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
This commit is contained in:
andrew@webrtc.org
2013-09-05 16:40:43 +00:00
parent 95e51f509c
commit 9080518a39
21 changed files with 50 additions and 55 deletions

View File

@ -33,7 +33,7 @@ int DecodeFromStorageTest(const CmdArgs& args) {
"decodeFromStorageTestTrace.txt";
webrtc::Trace::CreateTrace();
webrtc::Trace::SetTraceFile(trace_file.c_str());
webrtc::Trace::SetLevelFilter(webrtc::kTraceAll);
webrtc::Trace::set_level_filter(webrtc::kTraceAll);
webrtc::rtpplayer::PayloadTypes payload_types;
payload_types.push_back(webrtc::rtpplayer::PayloadCodecTuple(

View File

@ -34,7 +34,7 @@ int GenericCodecTest::RunTest(CmdArgs& args)
Trace::CreateTrace();
Trace::SetTraceFile(
(test::OutputPath() + "genericCodecTestTrace.txt").c_str());
Trace::SetLevelFilter(webrtc::kTraceAll);
Trace::set_level_filter(webrtc::kTraceAll);
get->Perform(args);
Trace::ReturnTrace();
delete get;

View File

@ -31,7 +31,7 @@ int MediaOptTest::RunTest(int testNum, CmdArgs& args)
{
Trace::CreateTrace();
Trace::SetTraceFile((test::OutputPath() + "mediaOptTestTrace.txt").c_str());
Trace::SetLevelFilter(webrtc::kTraceAll);
Trace::set_level_filter(webrtc::kTraceAll);
VideoCodingModule* vcm = VideoCodingModule::Create(1);
Clock* clock = Clock::GetRealTimeClock();
MediaOptTest* mot = new MediaOptTest(vcm, clock);

View File

@ -129,7 +129,7 @@ int MTRxTxTest(CmdArgs& args)
// Set up trace
Trace::CreateTrace();
Trace::SetTraceFile((test::OutputPath() + "MTRxTxTestTrace.txt").c_str());
Trace::SetLevelFilter(webrtc::kTraceAll);
Trace::set_level_filter(webrtc::kTraceAll);
FILE* sourceFile;
FILE* decodedFile;

View File

@ -36,7 +36,7 @@ int NormalTest::RunTest(const CmdArgs& args)
Trace::CreateTrace();
Trace::SetTraceFile(
(test::OutputPath() + "VCMNormalTestTrace.txt").c_str());
Trace::SetLevelFilter(webrtc::kTraceAll);
Trace::set_level_filter(webrtc::kTraceAll);
VideoCodingModule* vcm = VideoCodingModule::Create(1, clock,
&event_factory);
NormalTest VCMNTest(vcm, clock);

View File

@ -31,7 +31,7 @@ int RtpPlay(const CmdArgs& args) {
std::string trace_file = webrtc::test::OutputPath() + "receiverTestTrace.txt";
webrtc::Trace::CreateTrace();
webrtc::Trace::SetTraceFile(trace_file.c_str());
webrtc::Trace::SetLevelFilter(webrtc::kTraceAll);
webrtc::Trace::set_level_filter(webrtc::kTraceAll);
webrtc::rtpplayer::PayloadTypes payload_types;
payload_types.push_back(webrtc::rtpplayer::PayloadCodecTuple(

View File

@ -67,7 +67,7 @@ int RtpPlayMT(const CmdArgs& args) {
std::string trace_file = webrtc::test::OutputPath() + "receiverTestTrace.txt";
webrtc::Trace::CreateTrace();
webrtc::Trace::SetTraceFile(trace_file.c_str());
webrtc::Trace::SetLevelFilter(webrtc::kTraceAll);
webrtc::Trace::set_level_filter(webrtc::kTraceAll);
webrtc::rtpplayer::PayloadTypes payload_types;
payload_types.push_back(webrtc::rtpplayer::PayloadCodecTuple(