- Makes vt h264 decoder output CoreVideoFrameBuffer
- Makes iOS renderer convert frame buffer if it is not i420
BUG=
Review URL: https://codereview.webrtc.org/1853503003
Cr-Commit-Position: refs/heads/master@{#12224}
- Posts to WebRTC thread instead of Send
- Sample buffers are returned on capture session queue instead of main queue
- Camera switch happens on captures session queue
BUG=webrtc:5679, webrtc:4212
Review URL: https://codereview.webrtc.org/1838933004
Cr-Commit-Position: refs/heads/master@{#12186}
We shouldn't make external users define this flag to use our file.
BUG=
Review URL: https://codereview.webrtc.org/1825713003
Cr-Commit-Position: refs/heads/master@{#12106}
Also removes unused track states kLive and kFailed.
Since this also required a Video source to exist in all unit tests that create a track, a FakeVideoTrackSource is added and used in tests.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1790633002
Cr-Commit-Position: refs/heads/master@{#12098}
initWithFactory: is clumsy and makes classes difficult to mock out in
tests. By keeping methods on the factory, we can simply mock out the
factory's methods instead.
We can consider adding regular Obj-C like ctors if we move to making
the factory a singleton, but that requires further discussion.
BUG=
R=haysc@webrtc.org, hjon@webrtc.org
Review URL: https://codereview.webrtc.org/1820193002 .
Cr-Commit-Position: refs/heads/master@{#12089}
Reason for revert:
New attempt. Cl for removing videosourceinterface.h dep in chrome is landed here: https://codereview.chromium.org/1810273003/
Original issue's description:
> Revert of Delete empty API files and cleaned up includes. (patchset #2 id:20001 of https://codereview.webrtc.org/1809053002/ )
>
> Reason for revert:
> Breaks Chromium build. Need to remove the references to the obsolete header files from Chromium and reland.
>
> Original issue's description:
> > Delete empty API files and cleaned up includes.
> >
> > TBR=glaznev@webrtc.org
> >
> > BUG=webrtc:5426
> >
> > Committed: https://crrev.com/c9022f508644dc33c01b05cb22ebfc2be145d6b2
> > Cr-Commit-Position: refs/heads/master@{#12039}
>
> TBR=nisse@webrtc.org,glaznev@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5426
>
> Committed: https://crrev.com/246b5273986d5a5b140b3d1a656baa8d40c36276
> Cr-Commit-Position: refs/heads/master@{#12042}
TBR=nisse@webrtc.org,glaznev@webrtc.org,deadbeef@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1819733002
Cr-Commit-Position: refs/heads/master@{#12065}
This CL removes copy and assign support from Buffer and changes various
parameters from Buffer to CopyOnWriteBuffer so they can be passed along
and copied without actually copying the underlying data.
With this changed some parameters to be "const" and fixed an issue when
creating a CopyOnWriteBuffer with empty data.
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1823503002
Cr-Commit-Position: refs/heads/master@{#12062}
Reason for revert:
I'm really sorry for having to revert this but it seems this hit an unexpected compile error downstream:
webrtc/media/sctp/sctpdataengine.cc: In function 'void cricket::VerboseLogPacket(const void*, size_t, int)':
webrtc/media/sctp/sctpdataengine.cc:172:37: error: invalid conversion from 'const void*' to 'void*' [-fpermissive]
data, length, direction)) != NULL) {
^
In file included from webrtc/media/sctp/sctpdataengine.cc:20:0:
third_party/usrsctp/usrsctplib/usrsctp.h:964:1: error: initializing argument 1 of 'char* usrsctp_dumppacket(void*, size_t, int)' [-fpermissive]
usrsctp_dumppacket(void *, size_t, int);
^
I'm sure you can fix this easily and just re-land this CL, while I'm going to look into how to add this warning at the public bots (on Monday).
Original issue's description:
> Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
>
> This CL removes copy and assign support from Buffer and changes various
> parameters from Buffer to CopyOnWriteBuffer so they can be passed along
> and copied without actually copying the underlying data.
>
> With this changed some parameters to be "const" and fixed an issue when
> creating a CopyOnWriteBuffer with empty data.
>
> BUG=webrtc:5155
>
> Committed: https://crrev.com/944c39006f1c52aee20919676002dac7a42b1c05
> Cr-Commit-Position: refs/heads/master@{#12058}
TBR=kwiberg@webrtc.org,tkchin@webrtc.org,tommi@webrtc.org,pthatcher@webrtc.org,jbauch@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1817753003
Cr-Commit-Position: refs/heads/master@{#12060}
This CL removes copy and assign support from Buffer and changes various
parameters from Buffer to CopyOnWriteBuffer so they can be passed along
and copied without actually copying the underlying data.
With this changed some parameters to be "const" and fixed an issue when
creating a CopyOnWriteBuffer with empty data.
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1785713005
Cr-Commit-Position: refs/heads/master@{#12058}
Reason for revert:
Breaks Chromium build. Need to remove the references to the obsolete header files from Chromium and reland.
Original issue's description:
> Delete empty API files and cleaned up includes.
>
> TBR=glaznev@webrtc.org
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/c9022f508644dc33c01b05cb22ebfc2be145d6b2
> Cr-Commit-Position: refs/heads/master@{#12039}
TBR=nisse@webrtc.org,glaznev@webrtc.org,perkj@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/1813083002
Cr-Commit-Position: refs/heads/master@{#12042}
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}
Devices exist (specifically the 16GB 5th generation iPod Touch) that do not have
a rear-facing camera.
This CL:
- Adjusts RTCAVFoundationVideoCapturerInternal so initialization doesn't fail
because a rear-facing camera doesn't exist, but still logs a warning
- Provides a check for whether or not a rear-facing camera can be used (useful
for toggling UI elements)
- Turns an attempt to switch to the rear-facing camera into a no-op with a
warning.
BUG=
Review URL: https://codereview.webrtc.org/1416303003
Cr-Commit-Position: refs/heads/master@{#11992}
Adds setConfiguration back and renames statsId back to reportId.
BUG=
Review URL: https://codereview.webrtc.org/1778033002
Cr-Commit-Position: refs/heads/master@{#11936}
Moved VideoSourceInterface to MediaStreamInterface.h
Renamed VideoSourceTest to VideoCapturerTrackSourceTest
Renamed VideoSource to VideoCaptureTrackSource and cl lint and cl format.
BUG=webrtc:5426
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1770003002 .
Cr-Commit-Position: refs/heads/master@{#11893}
We'd like to completely replace rtc::scoped_ptr with std::unique_ptr.
This is a first trial CL to see if using unique_ptr causes any
problems.
(As a side effect of removing the scoped_ptr.h include in buffer.h,
I had to fix broken includes in no less than three files.)
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1687833006
Cr-Commit-Position: refs/heads/master@{#11588}
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.
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
The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.
I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc 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/1615433002
BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1610243002 .
Cr-Commit-Position: refs/heads/master@{#11545}
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}
