessor:
> https://webrtc-review.googlesource.com/c/src/+/218000
>
> It also changes the logic to make sure retransmits counts towards
> history pruning - and properly ignores padding/fec.
>
> Bug: webrtc:12713
> Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34293}
Bug: webrtc:12713
Change-Id: Iec123d71edafea98fe289acde007b57e212681f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222640
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34297}
This reverts commit e9ae4729e0.
Reason for revert: Internal test failure
Original change's description:
> Correctly handle retransmissions/padding in early loss detection.
>
> This CL makes sure we don't cull packets from the history based on
> incorrect ack mapping, just like it's predecessor:
> https://webrtc-review.googlesource.com/c/src/+/218000
>
> It also changes the logic to make sure retransmits counts towards
> history pruning - and properly ignores padding/fec.
>
> Bug: webrtc:12713
> Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34293}
TBR=danilchap@webrtc.org,terelius@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: Iaca6dc7739d953e97add5f5d516139b4819e43ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222601
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34294}
This CL makes sure we don't cull packets from the history based on
incorrect ack mapping, just like it's predecessor:
https://webrtc-review.googlesource.com/c/src/+/218000
It also changes the logic to make sure retransmits counts towards
history pruning - and properly ignores padding/fec.
Bug: webrtc:12713
Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34293}
Bilinear is faster but lesser quality, box is best quality. Our code
base has disagreed about which filter to use for quite some time,
causing aliasing bug reports. In an effort to avoid aliasing artifacts
and make our scaling filters more predictable, we're updating all uses
to kFilterBox.
WebRTC already uses kFilterBox everywhere except for these three
places. The main discrepency was between Chromium and WebRTC but that
has already been fixed. This CL fixes the last remaining bilinears.
This brings the WebRTC kFilterBox use count up from 11 to 14 and the
kFilterBilinear use count down from 3 to 0.
Bug: chromium:1212630
Change-Id: I5fe4aa92b9275d65b91ea97925533055d190d317
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221372
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34248}
This removes PacketRouter inheritance from RemoteBitrateObserver and TransportFeedbackSenderInterface.
Call binds methods for sending REMB and transport feedback messages from RemoteCongestionController to PacketRouter.
This is needed until the RTCPTranseiver is used instead of the RTP modules.
Bug: webrtc:12693
Change-Id: I7088de497cd6d1e15c98788ff3e6b0a2c8897ea8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215965
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33993}
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.
Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
to allow filtering of the frames that are being decoded.
Timestamp wraparound is not supported.
BUG=webrtc:12575
Change-Id: If08c46a377336e559475caefa934f6b82c46decc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211864
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33812}
This reverts commit cd5127b11e.
Reason for revert: Fuzzer build problems fixed.
Original change's description:
> Revert "Enable use of rtc::SystemTimeNanos() provided by Chromium"
>
> This reverts commit dfe19719e5.
>
> Reason for revert: Breaks fuzzers in Chromium builds. See https://ci.chromium.org/ui/p/chromium/builders/try/linux-libfuzzer-asan-rel/685438/overview. I am reverting since this blocks the roll but I will be in touch for a fix.
>
> Original change's description:
> > Enable use of rtc::SystemTimeNanos() provided by Chromium
> >
> > This is the third CL out of three to enable overriding
> > of the function SystemTimeNanos() in rtc_base/system_time.cc
> >
> > When WebRTC is built as part of Chromium the rtc::SystemTimeNanos()
> > function provided by Chromium will be used. This is controlled
> > by the build argument rtc_exclude_system_time which directly
> > maps to the macro WEBRTC_EXCLUDE_SYSTEM_TIME.
> >
> > By doing this we are making sure that the WebRTC and Chromium
> > clocks are the same.
> >
> > Bug: chromium:516700
> > Change-Id: If7f749c4aadefb1cfc07ba4c7e3f45dc6c31118b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208223
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33337}
>
> TBR=kron@webrtc.org
>
> Bug: chromium:516700
> Change-Id: I9ecd1784a6c1cdac8bae07d34f7df20c62a21a95
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208740
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33340}
Bug: chromium:516700
Change-Id: I4cd68bac1cc4befdb46351f5d6fb2cf1ef5c3062
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208742
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33341}
This reverts commit dfe19719e5.
Reason for revert: Breaks fuzzers in Chromium builds. See https://ci.chromium.org/ui/p/chromium/builders/try/linux-libfuzzer-asan-rel/685438/overview. I am reverting since this blocks the roll but I will be in touch for a fix.
Original change's description:
> Enable use of rtc::SystemTimeNanos() provided by Chromium
>
> This is the third CL out of three to enable overriding
> of the function SystemTimeNanos() in rtc_base/system_time.cc
>
> When WebRTC is built as part of Chromium the rtc::SystemTimeNanos()
> function provided by Chromium will be used. This is controlled
> by the build argument rtc_exclude_system_time which directly
> maps to the macro WEBRTC_EXCLUDE_SYSTEM_TIME.
>
> By doing this we are making sure that the WebRTC and Chromium
> clocks are the same.
>
> Bug: chromium:516700
> Change-Id: If7f749c4aadefb1cfc07ba4c7e3f45dc6c31118b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208223
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33337}
TBR=kron@webrtc.org
Bug: chromium:516700
Change-Id: I9ecd1784a6c1cdac8bae07d34f7df20c62a21a95
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208740
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33340}
This is the third CL out of three to enable overriding
of the function SystemTimeNanos() in rtc_base/system_time.cc
When WebRTC is built as part of Chromium the rtc::SystemTimeNanos()
function provided by Chromium will be used. This is controlled
by the build argument rtc_exclude_system_time which directly
maps to the macro WEBRTC_EXCLUDE_SYSTEM_TIME.
By doing this we are making sure that the WebRTC and Chromium
clocks are the same.
Bug: chromium:516700
Change-Id: If7f749c4aadefb1cfc07ba4c7e3f45dc6c31118b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208223
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33337}
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}
