generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Removed usage of the deprecated critical section constructor in audio_conference_mixer.

Browse Source 
Review URL: http://webrtc-codereview.appspot.com/320007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1226 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrike@webrtc.org 2011-12-16 23:00:30 +00:00
parent 8a44259ea8
commit cf5bcd1fd2
3 changed files with 28 additions and 28 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

44
src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
View File

@ -230,7 +230,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::TimeUntilNextProcess()
WEBRTC_TRACE(kTraceModuleCall, kTraceAudioMixerServer, _id,
"TimeUntilNextProcess()");
WebRtc_Word32 timeUntilNextProcess = 0;
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit.get());
if(_timeScheduler.TimeToNextUpdate(timeUntilNextProcess) != 0)
{
WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
@ -247,7 +247,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::Process()
WebRtc_UWord32 remainingParticipantsAllowedToMix =
kMaximumAmountOfMixedParticipants;
{
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit.get());
assert(_processCalls == 0);
_processCalls++;
@ -260,7 +260,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::Process()
ListWrapper additionalFramesList;
MapWrapper mixedParticipantsMap;
{
CriticalSectionScoped cs(*_cbCrit);
CriticalSectionScoped cs(_cbCrit.get());
WebRtc_Word32 lowFreq = GetLowestMixingFrequency();
// SILK can run in 12 kHz and 24 kHz. These frequencies are not
@ -276,7 +276,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::Process()
}
if(lowFreq <= 0)
{
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit.get());
_processCalls--;
return 0;
} else {
@ -303,7 +303,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::Process()
default:
assert(false);
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit.get());
_processCalls--;
return -1;
}
@ -348,7 +348,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::Process()
// TODO(henrike): it might be better to decide the number of channels
// with an API instead of dynamically.
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit.get());
if (!SetNumLimiterChannels(numberOfChannels))
retval = -1;
@ -391,7 +391,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::Process()
}
{
CriticalSectionScoped cs(*_cbCrit);
CriticalSectionScoped cs(_cbCrit.get());
if(_mixReceiver != NULL)
{
const AudioFrame** dummy = NULL;
@ -424,7 +424,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::Process()
ClearAudioFrameList(rampOutList);
ClearAudioFrameList(additionalFramesList);
{
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit.get());
_processCalls--;
}
return retval;
@ -435,7 +435,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::RegisterMixedStreamCallback(
{
WEBRTC_TRACE(kTraceModuleCall, kTraceAudioMixerServer, _id,
"RegisterMixedStreamCallback(mixReceiver)");
CriticalSectionScoped cs(*_cbCrit);
CriticalSectionScoped cs(_cbCrit.get());
if(_mixReceiver != NULL)
{
return -1;
@ -448,7 +448,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::UnRegisterMixedStreamCallback()
{
WEBRTC_TRACE(kTraceModuleCall, kTraceAudioMixerServer, _id,
"UnRegisterMixedStreamCallback()");
CriticalSectionScoped cs(*_cbCrit);
CriticalSectionScoped cs(_cbCrit.get());
if(_mixReceiver == NULL)
{
return -1;
@ -460,7 +460,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::UnRegisterMixedStreamCallback()
WebRtc_Word32 AudioConferenceMixerImpl::SetOutputFrequency(
const Frequency frequency)
{
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit.get());
const int error = _limiter->set_sample_rate_hz(frequency);
if(error != _limiter->kNoError)
{
@ -478,7 +478,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::SetOutputFrequency(
AudioConferenceMixer::Frequency
AudioConferenceMixerImpl::OutputFrequency() const
{
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit.get());
return _outputFrequency;
}
@ -520,7 +520,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::RegisterMixerStatusCallback(
return -1;
}
{
CriticalSectionScoped cs(*_cbCrit);
CriticalSectionScoped cs(_cbCrit.get());
if(_mixerStatusCallback != NULL)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
@ -530,7 +530,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::RegisterMixerStatusCallback(
_mixerStatusCallback = &mixerStatusCallback;
}
{
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit.get());
_amountOf10MsBetweenCallbacks = amountOf10MsBetweenCallbacks;
_amountOf10MsUntilNextCallback = 0;
_mixerStatusCb = true;
@ -543,7 +543,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::UnRegisterMixerStatusCallback()
WEBRTC_TRACE(kTraceModuleCall, kTraceAudioMixerServer, _id,
"UnRegisterMixerStatusCallback()");
{
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit.get());
if(!_mixerStatusCb)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
@ -553,7 +553,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::UnRegisterMixerStatusCallback()
_mixerStatusCb = false;
}
{
CriticalSectionScoped cs(*_cbCrit);
CriticalSectionScoped cs(_cbCrit.get());
_mixerStatusCallback = NULL;
}
return 0;
@ -574,7 +574,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::SetMixabilityStatus(
}
WebRtc_UWord32 amountOfMixableParticipants;
{
CriticalSectionScoped cs(*_cbCrit);
CriticalSectionScoped cs(_cbCrit.get());
const bool isMixed =
IsParticipantInList(participant,_participantList);
// API must be called with a new state.
@ -607,7 +607,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::SetMixabilityStatus(
// A MixerParticipant was added or removed. Make sure the scratch
// buffer is updated if necessary.
// Note: The scratch buffer may only be updated in Process().
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit.get());
_amountOfMixableParticipants = amountOfMixableParticipants;
return 0;
}
@ -618,7 +618,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::MixabilityStatus(
{
WEBRTC_TRACE(kTraceModuleCall, kTraceAudioMixerServer, _id,
"MixabilityStatus(participant,mixable)");
CriticalSectionScoped cs(*_cbCrit);
CriticalSectionScoped cs(_cbCrit.get());
mixable = IsParticipantInList(participant, _participantList);
return 0;
}
@ -628,7 +628,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::AmountOfMixables(
{
WEBRTC_TRACE(kTraceModuleCall, kTraceAudioMixerServer, _id,
"AmountOfMixables(amountOfMixableParticipants)");
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit.get());
amountOfMixableParticipants = _amountOfMixableParticipants;
return 0;
}
@ -639,7 +639,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::SetAnonymousMixabilityStatus(
WEBRTC_TRACE(kTraceModuleCall, kTraceAudioMixerServer, _id,
"SetAnonymousMixabilityStatus(participant,anonymous:%s)",
anonymous ? "true" : "false");
CriticalSectionScoped cs(*_cbCrit);
CriticalSectionScoped cs(_cbCrit.get());
if(IsParticipantInList(participant, _additionalParticipantList))
{
if(anonymous)
@ -681,7 +681,7 @@ WebRtc_Word32 AudioConferenceMixerImpl::AnonymousMixabilityStatus(
{
WEBRTC_TRACE(kTraceModuleCall, kTraceAudioMixerServer, _id,
"AnonymousMixabilityStatus(participant,mixable)");
CriticalSectionScoped cs(*_cbCrit);
CriticalSectionScoped cs(_cbCrit.get());
mixable = IsParticipantInList(participant,
_additionalParticipantList);
return 0;

8
src/modules/audio_conference_mixer/source/memory_pool_posix.h
View File

@ -70,7 +70,7 @@ MemoryPoolImpl<MemoryType>::~MemoryPoolImpl()
template<class MemoryType>
WebRtc_Word32 MemoryPoolImpl<MemoryType>::PopMemory(MemoryType*& memory)
{
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit);
if(_terminate)
{
memory = NULL;
@ -101,7 +101,7 @@ WebRtc_Word32 MemoryPoolImpl<MemoryType>::PushMemory(MemoryType*& memory)
{
return -1;
}
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit);
_outstandingMemory--;
if(_memoryPool.GetSize() > (_initialPoolSize << 1))
{
@ -119,14 +119,14 @@ WebRtc_Word32 MemoryPoolImpl<MemoryType>::PushMemory(MemoryType*& memory)
template<class MemoryType>
bool MemoryPoolImpl<MemoryType>::Initialize()
{
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit);
return CreateMemory(_initialPoolSize) == 0;
}
template<class MemoryType>
WebRtc_Word32 MemoryPoolImpl<MemoryType>::Terminate()
{
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit);
assert(_createdMemory == _outstandingMemory + _memoryPool.GetSize());
_terminate = true;

4
src/modules/audio_conference_mixer/source/time_scheduler.cc
View File

@ -29,7 +29,7 @@ TimeScheduler::~TimeScheduler()
WebRtc_Word32 TimeScheduler::UpdateScheduler()
{
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit);
if(!_isStarted)
{
_isStarted = true;
@ -79,7 +79,7 @@ WebRtc_Word32 TimeScheduler::UpdateScheduler()
WebRtc_Word32 TimeScheduler::TimeToNextUpdate(
WebRtc_Word32& updateTimeInMS) const
{
CriticalSectionScoped cs(*_crit);
CriticalSectionScoped cs(_crit);
// Missed periods means that the next UpdateScheduler() should happen
// immediately.
if(_missedPeriods > 0)