This inline function is no longer expanded on arm Android, but on x86 Android it
will still be expanded. Move it out-of-line to make things consistent.
This change list will also fix a potential bug on webrtc for Android:
Since the inline function won't be expanded on arm Android,
TickTime::MillisecondTimestamp and Clock::GetRealTimeClock()->TimeInMilliseconds
will be treated as function call, due to macro WEBRTC_CLOCK_TYPE_REALTIME's
guard defined in system_wrappers module they will get current time using
CLOCK_REALTIME.
But on x86 Android, the inline function will be expanded to where it's been
called, if the call happens in other compilation units which don't have
WEBRTC_CLOCK_TYPE_REALTIME definition, it will get current time using
CLOCK_MONOTONIC, while Clock::GetRealTimeClock()->TimeInMilliseconds will always
use CLOCK_REALTIME, then there will be two types of time in x86 Android which
will cause some weird issues like all received remote streams will be dropped
due to future render timestamp.
BUG=None
TEST=WebRTCViEDemo application works well on both arm and x86 Android
R=fischman@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1688004
Patch from Jeremy Mao <yujie.mao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4274 4adac7df-926f-26a2-2b94-8c16560cd09d
A new test target named 'modules_integrationtests' is created
and the following test targets were merged into it:
* audio_coding_module_test
* test_fec
* video_coding_integrationtests
* vp8_integrationtests
A couple of other targets were merged into modules_unittests:
* audio_coding_unittests
* audioproc_unittest
* common_unittests
* video_coding_unittests
* video_processing_unittests
* vp8_unittests
I wasn't able to merge audio_decoder_unittests and neteq_unittests due to
conflicts with different defines in these tests.
Some tests that have special requirements aren't merged into
modules_integrationtests yet. I took the opportunity to rename them
since the bot configs will need to be update anyway:
* audio_device_test_api -> audio_device_integrationtests
* video_capture_module_test -> video_capture_integrationtests
* video_render_module_test -> video_render_integrationtests
Exclude files were added for modules_integrationtests to make sure
the memcheck and tsan bots doesn't tests that are too slow
(audio_coding_module_test and vp8_integrationtests were previously
disabled on those bots).
Suppressions for AudioCodingModuleTest needed to be added to get
modules_integrationtests to pass memcheck (even if the test is
excluded from execution).
BUG=1843
TEST=local execution on Linux and trybots (passing except the merged tests of course)
R=andrew@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1656004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4228 4adac7df-926f-26a2-2b94-8c16560cd09d
Note that this means that there is no new code. The code has been taken directly from condition_variable_win.cc/h compensating minimally to be able to split up the two code paths.
Tested by:
1) Disabling native implementation and send to try bots.
2) Only return native implementation (i.e. if native implementation returns NULL there will be a crash when using the condition variable) and send to try bots.
3) The final cl sent to trybots.
All tests pass.
The changes are due to static analyzer code complaints.
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/1191004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3728 4adac7df-926f-26a2-2b94-8c16560cd09d
This is the first in a series of CLs to bring arbitrary resampling to webrtc.
* Replace Chromium-specific helpers with their respective webrtc versions.
* Add a second constructor to permit runtime selection of block_size.
* Add stringize_macros to system_wrappers.
BUG=webrtc:1395
TESTED=unit tests
Review URL: https://webrtc-codereview.appspot.com/1097012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3518 4adac7df-926f-26a2-2b94-8c16560cd09d
pthread_t is a pointer type on Mac OS X, and is thus 32 bits wide in the
32-bit environment and 64 bits wide in the 64-bit environment. WebRTC's
thread ID routines assume that thread IDs can always fit inside a uint32_t,
but this is not the case in the 64-bit Mac environment when using pthread_t
as the basis for a thread ID. Instead, switch to using the underlying Mach
port for the thread, which is a 32-bit quantity in both the 32-bit and 64-bit
environments.
The only place this seems to be used is in TraceImpl::AddThreadId, and it's
only used there for a thread ID for display.
This is a better fix than https://webrtc-codereview.appspot.com/929015 .
Review URL: https://webrtc-codereview.appspot.com/1063005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3427 4adac7df-926f-26a2-2b94-8c16560cd09d
The goal with this new clock interface is to have something which is used all
over WebRTC to make it easier to switch clock implementation depending on where
the components are used. This is a first step in that direction.
Next steps will be to, step by step, move all modules, video engine and voice
engine over to the new interface, effectively deprecating the old clock
interfaces. Long-term my vision is that we should be able to deprecate the clock
of WebRTC and rely on the user providing the implementation.
TEST=vie_auto_test, rtp_rtcp_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1041004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
atomicops.h are not necessary in trace_event.h similar to the port in WebKit.
It will cause a benign race condition detected by TSAN. If it shows up in
TSAN we will either suppress it or annotate it with dynamic annotations.
BUG=1215
Review URL: https://webrtc-codereview.appspot.com/982004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3267 4adac7df-926f-26a2-2b94-8c16560cd09d
