Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.
Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1917043005
Cr-Commit-Position: refs/heads/master@{#12509}
Unit tests are updated to test that screen share is not adapted but it does not change the VideoSinkWants in WebRtcVideoEngine2::SendStream due to a switch to screen share. The reason is that it works anyway and sprang is looking into how to do adaptation based on frame rate as well and use the adapter for screen share as well.
BUG=webrtc:5688, webrtc:5426
R=nisse@webrtc.org, pbos@webrtc.org, sprang@google.com
Review URL: https://codereview.webrtc.org/1836043004 .
Cr-Commit-Position: refs/heads/master@{#12240}
It appears that the adapt_frame_drops, effects_frame_drops, and capturer_frame_time statistics are never used. They are collected by cricket::VideoCapturer, and copied into VideoSenderInfo by the VideoMediaChannel::GetStats method.
So delete the code to generate the statistics, and the VariableInfo template which had no other uses.
BUG=webrtc:5426
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1804133003 .
Cr-Commit-Position: refs/heads/master@{#12032}
I readded virtual bool Pause(bool paused) for now with a dummy implementation since Chrome remoting override this method.
Original cl description:
Removed unused cricket::VideoCapturer methods:
void UpdateAspectRatio(int ratio_w, int ratio_h);
void ClearAspectRatio();
bool Pause(bool paused);
Restart(const VideoFormat& capture_format);
MuteToBlackThenPause(bool muted);
IsMuted() const
set_square_pixel_aspect_ratio
bool square_pixel_aspect_ratio()
This cl also remove the use of messages and posting of state change.
Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
It does not add restrictions on what thread frames are delivered on though.
There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.
BUG=webrtc:5426
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1744153002 .
Cr-Commit-Position: refs/heads/master@{#11809}
The purose of this cl is to remove dependency on cricket::VideoCapturer from WebRtcVideoChannel2.
This cl change CPU adaptation to use a new VideoSinkWants.Resolution
Cl is WIP and uploaded to start the discussion.
Tested on a N5 with hw acceleration turned off.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1695263002
Cr-Commit-Position: refs/heads/master@{#11804}
Reason for revert:
Breaks remoting::protocol::WebrtcVideoCapturerAdapter::Pause'
See https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/3689/steps/compile/logs/stdio
Original issue's description:
> Removed unused cricket::VideoCapturer methods:
>
> void UpdateAspectRatio(int ratio_w, int ratio_h);
> void ClearAspectRatio();
> ool Pause(bool paused);
> Restart(const VideoFormat& capture_format);
> MuteToBlackThenPause(bool muted);
> IsMuted() const
> set_square_pixel_aspect_ratio
> bool square_pixel_aspect_ratio()
>
> This cl also remove the use of messages and posting of state change.
> Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
> It does not add restrictions on what thread frames are delivered on though.
>
> There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/e9c0cdff2dad2553b6ff6820c0c7429cb2854861
> Cr-Commit-Position: refs/heads/master@{#11773}
TBR=magjed@webrtc.org,pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1740963002
Cr-Commit-Position: refs/heads/master@{#11777}
void UpdateAspectRatio(int ratio_w, int ratio_h);
void ClearAspectRatio();
ool Pause(bool paused);
Restart(const VideoFormat& capture_format);
MuteToBlackThenPause(bool muted);
IsMuted() const
set_square_pixel_aspect_ratio
bool square_pixel_aspect_ratio()
This cl also remove the use of messages and posting of state change.
Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
It does not add restrictions on what thread frames are delivered on though.
There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1733673002
Cr-Commit-Position: refs/heads/master@{#11773}
Further more, it adds a VideoBroadcaster than is used for delivering frames to multiple sinks.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1655793003
Cr-Commit-Position: refs/heads/master@{#11563}
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
