This will enable some low-level webrtc logging in a Chromium build,
while limiting the binary size impact.
For a Mac Release build, it results in an increase to Chrome.app of 37k
and libpeerconnection.so of 25k. For comparison, enabling full logs
costs 230k and 218k respectively.
BUG=b/11470432
TESTED=voe_cmd_test produces logs of the appropriate severity.
R=fischman@webrtc.org, henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5097 4adac7df-926f-26a2-2b94-8c16560cd09d
Analogous to the recent libjingle change: http://cl/54929753-p10.
This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather
than scoped_array and scoped_ptr_malloc respectively.
- Add Chromium's template-based COMPILE_ASSERT. We didn't have this
previously in order to support the macro in C. Instead, move the
existing macro to compile_assert_c.h.
- Additionally copy the move.h and template_util.h depedencies and add
the WARN_UNUSED_RESULT macro.
- Leave scoped_array and scoped_ptr_malloc for now, but mark as
deprecated.
- Remove scoped_ptr foo(NULL) use. The default constructor handles it.
- Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc.
- Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove
some repeated code.
TESTED=trybots
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2449005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.
Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort
Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled: 666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled: 673 ms (1.01x)
BUG=2314
R=henrik.lundin@webrtc.org, henrike@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
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
* The old resampler was found to have a wraparound bug.
* Remove support for the old resampler from PushResampler.
* Use PushResampler in AudioCodingModule.
* The old resampler must still be removed from the file utility.
BUG=webrtc:1867,webrtc:827
TESTED=unit tests, Chrome using apprtc and voe_cmd_test to verify wrap-around is corrected, voe_cmd_test running through all supported codec sample rates and channels to verify good quality audio
R=henrika@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1590004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4156 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
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
