5 Commits

Author SHA1 Message Date
andrew@webrtc.org
9080518a39 Restore severity precondition to logging.h.
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
2013-09-05 16:40:43 +00:00
andrew@webrtc.org
c1eb560a5c Replace the old resampler with SincResampler in the voice engine signal path.
* 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
2013-06-03 19:00:29 +00:00
andrew@webrtc.org
c3e5d3422b Add a logging_no_op.cc when enable_tracing==0.
This should hopefully fix static initializer warnings when rolling webrtc
in Chromium.

TEST=logging_unittest succeeds with enable_tracing==1 and fails appropriately with enable_tracing==0.

Review URL: https://webrtc-codereview.appspot.com/939026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3159 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-23 19:30:59 +00:00
andrew@webrtc.org
655d8f56f6 Add a kTraceTerseInfo level for non-verbose logging.
Review URL: https://webrtc-codereview.appspot.com/937023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3134 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-20 07:34:45 +00:00
andrew@webrtc.org
50419b0777 Add libjingle-style stream-style logging.
Add a highly stripped-down version of libjingle's base/logging.h. It is
a thin wrapper around WEBRTC_TRACE, maintaining the libjingle log
semantics to ease a transition to that format.

Also add some helper macros for easy API and function failure logging.

Review URL: https://webrtc-codereview.appspot.com/931010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3099 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-14 19:07:54 +00:00