This CL reduces the complexity of the Subtractor.ConvergenceMultiChannel
test by
1. Slightly reducing the amount of tested combinations for the non-debug
mode.
2. Drastically reduce the amount of tested combinations for the debug
mode.
Bug: webrtc:11295
Change-Id: I56bfa4a1463d26e5217b6a4d7f2ef54de7aab512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166529
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30343}
This makes it easier to maintain consistency between real time
and simulated time modes.
The RealTimeController is updated to use an explicit main thread,
this ensures that pending destruction tasks are run as the network
emulator goes out of scope.
Bug: webrtc:11255
Change-Id: Ie73ab778c78a68d7c58c0f857f14a8d8ac027c67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166164
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30342}
Before this change, if both app and encoder provided bitrate limits,
WebRTC ignored the limits provided by encoder. Now intersection of
these sets is used.
Also changed DCHECKs in GetEncoderBitrateLimits to allow zero values
of min_bitrate_bps and min_start_bitrate_bps.
Bug: none
Change-Id: Ib8be965ea43f51013b0a0f82fd4256a372432dda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166600
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30338}
This change clarifies the semantics of this field:
unset: Depends on context.
== 0: Invalid.
== 1: No temporal layering.
>= 2: Temporal layering.
We should try to remove the wrapping optional later.
Bug: webrtc:11297
Change-Id: Id765f2dc1d31a4ba3cd424978ac6054cd60152ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166528
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30336}
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.
Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.
CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn
Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
This squashes together several input signals that were spread out
through several calls into a single method and calling place:
SetEncoderSettings(), invoked from ReconfigureEncoder(). This is added
to the abstract interface.
This makes the following methods obsolete which are removed:
- SetEncoder(): The VideoEncoder was only used for GetEncoderInfo();
the VideoEncoder::EncoderInfo is now part of the EncoderSettings.
- SetEncoderConfig(): The VideoEncoderConfig is part of
EncoderSettings. The config is used for its codec_type and
content_type enums.
- SetCodecMaxFrameRate(): The max frame rate was the same as
VideoCodec::maxFramerate. VideoCodec is now part of EncoderSettings.
There may be some overlap in information between EncoderConfig and
VideoCodec, but that is outside the scope of this CL, which only makes
sure to bundle encoder settings-like information into one input signal.
Bug: webrtc:11222
Change-Id: I67c49c49c0a859cb7d5051939a461593c695a789
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166602
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30332}
Before reformatting GN files (see [1] for why this is needed), the
presubmit check to ensure targets are not violating package boundaries
needs to be fixed because its regular expressions don't always work with
the new format.
This CL removes the parsing of line numbers to relax the regular
expressions without losing any functionality.
Error before this CL:
***************
<PATH>/webrtc/src/BUILD.gn:674 in target 'android_junit_tests':
Source file 'examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java'
crosses boundary of package 'examples'.
<PATH>/webrtc/src/BUILD.gn:675 in target 'android_junit_tests':
Source file 'examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java'
crosses boundary of package 'examples'.
<PATH>/webrtc/src/BUILD.gn:676 in target 'android_junit_tests':
Source file 'examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java'
crosses boundary of package 'examples'.
<PATH>/webrtc/src/BUILD.gn:677 in target 'android_junit_tests':
Source file 'sdk/android/tests/src/org/webrtc/AndroidVideoDecoderTest.java'
crosses boundary of package 'sdk'.
<PATH>/webrtc/src/BUILD.gn:678 in target 'android_junit_tests':
Source file 'sdk/android/tests/src/org/webrtc/CameraEnumerationTest.java'
crosses boundary of package 'sdk'.
***************
Error after this CL:
***************
<PATH>/webrtc/src/BUILD.gn in target 'android_junit_tests':
Source file 'examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java'
crosses boundary of package 'examples'.
<PATH>/webrtc/src/BUILD.gn in target 'android_junit_tests':
Source file 'examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java'
crosses boundary of package 'examples'.
<PATH>/webrtc/src/BUILD.gn in target 'android_junit_tests':
Source file 'examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java'
crosses boundary of package 'examples'.
<PATH>/webrtc/src/BUILD.gn in target 'android_junit_tests':
Source file 'sdk/android/tests/src/org/webrtc/AndroidVideoDecoderTest.java'
crosses boundary of package 'sdk'.
<PATH>/webrtc/src/BUILD.gn in target 'android_junit_tests':
Source file 'sdk/android/tests/src/org/webrtc/CameraEnumerationTest.java'
crosses boundary of package 'sdk'.
***************
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11302
Change-Id: Ia39387d089a0c56a2c3ad9a7264c20eb5a38ac93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166535
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30331}
This is the first step to move //:android_junit_tests to the righ
package (the target is triggering presubmit errors every time //BUILD.gn
gets updated).
Next steps:
* Update recipes
* Remove //:android_junit_tests
Issues with GN formatting, introduced by [1] will be addressed
separately in a "format all" CL.
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11289
No-Presubmit: True
Change-Id: I70c0927d722911f82dd971c30c7ffb581aed69c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166603
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30328}
The extension (and thus structures to carry it) are designed
in particular for client<->SFU link. Putting the structure into api
acknowledges it can be reused by SFU projects
Bug: webrtc:10342
Change-Id: I8ca1f5046abadf6aa16200443c4892e9a2a928b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166467
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30324}
to allow writing DependencyDescriptor value that is not copiable.
and avoid copying RtpGenericFrameDescriptor
Bug: webrtc:10342
Change-Id: I6eefa9d06b90d7e858f224443ba6769975b556fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166171
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30322}
This CL does two things for the sake of getting us closer to adaptation
modules being injectable and usable without knowing implementation
details.
Firstly, RefreshTargetFramerate() is removed. The target frame rate is
dependent on two things: 1) the codec max frame rate, and 2) the video
source restrictions. If either of these two changes, the target frame
rate is updated - there is no need to trigger this externally; the
module already knows if either of these factors change.
The private method MaybeUpdateTargetFrameRate() is added to ensure
overuse_detector->OnTargetFramerateUpdated() happens when necessary.
In doing this, the frame rates are updated to use
absl::optional<double>. This documents its optionality and avoids
magical values (previously -1 was not a bug but meaning "missing"). It
also matches VideoSourceRestrictions::max_frame_rate()'s type.
Secondly, ResetAdaptationCounters() is renamed
ResetVideoSourceRestrictions(). This more accurately describes what it
is doing; it is resetting the restrictions (the adaptation counters
getting reset is merely an implementation specific side-effect of
this). This method is added to the generic interface.
The usefulness of being able to ResetVideoSourceRestrictions() is
questioned in a TODO - current usage of this is when "quality rampup"
finishes. Nevertheless, any module could implement this functionality
so it belongs to the interface for now.
Bug: webrtc:11222
Change-Id: I079785df55fc9894e85087ec98be3e4ebd0713c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166522
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30320}
This is a reland of 2a11b2451a4068746fa0c55fa210efd4a15e4423
There are no changes compared to the first attempt.
Original change's description:
> Enable using a custom NetEqFactory in simulations
>
> Bug: webrtc:11005
> Change-Id: I8a15f77953cbd3c29a75c7cfc77f926b138994b9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165580
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30286}
TBR=kwiberg
Bug: webrtc:11005
Change-Id: I4aa377e05916bd23f8f63aece9d0e27731c80d3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166465
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30319}
DesktopAndCursorComposer already handles a null MouseCursorMonitor. This
CL allows that code-path to be utilized by callers that already have a
MouseCursorMonitor, allowing its callbacks to be re-used by this class.
This is more efficient, and works around an apparent X Server deadlock
on Linux if multiple MouseCursorMonitors are simultaneously active.
The intended use-case for this is to allow the host-side cursor to be
composited into the desktop image if mouse-lock is active at the client.
Bug: chromium:1043325
Change-Id: I7e036850dd8c17fe55e57db252392062a847d10f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166581
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30312}
The only callers or non-trivial implementations of this that I could
find are in remoting/ in Chromium, which I plan on fixing once this
gets rolled.
Bug: chromium:1043325
Change-Id: Id5a33fc09bb066f979876b2a7dcbc3dc5c2d3dd9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166560
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30310}
We make sure the fake clock is constructed first thing,
so that all subsequent calls to GetClockForTesting() are
consistent and non-racy.
This proper scoping also allows to remove some explicit
destructions which are no longer necessary.
Bug: webrtc:11282
Change-Id: Id9263617c2e2b025b17d9bcb9cd415d651405a8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166043
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30309}
This will avoid getting into an inconsistent state where isInterrupted==YES while isActive==YES.
Bug: webrtc:11112
Change-Id: Ia4db85483e1e7a339f520d52a2feb475a73c262e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160140
Commit-Queue: Joe Chen <jsphchn@google.com>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30306}
These constants describes how value should be put on the wire and thus
belong to the extension builder/writer class rather than extension value class
Bug: None
Change-Id: I65ca3923eddcc2e48563ad69b98356c159ad86be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166461
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30305}
This script will be used when tests write proto-backed JSON. It still
has to reside source-side because we need to access the catapult Python
API to get at HistogramSet and reserved_infos, etc.
WebRTC tests will write proto-backed JSON, and this script can read
it because the Histogram class has been made capable of doing it.
Build information diagnostics are added, and then we upload in the
old JSON format (the dashboard can read the new format as well, but
there's no reason to implement export to the new format at this point).
We could imagine more outlandish solutions where the test binaries
themselves do the uploading, but then we would have to pass the
build information to them, and they would have to upload from the
shards. Alternatively, we could pass build information to tests so
they write it right into the histograms.
This solution is probably the best one for now since it's
1) consistent with how Chromium does it
2) flexible in the right ways
3) we don't have to worry if uploading from shards even works.
Bug: webrtc:11084
Change-Id: I8888ce9f24e0ca58f984d2c2e9af7740ee5e89b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166464
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30301}
This changes the behavior for adding virtual transport overhead so it
doesn't change the size of the actual payload buffer, only the
calculated packet size.
Bug: webrtc:9883
Change-Id: I6e24598378c4dd6a591d36ca3b162e933ff4ef7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164523
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30298}
The "has_input_video_ logic" is simplified to abort AdaptUp() and
AdaptDown() directly rather than in each calling place of the
VideoSourceRestrictor. The intent is no change in behavior.
The degradation_preference_ is removed from the VideoSourceRestrictor
as its only usage was DCHECKing (not worth it).
ResourceAdaptationModuleInterface gets SetHasInputVideo() and
SetDegradationPreference(), making these things controllable without
knowing implementation details.
StartCheckForOveruse() and StopCheckForOveruse() are renamed to
StartResourceAdaptation() and StopResourceAdaptation().
Bug: webrtc:11222
Change-Id: Id2d7f34d427dfb3ecd4831b1a245d07becae6520
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166173
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30296}
This is a reland of f3aa6326b8e21f627b9fba72040122723251999b
Original change's description:
> Replace the ExperimentalAgc config with the new config format
>
> This CL replaces the use of the ExperimentalAgc config with
> using the new config format.
>
> Beyond that, some further changes were made to how the analog
> and digital AGCs are initialized/called. While these can be
> made in a separate CL, I believe the code changes becomes more
> clear by bundling those with the replacement of the
> ExperimentalAgc config.
>
> TBR: saza@webrtc.org
> Bug: webrtc:5298
> Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30149}
Bug: webrtc:5298
Change-Id: I6db03628ed3fa2ecd36544fe9181dd8244d7e2df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165760
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30295}
