This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
This CL makes AddRef() and Release() const member methods and the refcount integer mutable. This is reasonable, because they only manage the lifetime of the object, and this is also how it's done in Chromium.
The purpose is to be able to capture a const pointer in a scoped_refptr, which is currenty impossible. The practial problem this CL solves is this:
void Foo::Bar() const {}
rtc::Callback0<void> Foo::MakeClosure() const {
return rtc::Bind(&Foo::Bar, this);
}
We currently capture |this| as const Foo*. With this CL, |this| will be captured as scoped_refptr<const Foo>.
A test is also added in bind_unittest to check this behaviour.
BUG=webrtc:5065
R=perkj@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1403683004 .
Cr-Commit-Position: refs/heads/master@{#10253}
It is just a pass through to webrtc/video_frame.h. Updated the callers
to include webrtc/video_frame.h instead and removed i420_video_frame.h.
This should fix pbos' TODO in i420_video_frame.h.
Tested on Linux with the following command lines:
$ rm -rf out/
$ ./webrtc/build/gyp_webrtc
$ ninja -C out/Debug
BUG=None
TEST=see above
R=magjed@webrtc.org, pbos@webrtc.org, tommi@webrtc.orgTBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46819004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8973}
Original cl description:
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
This cl was previously reverted in https://webrtc-codereview.appspot.com/46549004/.
Patchset 1 contains the original patch after rebase.
Patshet 2 fix webrtc_perf_tests reported in chromium:465306
Note that chromium:465287 is being fixed in https://webrtc-codereview.appspot.com/43829004/
BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47629004
Cr-Commit-Position: refs/heads/master@{#8776}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8776 4adac7df-926f-26a2-2b94-8c16560cd09d
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46429004
Cr-Commit-Position: refs/heads/master@{#8633}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8633 4adac7df-926f-26a2-2b94-8c16560cd09d
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.
This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".
BUG=chromium:81439
TEST=none
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
Also enables running video_capture_tests_apk on the WebRTC/Chromium APK bots,
assuming GYP_DEFINES includes include_tests=1 and
include_internal_video_capture=1.
This required running VideoCaptureAndroid's camera capture on a dedicated thread, matching other platform's video_capture impls.
BUG=2974,3152
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5868 4adac7df-926f-26a2-2b94-8c16560cd09d
