Before 10.12, OSX may report 1X cursor on Retina screen. (See crbug.com/632995.)
After 10.12, OSX may report 2X cursor on non-Retina screen. (See
crbug.com/671436.) So scaling the cursor if the image size doesn't meet the
expected size on either Retina or non-Retina screen.
Also corrects the cursor caching and change detection, so we can only do scalingat cursor changing for better performance.
As to screen capture on OSX, the captured frame already contains the current
cursor. So the MouseCursorMonitorMac is not needed for ScreenCapture for
performance purpose.
BUG=671436
Review-Url: https://codereview.webrtc.org/2908853002
Cr-Commit-Position: refs/heads/master@{#18393}
Reason for revert:
Broken downstream projects
Original issue's description:
> Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls.
>
> Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
>
> BUG=webrtc:5079
> R=deadbeef@webrtc.org, hbos@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2863123002 .
> Cr-Commit-Position: refs/heads/master@{#18384}
> Committed: e80f4c91d0TBR=hbos@webrtc.org,deadbeef@webrtc.org,holmer@google.com,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2916793003
Cr-Commit-Position: refs/heads/master@{#18386}
A recent change (https://codereview.webrtc.org/2855143002/) introduced
a bug in RtcEventLogSource::NextPacket(). The rtp_packet_index_ must
be incremented when a valid packet is found and delivered. Otherwise,
the same packet will be delivered over and over again.
The recent change also altered the way that audio packets are sifted out. Now, the RTP header is always parsed before discarding any non-audio packets. This means that RtpHeaderParser::Parse is always called, also with video packets, which sometimes contain padding. When header-only dumps (such as RtcEventLogs) are created, the payload is stripped, and the payload length is equal to
the RTP header length. However, if the original packet was padded, the
RTP header will carry information about this padding length, and the
parser will check that the pyaload length is at least the header +
padding. This is not the case for header-only dumps, and the parser will return an error. In this CL, we ignore that error when a header-only packet has padding length larger than 0.
BUG=webrtc:7538
Review-Url: https://codereview.webrtc.org/2912323003
Cr-Commit-Position: refs/heads/master@{#18385}
Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
BUG=webrtc:5079
R=deadbeef@webrtc.org, hbos@webrtc.org
Review-Url: https://codereview.webrtc.org/2863123002 .
Cr-Commit-Position: refs/heads/master@{#18384}
Will also ensure that full-duplex audio now works on iOS simulators.
Bug: b/37580746
Change-Id: Iab1af39b0e6e6c124435814558caf77c474bd612
Reviewed-on: https://chromium-review.googlesource.com/519246
Commit-Queue: Henrik Andreasson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18381}
ScheduledExecutorService silently ignores exceptions thrown by the
runnable. This makes debugging issues unnecessarily difficult.
Bug: None
Change-Id: I7deb43b96e5639c096b9aed9c6ff9b197b62f59f
Reviewed-on: https://chromium-review.googlesource.com/521084
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18378}
Decision was made to keep this API for the time being.
Bug: webrtc:7710
Change-Id: Ief41ffb2ec2345e3a74fc72927d038be1ff5941c
No-Try: True
Reviewed-on: https://chromium-review.googlesource.com/521085
Reviewed-by: Henrik Andreasson <henrika@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18377}
Reason for revert:
Fixing and re-landing.
Original issue's description:
> Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ )
>
> Reason for revert:
> ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here.
> #include "webrtc/base/checks.h"
> ^-------------------
> The target:
> //webrtc/test:fileutils_unittests
> is including a file from the target:
> //webrtc/base:rtc_base_approved
>
> It's usually best to depend directly on the destination target.
> In some cases, the destination target is considered a subcomponent
> of an intermediate target. In this case, the intermediate target
> should depend publicly on the destination to forward the ability
> to include headers.
>
> Dependency chain (there may also be others):
> //webrtc/test:fileutils_unittests -->
> //webrtc/test:fileutils --[private]-->
> //webrtc/base:rtc_base_approved
>
>
> Original issue's description:
> > Enabling `gn check` on webrtc/test
> >
> > BUG=webrtc:6828
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2911203002
> > Cr-Commit-Position: refs/heads/master@{#18372}
> > Committed: db5bb404b0
>
> TBR=kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6828
>
> Review-Url: https://codereview.webrtc.org/2920763002
> Cr-Commit-Position: refs/heads/master@{#18375}
> Committed: 1a6f143d07TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2918793002
Cr-Commit-Position: refs/heads/master@{#18376}
Reason for revert:
ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here.
#include "webrtc/base/checks.h"
^-------------------
The target:
//webrtc/test:fileutils_unittests
is including a file from the target:
//webrtc/base:rtc_base_approved
It's usually best to depend directly on the destination target.
In some cases, the destination target is considered a subcomponent
of an intermediate target. In this case, the intermediate target
should depend publicly on the destination to forward the ability
to include headers.
Dependency chain (there may also be others):
//webrtc/test:fileutils_unittests -->
//webrtc/test:fileutils --[private]-->
//webrtc/base:rtc_base_approved
Original issue's description:
> Enabling `gn check` on webrtc/test
>
> BUG=webrtc:6828
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2911203002
> Cr-Commit-Position: refs/heads/master@{#18372}
> Committed: db5bb404b0TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2920763002
Cr-Commit-Position: refs/heads/master@{#18375}
Constants kRTCVp8CodecName, kRTCVp9CodecName, kRTCH264CodecName declared
in RTCRtpCodecParameters.h were defined without the "RTC" part in their
names, thus using them in the app code resulted in linking errors. This
patch fixes the naming mismatch.
BUG=webrtc:7721
Review-Url: https://codereview.webrtc.org/2910863002
Cr-Commit-Position: refs/heads/master@{#18363}
Similar to the existing constants for the media constraint-keys intended
for media sources, add the constants for the standard keys to generate
offers and answers.
This patch also adds a few comments to RTCMediaConstraints.h, to give
a better clue on the intended usage scope of declared media-constraint
keys and values.
BUG=webrtc:7722
Review-Url: https://codereview.webrtc.org/2908013002
Cr-Commit-Position: refs/heads/master@{#18362}
Only will be used if WEBRTC_POSIX and WEBRTC_LINUX are both defined and
"epoll_create" doesn't return an error. Otherwise the default "select"-based
IO loop will be used.
BUG=webrtc:7585
Review-Url: https://codereview.webrtc.org/2880923002
Cr-Commit-Position: refs/heads/master@{#18359}
1. Create unit tests for RtpDemuxer.
2. Add an RTC_DCHECK in RtpDemuxer that makes sure that the sink<->ssrc multimap does not allow multiple instances of the same association.
BUG=None
Review-Url: https://codereview.webrtc.org/2902823004
Cr-Commit-Position: refs/heads/master@{#18357}
Delete unused member |rtp_receiver_|, and simplify a return statement.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2912363002
Cr-Commit-Position: refs/heads/master@{#18354}
It seems it got lost in a rebase with this CL:
https://codereview.webrtc.org/2893843003/
Bug: None
Change-Id: Iaf4952593c1a1a9490d31c595b05ab155c0a809e
Reviewed-on: https://chromium-review.googlesource.com/519167
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Kári Tristan Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18353}
