Index: webrtc/modules/audio_conference_mixer/source/time_scheduler.cc |
diff --git a/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc b/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc |
index 30b2933b61c4d3485233a85f2931230401060cd1..877d98b053306cb65287b2a1299a3951f01742de 100644 |
--- a/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc |
+++ b/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc |
@@ -10,27 +10,17 @@ |
#include "webrtc/base/timeutils.h" |
#include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h" |
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
namespace webrtc { |
TimeScheduler::TimeScheduler(const int64_t periodicityInMs) |
- : _crit(CriticalSectionWrapper::CreateCriticalSection()), |
- _isStarted(false), |
+ : _isStarted(false), |
_lastPeriodMark(), |
_periodicityInMs(periodicityInMs), |
_periodicityInTicks(periodicityInMs * rtc::kNumNanosecsPerMillisec), |
- _missedPeriods(0) |
- { |
- } |
+ _missedPeriods(0) {} |
-TimeScheduler::~TimeScheduler() |
-{ |
- delete _crit; |
-} |
- |
-int32_t TimeScheduler::UpdateScheduler() |
-{ |
- CriticalSectionScoped cs(_crit); |
+int32_t TimeScheduler::UpdateScheduler() { |
+ rtc::CritScope cs(&_crit); |
if(!_isStarted) |
{ |
_isStarted = true; |
@@ -79,7 +69,7 @@ int32_t TimeScheduler::UpdateScheduler() |
int32_t TimeScheduler::TimeToNextUpdate( |
int64_t& updateTimeInMS) const |
{ |
- CriticalSectionScoped cs(_crit); |
+ rtc::CritScope cs(&_crit); |
// Missed periods means that the next UpdateScheduler() should happen |
// immediately. |
if(_missedPeriods > 0) |