If the content area of a window is not covered by the content area of another
window, we do not treat them as overlapping. This can fix the issue that two
fullscreen windows cover each other, or a fullscreen window is covered by the
shadow effect of task bar.
Bug: chromium:741770
Change-Id: I92dc636bc8445d50b00390cf3332695f69daf9c6
Reviewed-on: https://chromium-review.googlesource.com/628244
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19532}
WindowCapturerWin wrongly calculate the image size if the application it target
does not support high DPI. It causes part of the output frame black. See bug for
details.
Bug: webrtc:8112
Change-Id: I33c66dfa977ec08a29c56ff86ae37320b1459c87
Reviewed-on: https://chromium-review.googlesource.com/634383
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19531}
Reason for revert:
Might be the root cause of an internal test failure.
Original issue's description:
> Verify sender ssrc when receiving rtcp target bitrate
>
> BUG=webrtc:8137
>
> Review-Url: https://codereview.webrtc.org/3000373002
> Cr-Commit-Position: refs/heads/master@{#19524}
> Committed: a7a4beb419TBR=danilchap@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:8137
Review-Url: https://codereview.webrtc.org/3005633002
Cr-Commit-Position: refs/heads/master@{#19529}
A crash may randomly happen in IsWindowMinimized(), the potential reason is that
|on_screen| is not retrieved from |window| with kCGWindowIsOnScreen property. So
add a == NULL check before executing CFBooleanGetValue().
Bug: chromium:758554
Change-Id: I25ad1ddbb21ec049ef237e55a8d25156bcd982c7
Reviewed-on: https://chromium-review.googlesource.com/634043
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19527}
This avoids infinite recursion in case the recovered packet carries a
RED header.
BUG=chromium:754748
Review-Url: https://codereview.webrtc.org/3004553002
Cr-Commit-Position: refs/heads/master@{#19525}
I think this was previously used in libjingle but it is not
used in WebRTC.
BUG=webrtc:8082
Review-Url: https://codereview.webrtc.org/2999703002
Cr-Commit-Position: refs/heads/master@{#19523}
These tests are very old and come from a time when we tested each method in the
ADM as if the ADM should function as a standalone component.
Several tests are already disabled and we test combinations of APIs that are no
longer valid (since the ADM is now used in a more fixed way in VoE).
The tests does not verify media (we have other tests under
voice_engine/test/auto_test) which starts media and verifies that it works OK.
There are also a a more extensive set of ADM tests for Android and iOS.
You could also say that these tests tests the most "hardware related parts of
the ADM", but not those that we expose via the VoEHardware API.
Hence, not much value to maintain them imo.
NOTRY=TRUE
BUG=webrtc:7250
Review-Url: https://codereview.webrtc.org/2726433003
Cr-Commit-Position: refs/heads/master@{#19522}
This function is currently only available in EglBase14, which is not in
the api. This CL adds the swapBuffer method to the public EglBase
interface. For EglBase10, the presentation time is just ignored.
BUG=webrtc:8155
Review-Url: https://codereview.webrtc.org/3003843002
Cr-Commit-Position: refs/heads/master@{#19518}
In later steps the Win/Mac implementation will also be refactored.
The rtc_task_queue target is split up in three separate targets:
rtc_task_queue_api:
Contains the header file task_queue.h but no implementation.
Only external TaskQueue implementations should directly depend on this target.
rtc_task_queue_impl:
Contains the default implementation of task_queue.h.
Only external application targets should directly depend on this target.
rtc_task_queue:
WebRTC targets should depend on this target. It unconditionally depend on rtc_task_queue_api and depending on the new build flag,|rtc_link_task_queue_impl|, depend on rtc_task_queue_impl.
BUG=webrtc:8160
Review-Url: https://codereview.webrtc.org/3003643002
Cr-Commit-Position: refs/heads/master@{#19516}
Now we always process |rate_profile.num_frames| number of frames.
This means that the output of the tests in
videoprocessor_integrationtest.cc will be slightly different,
as we will no process 300 frames, instead of 299. No rate control
or quality thresholds need to be updated, however.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/3004583002
Cr-Commit-Position: refs/heads/master@{#19515}
The whole point of all the audio codec stuff we've recently published
in api/ is to function as lego bricks so that building stuff like our
builtin audio codec factories will be easy.
(This has landed once before, but got reverted because of Chromium test
failures---apparently, someone isn't ignoring the case of the codec names
like they're supposed to. The quick fix was to preserve the same case
used by the old implementation.)
BUG=webrtc:7821, webrtc:7822
Review-Url: https://codereview.webrtc.org/2998263002
Cr-Commit-Position: refs/heads/master@{#19512}
When adapting a frame, any initial cropping coordinate was replaced
by the coordinate calculated for adapting. Offset initial coordinate
instead.
BUG=webrtc:7880
Review-Url: https://codereview.webrtc.org/2975613002
Cr-Commit-Position: refs/heads/master@{#19509}
RtpVideoStreamReceiver::receive_cs_ is not really necessary, since all of the functions where that lock is acquired, are arrived at from functions of BaseChannel which DCHECK being called from BaseChannel::worker_thread_.
BUG=webrtc:8037
Review-Url: https://codereview.webrtc.org/2987933003
Cr-Commit-Position: refs/heads/master@{#19508}
Adds two new stats to RTCMediaStreamTrackStats:
* totalSamplesReceived is the total number of samples received on
the audio channel and includes real and synthetic samples.
* concealedSamples is the total number of synthetic samples
received on the audio channel used to conceal packet loss.
Bug: webrtc:8076
Change-Id: I36e9828525ec341490cf3310a972b56a8b443667
Reviewed-on: https://chromium-review.googlesource.com/615902
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19506}
Clang format wants to put all parameters to WEBRTC_RTCSTATS_IMPL on
their own line, but we prefer to keep the header on the first line and
put the variable arguments (the stats) on their own lines.
Bug: None
Change-Id: Ic53104b46fb3c91c134d524027a150ba10055ad5
Reviewed-on: https://chromium-review.googlesource.com/621409
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19505}
The original change https://chromium-review.googlesource.com/c/575315 and
https://chromium-review.googlesource.com/c/590508 have not been well-considered.
So this change reverts part of two changes and adds a
DesktopFrame::set_top_left() function
A DesktopFrame usually contains a very large chunk of memory, which should be
reused as much as possible to reduce the memory allocations. The size of the
memory usually controls by the DesktopFrame::size(). So it's reasonable to const
DesktopFrame::size_: changing it is wrong if the underly buffer is not large
enough.
But DesktopFrame::top_left() is a different story, same as capturer_id,
capture_time_ms and other information in the DesktopFrame, it can be changed to
any value without needing to reconstruct a DesktopFrame instance. So instead of
adding it to the constructor, a DesktopFrame::set_top_left() is added to adjust
the top-left of the DesktopFrame in the entire display coordinate.
After adding DesktopFrame::set_top_left(), we have five variables in a
DesktopFrame which is not initialized in the constructor. For any kind of
wrapper DesktopFrame, say, SharedDesktopFrame and CroppedDesktopFrame, they
needs to copy these five variables after constructing themselves. This is not
convenient and easily to be broken if an implementation forgot to copy them.
So DesktopFrame::MoveFrameInfoFrom() and DesktopFrame::CopyFrameInfoFrom() are
added to the DesktopFrame to help derived classes to copy or move these
variables in one function call.
The difference between MoveFrameInfoFrom() and CopyFrameInfoFrom() is that the
former one uses DesktopRegion::Swap() to move the DesktopRegion from the source
DesktopFrame to this instance, while the later one uses copy-operator to copy
the DesktopRegion from the source DesktopFrame.
So CopyFrameInfoFrom() is usually used when sharing a source DesktopFrame with
several clients. I.e. the source DesktopFrame should be kept unchanged. For
example, BasicDesktopFrame::CopyOf() and SharedDesktopFrame::Share().
On the other side, MoveFrameInfoFrom() is usually used when wrapping a
DesktopFrame. E.g. CroppedDesktopFrame and DesktopFrameWithCursor.
Bug: webrtc:7950
Change-Id: I8b23418960fb681d2ea1f012d1b453f514da2272
Reviewed-on: https://chromium-review.googlesource.com/622453
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19504}
Windows cannot capture contents on VMs hosted in GCE, disable them to unblock
GCE hosting.
Bug: webrtc:8153
Change-Id: Iacdce15008cc092dce36d08b1d5565bbaa5def1f
Reviewed-on: https://chromium-review.googlesource.com/634083
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19502}
NetEq network statistics contains discard rate but has not been used and even not been implemented until recently.
According to w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsdiscarded,
this statistics needs to be replaced with an accumulative stats. Such work will be carried out separately.
Meanwhile, we need to add a rate to reflect rate of discarded redundant packets. See webrtc:8025.
In this CL, we replace the existing discard rate with secondary discarded rate, so as to
1. fulfill the requests on webrtc:8025
2. get ready to implement an accumulative statistics for discarded packets.
BUG: webrtc:7903,webrtc:8025
Change-Id: Idbf143a105db76ca15f0af54848e1448f2a810ec
Reviewed-on: https://chromium-review.googlesource.com/582863
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19495}
We have been using the Linux way of getting the thread id on Android,
even though we have a special code path for getting it on Android.
BUG=webrtc:8110
Review-Url: https://codereview.webrtc.org/3001333002
Cr-Commit-Position: refs/heads/master@{#19494}
Fixes bug where callback is set to null on reinitialization. Also fixes
a race condition where callback can be null in onTextureFrameAvailable.
BUG=webrtc:8124
Review-Url: https://codereview.webrtc.org/3002093002
Cr-Commit-Position: refs/heads/master@{#19493}
This CL ensures that the adaptive filter delay is not used for fine
tune echo removal unless the render and capture signals have been
properly aligned.
BUG=webrtc:8189
Review-Url: https://codereview.webrtc.org/3003303002
Cr-Commit-Position: refs/heads/master@{#19492}
This CL robustifies the inaudible echo detection in AEC3 such that a
requirement is that either the render and capture signals are aligned
or that a headset has been detected. This ensures that the inaudible
detection has been able to base the desicion on reliable signals.
BUG=webrtc:8150
Review-Url: https://codereview.webrtc.org/3005503002
Cr-Commit-Position: refs/heads/master@{#19491}
This will help debugging issues with the fallback, in cases where
only LS_WARNING logs are available.
BUG=none
Review-Url: https://codereview.webrtc.org/3007523002
Cr-Commit-Position: refs/heads/master@{#19488}
Add GUARDED_BY annotation to DirectTransport to demonstrate that next_scheduled_task_ is properly protected.
SendRtp(), SendRtcp(), etc., remain as unprotected as they were before the migration from using a thread to using a task-queue.
BUG=webrtc:8113
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/3002163003
Cr-Commit-Position: refs/heads/master@{#19487}
We can't handle no value here anyway and end up setting a default
at each call site. The defaults aren't even the same in each place.
BUG=None
Review-Url: https://codereview.webrtc.org/2998293002
Cr-Commit-Position: refs/heads/master@{#19485}
Testcode updated to use webrtc::test::TempFilename.
Also deletes now unused functions AppleDataDirectory,
AppleTempDirectory and AppleAppName.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2995413002
Cr-Commit-Position: refs/heads/master@{#19483}
