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Remove usage of Atomic32Wrapper from a few places.

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In these places, it doesn't make much sense to use an atomic variable we were using
Atomic32Wrapper::operator= anyway (which does not use atomic operations).
Review URL: https://webrtc-codereview.appspot.com/492005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2042 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tommi@webrtc.org 2012-04-17 13:33:28 +00:00
parent 52c0fec34c
commit 7ab51497a7
6 changed files with 39 additions and 71 deletions
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6
src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
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@ -52,7 +52,7 @@ MixHistory::~MixHistory()
WebRtc_Word32 MixHistory::IsMixed(bool& mixed) const
{
mixed = (_isMixed.Value() == 1);
mixed = _isMixed;
return 0;
}
@ -65,13 +65,13 @@ WebRtc_Word32 MixHistory::WasMixed(bool& wasMixed) const
WebRtc_Word32 MixHistory::SetIsMixed(const bool mixed)
{
_isMixed = mixed ? 1 : 0;
_isMixed = mixed;
return 0;
}
void MixHistory::ResetMixedStatus()
{
_isMixed = 0;
_isMixed = false;
}
AudioConferenceMixer* AudioConferenceMixer::Create(int id)

5
src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
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@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@ -11,7 +11,6 @@
#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
#include "atomic32_wrapper.h"
#include "audio_conference_mixer.h"
#include "engine_configurations.h"
#include "level_indicator.h"
@ -44,7 +43,7 @@ public:
void ResetMixedStatus();
private:
Atomic32Wrapper _isMixed; // 0 = false, 1 = true
bool _isMixed;
};
class AudioConferenceMixerImpl : public AudioConferenceMixer

39
src/modules/audio_device/main/source/win/audio_device_core_win.cc
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@ -2976,11 +2976,11 @@ WebRtc_Word32 AudioDeviceWindowsCore::StopRecording()
}
}
_UnLock();
// Reset the recording delay value.
_sndCardRecDelay = 0;
_UnLock();
return err;
}
@ -3148,10 +3148,10 @@ WebRtc_Word32 AudioDeviceWindowsCore::StopPlayout()
"Recording should be stopped before playout when using the "
"built-in AEC");
}
} // critScoped
// Reset the playout delay value.
_sndCardPlayDelay = 0;
// Reset the playout delay value.
_sndCardPlayDelay = 0;
} // critScoped
return 0;
}
@ -3162,7 +3162,8 @@ WebRtc_Word32 AudioDeviceWindowsCore::StopPlayout()
WebRtc_Word32 AudioDeviceWindowsCore::PlayoutDelay(WebRtc_UWord16& delayMS) const
{
delayMS = static_cast<WebRtc_UWord16>(_sndCardPlayDelay.Value());
CriticalSectionScoped critScoped(&_critSect);
delayMS = static_cast<WebRtc_UWord16>(_sndCardPlayDelay);
return 0;
}
@ -3172,7 +3173,8 @@ WebRtc_Word32 AudioDeviceWindowsCore::PlayoutDelay(WebRtc_UWord16& delayMS) cons
WebRtc_Word32 AudioDeviceWindowsCore::RecordingDelay(WebRtc_UWord16& delayMS) const
{
delayMS = static_cast<WebRtc_UWord16>(_sndCardRecDelay.Value());
CriticalSectionScoped critScoped(&_critSect);
delayMS = static_cast<WebRtc_UWord16>(_sndCardRecDelay);
return 0;
}
@ -3209,21 +3211,18 @@ WebRtc_Word32 AudioDeviceWindowsCore::SetPlayoutBuffer(const AudioDeviceModule::
WebRtc_Word32 AudioDeviceWindowsCore::PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const
{
{
CriticalSectionScoped lock(&_critSect);
type = _playBufType;
}
if (type == AudioDeviceModule::kFixedBufferSize)
{
CriticalSectionScoped lock(&_critSect);
sizeMS = _playBufDelayFixed;
}
else
{
// Use same value as for PlayoutDelay
sizeMS = static_cast<WebRtc_UWord16>(_sndCardPlayDelay.Value());
}
if (type == AudioDeviceModule::kFixedBufferSize)
{
sizeMS = _playBufDelayFixed;
}
else
{
// Use same value as for PlayoutDelay
sizeMS = static_cast<WebRtc_UWord16>(_sndCardPlayDelay);
}
return 0;
}
@ -4068,7 +4067,7 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread()
(((((UINT64)t1.QuadPart * _perfCounterFactor) - recTime)
/ 10000) + (10*syncBufIndex) / _recBlockSize - 10);
WebRtc_UWord32 sndCardPlayDelay =
static_cast<WebRtc_UWord32>(_sndCardPlayDelay.Value());
static_cast<WebRtc_UWord32>(_sndCardPlayDelay);
_sndCardRecDelay = sndCardRecDelay;

5
src/modules/audio_device/main/source/win/audio_device_core_win.h
View File

@ -25,7 +25,6 @@
#include <mediaobj.h> // IMediaObject
#include <Mmdeviceapi.h> // MMDevice
#include "atomic32_wrapper.h"
#include "critical_section_wrapper.h"
#include "scoped_refptr.h"
@ -326,7 +325,7 @@ private: // WASAPI
WebRtc_UWord32 _playBlockSize;
WebRtc_UWord32 _devicePlayBlockSize;
WebRtc_UWord32 _playChannels;
Atomic32Wrapper _sndCardPlayDelay;
WebRtc_UWord32 _sndCardPlayDelay;
UINT64 _writtenSamples;
LONGLONG _playAcc;
@ -335,7 +334,7 @@ private: // WASAPI
WebRtc_UWord32 _recBlockSize;
WebRtc_UWord32 _recChannels;
UINT64 _readSamples;
Atomic32Wrapper _sndCardRecDelay;
WebRtc_UWord32 _sndCardRecDelay;
float _sampleDriftAt48kHz;
float _driftAccumulator;

51
src/modules/udp_transport/source/udp_socket2_windows.cc
View File

@ -47,7 +47,7 @@ UdpSocket2Windows::UdpSocket2Windows(const WebRtc_Word32 id,
_terminate(false),
_addedToMgr(false),
_safeTodelete(false),
_outstandingCallsDisabled(0),
_outstandingCallsDisabled(false),
_clientHandle(NULL),
_flowHandle(NULL),
_filterHandle(NULL),
@ -449,10 +449,10 @@ void UdpSocket2Windows::IOCompleted(PerIoContext* pIOContext,
!pIOContext->ioInitiatedByThreadWrapper &&
(error == ERROR_OPERATION_ABORTED) &&
(pIOContext->ioOperation == OP_READ) &&
_outstandingCallsDisabled.Value() == 1)
_outstandingCallsDisabled)
{
// !pIOContext->initiatedIOByThreadWrapper indicate that the I/O
// was not initiaded by a ThreadWrapper thread.
// was not initiated by a ThreadWrapper thread.
// This may happen if the thread that initiated receiving (e.g.
// by calling StartListen())) is deleted before any packets have
// been received.
@ -462,39 +462,10 @@ void UdpSocket2Windows::IOCompleted(PerIoContext* pIOContext,
// that is controlled by the socket implementation.
// Note 2: This is more likely to happen to RTCP packets as
// they are less frequent than RTP packets.
// Note 3: _outstandingCallsDisabled being false (= 1) indicates
// Note 3: _outstandingCallsDisabled being false indicates
// that the socket isn't being shut down.
// Note 4: This should only happen buffers set to recevie packets
// Note 4: This should only happen buffers set to receive packets
// (OP_READ).
if (_outstandingCallsDisabled.Value() != 1)
{
WEBRTC_TRACE(
kTraceDebug,
kTraceTransport,
_id,
"UdpSocket2Windows::IOCompleted(pIOContext=%p,\
ioSize=%.lu, error=%.lu) Received operation aborted but continuing since\
pIOContext->ioInitiatedByThreadWrapper == false",
pIOContext,
ioSize,
error);
WebRtc_Word32 ioOp = pIOContext ?
(WebRtc_Word32)pIOContext->ioOperation : -1;
WebRtc_Word32 ioInit = pIOContext ?
(WebRtc_Word32)pIOContext->ioInitiatedByThreadWrapper : -1;
WEBRTC_TRACE(
kTraceDebug,
kTraceTransport,
_id,
"pIOContext->ioOperation=%d,\
pIOContext->ioInitiatedByThreadWrapper=%d, _outstandingCallsDisabled=%d,\
_incomingCb=%p, this=%p",
ioOp,
ioInit,
(WebRtc_Word32)_outstandingCallsDisabled.Value(),
_incomingCb,
this);
}
} else {
if(pIOContext == NULL)
{
@ -1300,7 +1271,7 @@ WebRtc_Word32 UdpSocket2Windows::CreateFlowSpec(WebRtc_Word32 serviceType,
bool UdpSocket2Windows::NewOutstandingCall()
{
assert(_outstandingCallsDisabled.Value() == 0);
assert(!_outstandingCallsDisabled);
++_outstandingCalls;
return true;
@ -1310,9 +1281,9 @@ void UdpSocket2Windows::OutstandingCallCompleted()
{
_ptrDestRWLock->AcquireLockShared();
++_outstandingCallComplete;
if((--_outstandingCalls == 0) && (_outstandingCallsDisabled.Value() == 1))
if((--_outstandingCalls == 0) && _outstandingCallsDisabled)
{
// When there are no outstanding calls and new outstandning calls are
// When there are no outstanding calls and new outstanding calls are
// disabled it is time to terminate.
_terminate = true;
}
@ -1332,13 +1303,13 @@ void UdpSocket2Windows::OutstandingCallCompleted()
void UdpSocket2Windows::DisableNewOutstandingCalls()
{
_ptrDestRWLock->AcquireLockExclusive();
if(_outstandingCallsDisabled.Value() == 1)
if(_outstandingCallsDisabled)
{
// Outstandning calls are already disabled.
_ptrDestRWLock->ReleaseLockExclusive();
return;
}
_outstandingCallsDisabled = 1;
_outstandingCallsDisabled = true;
const bool noOutstandingCalls = (_outstandingCalls.Value() == 0);
_ptrDestRWLock->ReleaseLockExclusive();
@ -1364,7 +1335,7 @@ void UdpSocket2Windows::WaitForOutstandingCalls()
void UdpSocket2Windows::RemoveSocketFromManager()
{
// New outstanding calls should be disabled at this point.
assert(_outstandingCallsDisabled.Value() != 0);
assert(_outstandingCallsDisabled);
if(_addedToMgr)
{

4
src/modules/udp_transport/source/udp_socket2_windows.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@ -139,7 +139,7 @@ private:
bool _safeTodelete;
RWLockWrapper* _ptrDestRWLock;
Atomic32Wrapper _outstandingCallsDisabled; // 0 = false, 1 = true
bool _outstandingCallsDisabled;
bool NewOutstandingCall();
void OutstandingCallCompleted();
void DisableNewOutstandingCalls();