DesktopAndCursorComposer adds the cursor image to the desktop, but does
not change the updated_region, so it generally doesn't encode correctly
unless the mouse is moving over a region that is changing. This CL
extends the updated region to include the union of the old and new
cursor rects, with an optimization for the case where the cursor has
neither moved nor changed.
Bug: chromium:1043325
Change-Id: I52076c96528820833fda6aa95f5b1fbc0f613909
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166545
Reviewed-by: Sergey Ulanov <sergeyu@google.com>
Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30374}
When creating a NetEqController it can be useful to have access to a
webrtc::Clock*. Also, NetEqControllers should have access to the
contents of the sync buffer when making decisions.
Bug: webrtc:11005
Change-Id: I7fdba75ce661b2ace52458620a8c1f3c990e5ac2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167208
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30368}
This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
Reason for revert: Breaks Chromium import due to flaky test in Chromium.
Original change's description:
> Reland "Distinguish between send and receive codecs"
>
> This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
>
> Reason for revert: Fixed negotiation of send-only clients.
>
> Original change's description:
> > Revert "Distinguish between send and receive codecs"
> >
> > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> >
> > Reason for revert: breaks negotiation with send-only clients
> >
> > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> >
> > Original change's description:
> > > Distinguish between send and receive codecs
> > >
> > > Even though send and receive codecs may be the same, they might have
> > > different support in HW. Distinguish between send and receive codecs
> > > to be able to keep track of which codecs have HW support.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30284}
> >
> > TBR=steveanton@webrtc.org,kron@webrtc.org
> >
> > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30292}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
>
> Bug: chromium:1029737
> Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30348}
TBR=steveanton@webrtc.org,kron@webrtc.org
Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30360}
This commit enables developers to configure the "," delay value from
the WebRTC spec value of 2 seconds. This flexibility allows developers
to comply with existing WebRTC clients.
Bug: webrtc:11273
Change-Id: Ia94b99e041df882e2396d0926a8f4188afe55885
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165700
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30354}
This defines the following methods:
- OnFrame(), replaces SetLastFramePixelCount().
- OnFrameDroppedDueToSize(), a rename of FrameDroppedDueToSize() to
match the other methods.
- OnEncodeStarted(), a rename of the incorrectly named FrameCaptured().
- OnEncodeCompleted(), a rename of the poorly named FrameSent().
In order to get rid of SetLastFramePixelCount(), the "we don't know the
frame size" use case - which was previously implicitly avoided by
invoking SetLastFramePixelCount() with a made-up value for
last_frame_info_ - is now avoided using ".value_or()" in
LastInputFrameSizeOrDefault(). This does mean that a constant 144p
resolution value is referenced in two places, but the fact that this is
a magic value is at least made explicit. This may help future
improvements.
Bug: webrtc:11222
Change-Id: I3b28daa8c5ecf57c6537957d4759f15e24bb2234
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166961
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30352}
All uses of encoded_image are const, except for the copy for running on
the encoder_queue_.
Bug: None
Change-Id: I7fc8cb46f6afb42a2d27961d3d3ff8d9e63fe1b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166442
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30351}
This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
Reason for revert: Fixed negotiation of send-only clients.
Original change's description:
> Revert "Distinguish between send and receive codecs"
>
> This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
>
> Reason for revert: breaks negotiation with send-only clients
>
> (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
>
> Original change's description:
> > Distinguish between send and receive codecs
> >
> > Even though send and receive codecs may be the same, they might have
> > different support in HW. Distinguish between send and receive codecs
> > to be able to keep track of which codecs have HW support.
> >
> > Bug: chromium:1029737
> > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30284}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
> Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30292}
TBR=steveanton@webrtc.org,kron@webrtc.org
Bug: chromium:1029737
Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30348}
The poorly named SetEncoderStartBitrate() is renamed
SetEncoderTargetBitrate() and added to the abstract resource adaptation
module interface.
The so-called "start bitrate" was updated to match the target bitrate,
so this was only ever a "start bitrate" until we had any estimates. The
variable is renamed in VideoStreamEncoder as well, and usage of optional
types are introduced to avoid magical values in a few places in the
existing code.
Bug: webrtc:11222
Change-Id: Idde92f68f34616aa3c34ab77a791fdbe7ea7af26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166880
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@{#30347}
Add it to default build target, so it won't get broken accidentally
again. Fix configuration issue with field trials (new parameter was
added recently, but wasn't set by video_replay)
Bug: webrtc:11287
Change-Id: I9c18746d899acd7ac68c1b9b3a646b862c41897a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166900
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30345}
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}
