Removed usage of the deprecated critical section constructor in rtp_rtcp.
Review URL: http://webrtc-codereview.appspot.com/315004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1173 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/modules/rtp_rtcp/source/bandwidth_management.cc b/src/modules/rtp_rtcp/source/bandwidth_management.cc
index aa279e9..9d5cf68 100644
--- a/src/modules/rtp_rtcp/source/bandwidth_management.cc
+++ b/src/modules/rtp_rtcp/source/bandwidth_management.cc
@@ -19,7 +19,7 @@
BandwidthManagement::BandwidthManagement(const WebRtc_Word32 id) :
_id(id),
- _critsect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _critsect(CriticalSectionWrapper::CreateCriticalSection()),
_lastPacketLossExtendedHighSeqNum(0),
_lastReportAllLost(false),
_lastLoss(0),
@@ -39,7 +39,7 @@
BandwidthManagement::~BandwidthManagement()
{
- delete &_critsect;
+ delete _critsect;
}
WebRtc_Word32
diff --git a/src/modules/rtp_rtcp/source/bandwidth_management.h b/src/modules/rtp_rtcp/source/bandwidth_management.h
index aeb32b6..6761820 100644
--- a/src/modules/rtp_rtcp/source/bandwidth_management.h
+++ b/src/modules/rtp_rtcp/source/bandwidth_management.h
@@ -64,7 +64,7 @@
WebRtc_Word32 _id;
- CriticalSectionWrapper& _critsect;
+ CriticalSectionWrapper* _critsect;
// incoming filters
WebRtc_UWord32 _lastPacketLossExtendedHighSeqNum;
diff --git a/src/modules/rtp_rtcp/source/dtmf_queue.cc b/src/modules/rtp_rtcp/source/dtmf_queue.cc
index 72cd735..749309b 100644
--- a/src/modules/rtp_rtcp/source/dtmf_queue.cc
+++ b/src/modules/rtp_rtcp/source/dtmf_queue.cc
@@ -14,7 +14,7 @@
namespace webrtc {
DTMFqueue::DTMFqueue():
- _DTMFCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _DTMFCritsect(CriticalSectionWrapper::CreateCriticalSection()),
_nextEmptyIndex(0)
{
memset(_DTMFKey,0, sizeof(_DTMFKey));
@@ -24,7 +24,7 @@
DTMFqueue::~DTMFqueue()
{
- delete &_DTMFCritsect;
+ delete _DTMFCritsect;
}
WebRtc_Word32
diff --git a/src/modules/rtp_rtcp/source/dtmf_queue.h b/src/modules/rtp_rtcp/source/dtmf_queue.h
index 67c47d0..8451a21 100644
--- a/src/modules/rtp_rtcp/source/dtmf_queue.h
+++ b/src/modules/rtp_rtcp/source/dtmf_queue.h
@@ -29,7 +29,7 @@
void ResetDTMF();
private:
- CriticalSectionWrapper& _DTMFCritsect;
+ CriticalSectionWrapper* _DTMFCritsect;
WebRtc_UWord8 _nextEmptyIndex;
WebRtc_UWord8 _DTMFKey[DTMF_OUTBAND_MAX];
WebRtc_UWord16 _DTMFLen[DTMF_OUTBAND_MAX];
diff --git a/src/modules/rtp_rtcp/source/rtcp_receiver.cc b/src/modules/rtp_rtcp/source/rtcp_receiver.cc
index dabfe71..9b76e46 100644
--- a/src/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/src/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -35,10 +35,11 @@
_method(kRtcpOff),
_lastReceived(0),
_rtpRtcp(*owner),
- _criticalSectionFeedbacks(*CriticalSectionWrapper::CreateCriticalSection()),
+ _criticalSectionFeedbacks(CriticalSectionWrapper::CreateCriticalSection()),
_cbRtcpFeedback(NULL),
_cbVideoFeedback(NULL),
- _criticalSectionRTCPReceiver(*CriticalSectionWrapper::CreateCriticalSection()),
+ _criticalSectionRTCPReceiver(
+ CriticalSectionWrapper::CreateCriticalSection()),
_SSRC(0),
_remoteSSRC(0),
_remoteSenderInfo(),
@@ -53,8 +54,8 @@
RTCPReceiver::~RTCPReceiver()
{
- delete &_criticalSectionRTCPReceiver;
- delete &_criticalSectionFeedbacks;
+ delete _criticalSectionRTCPReceiver;
+ delete _criticalSectionFeedbacks;
bool loop = true;
do
@@ -481,13 +482,13 @@
}
}
- _criticalSectionRTCPReceiver.Leave();
+ _criticalSectionRTCPReceiver->Leave();
// to avoid problem with accuireing _criticalSectionRTCPSender while holding _criticalSectionRTCPReceiver
WebRtc_UWord32 sendTimeMS =
_rtpRtcp.SendTimeOfSendReport(rtcpPacket.ReportBlockItem.LastSR);
- _criticalSectionRTCPReceiver.Enter();
+ _criticalSectionRTCPReceiver->Enter();
// ReportBlockItem.SSRC is who it's to
// we store all incoming reports, used in conference relay
diff --git a/src/modules/rtp_rtcp/source/rtcp_receiver.h b/src/modules/rtp_rtcp/source/rtcp_receiver.h
index cac5a69..5d8f143 100644
--- a/src/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/src/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -187,11 +187,11 @@
WebRtc_UWord32 _lastReceived;
ModuleRtpRtcpImpl& _rtpRtcp;
- CriticalSectionWrapper& _criticalSectionFeedbacks;
+ CriticalSectionWrapper* _criticalSectionFeedbacks;
RtcpFeedback* _cbRtcpFeedback;
RtpVideoFeedback* _cbVideoFeedback;
- CriticalSectionWrapper& _criticalSectionRTCPReceiver;
+ CriticalSectionWrapper* _criticalSectionRTCPReceiver;
WebRtc_UWord32 _SSRC;
WebRtc_UWord32 _remoteSSRC;
diff --git a/src/modules/rtp_rtcp/source/rtcp_sender.cc b/src/modules/rtp_rtcp/source/rtcp_sender.cc
index cf9885c..2ae88fc 100644
--- a/src/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/src/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -31,10 +31,10 @@
_clock(*clock),
_method(kRtcpOff),
_rtpRtcp(*owner),
- _criticalSectionTransport(*CriticalSectionWrapper::CreateCriticalSection()),
+ _criticalSectionTransport(CriticalSectionWrapper::CreateCriticalSection()),
_cbTransport(NULL),
- _criticalSectionRTCPSender(*CriticalSectionWrapper::CreateCriticalSection()),
+ _criticalSectionRTCPSender(CriticalSectionWrapper::CreateCriticalSection()),
_usingNack(false),
_sending(false),
_sendTMMBN(false),
@@ -112,8 +112,8 @@
_csrcCNAMEs.Erase(item);
item = _csrcCNAMEs.First();
}
- delete &_criticalSectionTransport;
- delete &_criticalSectionRTCPSender;
+ delete _criticalSectionTransport;
+ delete _criticalSectionRTCPSender;
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
}
diff --git a/src/modules/rtp_rtcp/source/rtcp_sender.h b/src/modules/rtp_rtcp/source/rtcp_sender.h
index e744afd..109d4b5 100644
--- a/src/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/src/modules/rtp_rtcp/source/rtcp_sender.h
@@ -178,10 +178,10 @@
ModuleRtpRtcpImpl& _rtpRtcp;
- CriticalSectionWrapper& _criticalSectionTransport;
+ CriticalSectionWrapper* _criticalSectionTransport;
Transport* _cbTransport;
- CriticalSectionWrapper& _criticalSectionRTCPSender;
+ CriticalSectionWrapper* _criticalSectionRTCPSender;
bool _usingNack;
bool _sending;
bool _sendTMMBN;
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver.cc b/src/modules/rtp_rtcp/source/rtp_receiver.cc
index 26fdb9b..570fa7e 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver.cc
+++ b/src/modules/rtp_rtcp/source/rtp_receiver.cc
@@ -31,11 +31,12 @@
_id(id),
_audio(audio),
_rtpRtcp(*owner),
- _criticalSectionCbs(*CriticalSectionWrapper::CreateCriticalSection()),
+ _criticalSectionCbs(CriticalSectionWrapper::CreateCriticalSection()),
_cbRtpFeedback(NULL),
_cbRtpData(NULL),
- _criticalSectionRTPReceiver(*CriticalSectionWrapper::CreateCriticalSection()),
+ _criticalSectionRTPReceiver(
+ CriticalSectionWrapper::CreateCriticalSection()),
_lastReceiveTime(0),
_lastReceivedPayloadLength(0),
_lastReceivedPayloadType(-1),
@@ -101,8 +102,8 @@
_cbRtpFeedback->OnIncomingCSRCChanged(_id,_currentRemoteCSRC[i], false);
}
}
- delete &_criticalSectionCbs;
- delete &_criticalSectionRTPReceiver;
+ delete _criticalSectionCbs;
+ delete _criticalSectionRTPReceiver;
// empty map
bool loop = true;
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver.h b/src/modules/rtp_rtcp/source/rtp_receiver.h
index 03a3b0d..a289fbf 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver.h
+++ b/src/modules/rtp_rtcp/source/rtp_receiver.h
@@ -174,11 +174,11 @@
const bool _audio;
ModuleRtpRtcpImpl& _rtpRtcp;
- CriticalSectionWrapper& _criticalSectionCbs;
+ CriticalSectionWrapper* _criticalSectionCbs;
RtpFeedback* _cbRtpFeedback;
RtpData* _cbRtpData;
- CriticalSectionWrapper& _criticalSectionRTPReceiver;
+ CriticalSectionWrapper* _criticalSectionRTPReceiver;
mutable WebRtc_UWord32 _lastReceiveTime;
WebRtc_UWord16 _lastReceivedPayloadLength;
WebRtc_Word8 _lastReceivedPayloadType;
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index 2458725..cb75c9e 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -31,14 +31,14 @@
_cngPayloadType(-1),
_G722PayloadType(-1),
_lastReceivedG722(false),
- _criticalSectionFeedback(*CriticalSectionWrapper::CreateCriticalSection()),
+ _criticalSectionFeedback(CriticalSectionWrapper::CreateCriticalSection()),
_cbAudioFeedback(NULL)
{
}
RTPReceiverAudio::~RTPReceiverAudio()
{
- delete &_criticalSectionFeedback;
+ delete _criticalSectionFeedback;
}
WebRtc_Word32
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver_audio.h b/src/modules/rtp_rtcp/source/rtp_receiver_audio.h
index 6fe9bf5..8a7172a 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -89,7 +89,7 @@
WebRtc_Word8 _G722PayloadType;
bool _lastReceivedG722;
- CriticalSectionWrapper& _criticalSectionFeedback;
+ CriticalSectionWrapper* _criticalSectionFeedback;
RtpAudioFeedback* _cbAudioFeedback;
};
} // namespace webrtc
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver_video.cc b/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
index a681760..051f752 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -28,10 +28,10 @@
ModuleRtpRtcpImpl* owner):
_id(id),
_rtpRtcp(*owner),
- _criticalSectionFeedback(*CriticalSectionWrapper::CreateCriticalSection()),
+ _criticalSectionFeedback(CriticalSectionWrapper::CreateCriticalSection()),
_cbVideoFeedback(NULL),
- _criticalSectionReceiverVideo(*CriticalSectionWrapper::CreateCriticalSection()),
-
+ _criticalSectionReceiverVideo(
+ CriticalSectionWrapper::CreateCriticalSection()),
_completeFrame(false),
_packetStartTimeMs(0),
_receivedBW(),
@@ -49,8 +49,8 @@
RTPReceiverVideo::~RTPReceiverVideo()
{
- delete &_criticalSectionFeedback;
- delete &_criticalSectionReceiverVideo;
+ delete _criticalSectionFeedback;
+ delete _criticalSectionReceiverVideo;
delete _receiveFEC;
}
@@ -228,7 +228,7 @@
{
WebRtc_Word32 retVal = 0;
- _criticalSectionReceiverVideo.Enter();
+ _criticalSectionReceiverVideo->Enter();
_videoBitRate.Update(payloadDataLength, nowMS);
@@ -242,7 +242,7 @@
{
if(_receiveFEC == NULL)
{
- _criticalSectionReceiverVideo.Leave();
+ _criticalSectionReceiverVideo->Leave();
return -1;
}
if (rtpHeader->header.timestamp != TimeStamp())
@@ -269,7 +269,7 @@
_receiveFEC->AddReceivedFECInfo(rtpHeader,incomingRtpPacket, FECpacket);
}
}
- _criticalSectionReceiverVideo.Leave();
+ _criticalSectionReceiverVideo->Leave();
if(retVal == 0 && FECpacket )
{
@@ -296,18 +296,18 @@
// Update the remote rate control object and update the overuse
// detector with the current rate control region.
- _criticalSectionReceiverVideo.Enter();
+ _criticalSectionReceiverVideo->Enter();
const RateControlInput input(_overUseDetector.State(),
_videoBitRate.BitRate(nowMS),
_overUseDetector.NoiseVar());
- _criticalSectionReceiverVideo.Leave();
+ _criticalSectionReceiverVideo->Leave();
// Call the callback outside critical section
const RateControlRegion region = _rtpRtcp.OnOverUseStateUpdate(input);
- _criticalSectionReceiverVideo.Enter();
+ _criticalSectionReceiverVideo->Enter();
_overUseDetector.SetRateControlRegion(region);
- _criticalSectionReceiverVideo.Leave();
+ _criticalSectionReceiverVideo->Leave();
return retVal;
}
@@ -356,7 +356,7 @@
const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadDataLength)
{
- _criticalSectionReceiverVideo.Enter();
+ _criticalSectionReceiverVideo->Enter();
_currentFecFrameDecoded = true;
@@ -450,7 +450,7 @@
retVal = ReceiveMPEG4Codec(rtpHeader,payloadData, payloadDataLength);
break;
default:
- _criticalSectionReceiverVideo.Leave();
+ _criticalSectionReceiverVideo->Leave();
assert(((void)"ParseCodecSpecific videoType can not be unknown here!", false));
return -1;
}
@@ -470,7 +470,7 @@
const bool success = rtpPayloadParser.Parse(parsedPacket);
// from here down we only work on local data
- _criticalSectionReceiverVideo.Leave();
+ _criticalSectionReceiverVideo->Leave();
if (!success)
{
@@ -498,17 +498,17 @@
const bool success = rtpPayloadParser.Parse(parsedPacket);
if (!success)
{
- _criticalSectionReceiverVideo.Leave();
+ _criticalSectionReceiverVideo->Leave();
return -1;
}
if (IP_PACKET_SIZE < parsedPacket.info.H263.dataLength +
(parsedPacket.info.H263.insert2byteStartCode ? 2 : 0))
{
- _criticalSectionReceiverVideo.Leave();
+ _criticalSectionReceiverVideo->Leave();
return -1;
}
// from here down we only work on local data
- _criticalSectionReceiverVideo.Leave();
+ _criticalSectionReceiverVideo->Leave();
return ReceiveH263CodecCommon(parsedPacket, rtpHeader);
}
@@ -590,11 +590,11 @@
const bool success = rtpPayloadParser.Parse(parsedPacket);
if (!success)
{
- _criticalSectionReceiverVideo.Leave();
+ _criticalSectionReceiverVideo->Leave();
return -1;
}
// from here down we only work on local data
- _criticalSectionReceiverVideo.Leave();
+ _criticalSectionReceiverVideo->Leave();
rtpHeader->frameType = (parsedPacket.frameType == ModuleRTPUtility::kIFrame) ? kVideoFrameKey : kVideoFrameDelta;
rtpHeader->type.Video.isFirstPacket = parsedPacket.info.MPEG4.isFirstPacket;
@@ -622,7 +622,7 @@
const bool success = rtpPayloadParser.Parse(parsedPacket);
// from here down we only work on local data
- _criticalSectionReceiverVideo.Leave();
+ _criticalSectionReceiverVideo->Leave();
if (!success)
{
@@ -681,7 +681,7 @@
{
rtpHeader->type.Video.isFirstPacket = true;
}
- _criticalSectionReceiverVideo.Leave();
+ _criticalSectionReceiverVideo->Leave();
if(CallbackOfReceivedPayloadData(payloadData, payloadDataLength, rtpHeader) != 0)
{
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver_video.h b/src/modules/rtp_rtcp/source/rtp_receiver_video.h
index 737699a..7893426 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -132,10 +132,10 @@
WebRtc_Word32 _id;
ModuleRtpRtcpImpl& _rtpRtcp;
- CriticalSectionWrapper& _criticalSectionFeedback;
+ CriticalSectionWrapper* _criticalSectionFeedback;
RtpVideoFeedback* _cbVideoFeedback;
- CriticalSectionWrapper& _criticalSectionReceiverVideo;
+ CriticalSectionWrapper* _criticalSectionReceiverVideo;
// bandwidth
bool _completeFrame;
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index a18f690..23d55b2 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -89,8 +89,9 @@
_lastProcessTime(clock->GetTimeInMS()),
_packetOverHead(28), // IPV4 UDP
- _criticalSectionModulePtrs(*CriticalSectionWrapper::CreateCriticalSection()),
- _criticalSectionModulePtrsFeedback(*CriticalSectionWrapper::CreateCriticalSection()),
+ _criticalSectionModulePtrs(CriticalSectionWrapper::CreateCriticalSection()),
+ _criticalSectionModulePtrsFeedback(
+ CriticalSectionWrapper::CreateCriticalSection()),
_defaultModule(NULL),
_audioModule(NULL),
_videoModule(NULL),
@@ -165,8 +166,8 @@
}
#endif
- delete &_criticalSectionModulePtrs;
- delete &_criticalSectionModulePtrsFeedback;
+ delete _criticalSectionModulePtrs;
+ delete _criticalSectionModulePtrsFeedback;
}
WebRtc_Word32
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index c0269cf..5da611a 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -544,8 +544,8 @@
WebRtc_UWord32 _lastProcessTime;
WebRtc_UWord16 _packetOverHead;
- CriticalSectionWrapper& _criticalSectionModulePtrs;
- CriticalSectionWrapper& _criticalSectionModulePtrsFeedback;
+ CriticalSectionWrapper* _criticalSectionModulePtrs;
+ CriticalSectionWrapper* _criticalSectionModulePtrsFeedback;
ModuleRtpRtcpImpl* _defaultModule;
ModuleRtpRtcpImpl* _audioModule;
ModuleRtpRtcpImpl* _videoModule;
diff --git a/src/modules/rtp_rtcp/source/rtp_sender.cc b/src/modules/rtp_rtcp/source/rtp_sender.cc
index 617faf7..2af0dbf 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender.cc
@@ -27,8 +27,8 @@
_audioConfigured(audio),
_audio(NULL),
_video(NULL),
- _sendCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
- _transportCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _sendCritsect(CriticalSectionWrapper::CreateCriticalSection()),
+ _transportCritsect(CriticalSectionWrapper::CreateCriticalSection()),
_transport(NULL),
@@ -48,7 +48,7 @@
_storeSentPackets(false),
_storeSentPacketsNumber(0),
- _prevSentPacketsCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _prevSentPacketsCritsect(CriticalSectionWrapper::CreateCriticalSection()),
_prevSentPacketsIndex(0),
_ptrPrevSentPackets(NULL),
_prevSentPacketsSeqNum(NULL),
@@ -107,9 +107,9 @@
_ssrcDB.ReturnSSRC(_ssrc);
SSRCDatabase::ReturnSSRCDatabase();
- delete &_prevSentPacketsCritsect;
- delete &_sendCritsect;
- delete &_transportCritsect;
+ delete _prevSentPacketsCritsect;
+ delete _sendCritsect;
+ delete _transportCritsect;
// empty map
bool loop = true;
diff --git a/src/modules/rtp_rtcp/source/rtp_sender.h b/src/modules/rtp_rtcp/source/rtp_sender.h
index b4d17a3..0d47926 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.h
+++ b/src/modules/rtp_rtcp/source/rtp_sender.h
@@ -267,9 +267,9 @@
RTPSenderAudio* _audio;
RTPSenderVideo* _video;
- CriticalSectionWrapper& _sendCritsect;
+ CriticalSectionWrapper* _sendCritsect;
- CriticalSectionWrapper& _transportCritsect;
+ CriticalSectionWrapper* _transportCritsect;
Transport* _transport;
bool _sendingMedia;
@@ -288,7 +288,7 @@
bool _storeSentPackets;
WebRtc_UWord16 _storeSentPacketsNumber;
- CriticalSectionWrapper& _prevSentPacketsCritsect;
+ CriticalSectionWrapper* _prevSentPacketsCritsect;
WebRtc_Word32 _prevSentPacketsIndex;
WebRtc_Word8** _ptrPrevSentPackets;
WebRtc_UWord16* _prevSentPacketsSeqNum;
diff --git a/src/modules/rtp_rtcp/source/rtp_sender_audio.cc b/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 5d690c1..854f394 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -19,9 +19,9 @@
_id(id),
_clock(*clock),
_rtpSender(rtpSender),
- _audioFeedbackCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _audioFeedbackCritsect(CriticalSectionWrapper::CreateCriticalSection()),
_audioFeedback(NULL),
- _sendAudioCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
_frequency(8000),
_packetSizeSamples(160),
_dtmfEventIsOn(false),
@@ -46,8 +46,8 @@
RTPSenderAudio::~RTPSenderAudio()
{
- delete &_sendAudioCritsect;
- delete &_audioFeedbackCritsect;
+ delete _sendAudioCritsect;
+ delete _audioFeedbackCritsect;
}
WebRtc_Word32
@@ -296,7 +296,7 @@
// A source MAY send events and coded audio packets for the same time
// but we don't support it
{
- _sendAudioCritsect.Enter();
+ _sendAudioCritsect->Enter();
if (_dtmfEventIsOn)
{
@@ -307,7 +307,7 @@
if(_packetSizeSamples > (captureTimeStamp - _dtmfTimestampLastSent) )
{
// not time to send yet
- _sendAudioCritsect.Leave();
+ _sendAudioCritsect->Leave();
return 0;
}
}
@@ -333,7 +333,7 @@
_dtmfTimeLastSent = _clock.GetTimeInMS();
}
// don't hold the critsect while calling SendTelephoneEventPacket
- _sendAudioCritsect.Leave();
+ _sendAudioCritsect->Leave();
if(send)
{
if(dtmfDurationSamples > 0xffff)
@@ -357,7 +357,7 @@
}
return(0);
}
- _sendAudioCritsect.Leave();
+ _sendAudioCritsect->Leave();
}
if(payloadSize == 0 || payloadData == NULL)
{
@@ -627,7 +627,7 @@
}
do
{
- _sendAudioCritsect.Enter();
+ _sendAudioCritsect->Enter();
//Send DTMF data
_rtpSender->BuildRTPheader(dtmfbuffer, _dtmfPayloadType, markerBit, dtmfTimeStamp);
@@ -661,7 +661,7 @@
dtmfbuffer[13] = E|R|volume;
ModuleRTPUtility::AssignUWord16ToBuffer(dtmfbuffer+14, duration);
- _sendAudioCritsect.Leave();
+ _sendAudioCritsect->Leave();
retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12);
sendCount--;
diff --git a/src/modules/rtp_rtcp/source/rtp_sender_audio.h b/src/modules/rtp_rtcp/source/rtp_sender_audio.h
index 0e0b20b..d51a473 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/src/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -95,10 +95,10 @@
WebRtc_Word32 _id;
RtpRtcpClock& _clock;
RTPSenderInterface* _rtpSender;
- CriticalSectionWrapper& _audioFeedbackCritsect;
+ CriticalSectionWrapper* _audioFeedbackCritsect;
RtpAudioFeedback* _audioFeedback;
- CriticalSectionWrapper& _sendAudioCritsect;
+ CriticalSectionWrapper* _sendAudioCritsect;
WebRtc_UWord32 _frequency;
WebRtc_UWord16 _packetSizeSamples;
diff --git a/src/modules/rtp_rtcp/source/rtp_sender_video.cc b/src/modules/rtp_rtcp/source/rtp_sender_video.cc
index ecd9e8d..a8a78d0 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -30,7 +30,7 @@
RTPSenderInterface* rtpSender) :
_id(id),
_rtpSender(*rtpSender),
- _sendVideoCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _sendVideoCritsect(CriticalSectionWrapper::CreateCriticalSection()),
_videoType(kRtpNoVideo),
_videoCodecInformation(NULL),
@@ -63,7 +63,7 @@
{
delete _videoCodecInformation;
}
- delete &_sendVideoCritsect;
+ delete _sendVideoCritsect;
}
WebRtc_Word32
diff --git a/src/modules/rtp_rtcp/source/rtp_sender_video.h b/src/modules/rtp_rtcp/source/rtp_sender_video.h
index 47f5ec7..989205f 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/src/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -150,7 +150,7 @@
WebRtc_Word32 _id;
RTPSenderInterface& _rtpSender;
- CriticalSectionWrapper& _sendVideoCritsect;
+ CriticalSectionWrapper* _sendVideoCritsect;
RtpVideoCodecTypes _videoType;
VideoCodecInformation* _videoCodecInformation;
WebRtc_UWord32 _maxBitrate;
diff --git a/src/modules/rtp_rtcp/source/ssrc_database.cc b/src/modules/rtp_rtcp/source/ssrc_database.cc
index befc5ee..b3e9ab0 100644
--- a/src/modules/rtp_rtcp/source/ssrc_database.cc
+++ b/src/modules/rtp_rtcp/source/ssrc_database.cc
@@ -54,7 +54,7 @@
WebRtc_UWord32
SSRCDatabase::CreateSSRC()
{
- CriticalSectionScoped lock(*_critSect);
+ CriticalSectionScoped lock(_critSect);
WebRtc_UWord32 ssrc = GenerateRandom();
@@ -103,7 +103,7 @@
WebRtc_Word32
SSRCDatabase::RegisterSSRC(const WebRtc_UWord32 ssrc)
{
- CriticalSectionScoped lock(*_critSect);
+ CriticalSectionScoped lock(_critSect);
#ifndef WEBRTC_NO_STL
@@ -143,7 +143,7 @@
WebRtc_Word32
SSRCDatabase::ReturnSSRC(const WebRtc_UWord32 ssrc)
{
- CriticalSectionScoped lock(*_critSect);
+ CriticalSectionScoped lock(_critSect);
#ifndef WEBRTC_NO_STL
_ssrcMap.erase(ssrc);
diff --git a/src/modules/rtp_rtcp/source/tmmbr_help.cc b/src/modules/rtp_rtcp/source/tmmbr_help.cc
index 43537f8..cf34e08 100644
--- a/src/modules/rtp_rtcp/source/tmmbr_help.cc
+++ b/src/modules/rtp_rtcp/source/tmmbr_help.cc
@@ -62,7 +62,7 @@
}
TMMBRHelp::TMMBRHelp(const bool audio) :
- _criticalSection(*CriticalSectionWrapper::CreateCriticalSection()),
+ _criticalSection(CriticalSectionWrapper::CreateCriticalSection()),
_audio(audio),
_candidateSet(),
_boundingSet(),
@@ -78,7 +78,7 @@
delete [] _ptrMaxPRBoundingSet;
_ptrIntersectionBoundingSet = 0;
_ptrMaxPRBoundingSet = 0;
- delete &_criticalSection;
+ delete _criticalSection;
}
TMMBRSet*
diff --git a/src/modules/rtp_rtcp/source/tmmbr_help.h b/src/modules/rtp_rtcp/source/tmmbr_help.h
index 0575f1d..35704fe 100644
--- a/src/modules/rtp_rtcp/source/tmmbr_help.h
+++ b/src/modules/rtp_rtcp/source/tmmbr_help.h
@@ -65,7 +65,7 @@
WebRtc_Word32 FindTMMBRBoundingSet(WebRtc_Word32 numCandidates, TMMBRSet& candidateSet);
private:
- CriticalSectionWrapper& _criticalSection;
+ CriticalSectionWrapper* _criticalSection;
const bool _audio;
TMMBRSet _candidateSet;
TMMBRSet _boundingSet;
diff --git a/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.cpp b/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.cpp
index edc26c8..2940abd 100644
--- a/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.cpp
+++ b/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.cpp
@@ -199,7 +199,7 @@
_procThread(NULL),
_startTimeMs(-1),
_stopTimeMs(-1),
-_statCritSect(*CriticalSectionWrapper::CreateCriticalSection())
+_statCritSect(CriticalSectionWrapper::CreateCriticalSection())
{
_sendrec = new TestSenderReceiver();
}
@@ -212,7 +212,7 @@
Stop();
}
- _statCritSect.Enter();
+ _statCritSect->Enter();
delete &_statCritSect;
if (_sendrec)
diff --git a/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h b/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h
index 2dccdb0..bab1b94 100644
--- a/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h
+++ b/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h
@@ -76,7 +76,7 @@
WebRtc_Word64 _stopTimeMs;
// Statistics, protected by separate CritSect
- CriticalSectionWrapper& _statCritSect;
+ CriticalSectionWrapper* _statCritSect;
StatVec _rateVecKbps;
StatVec _rttVecMs;
StatVec _lossVec;
diff --git a/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc b/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc
index e708b0f..9c81fd0 100644
--- a/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc
+++ b/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc
@@ -386,7 +386,7 @@
_legendEnabled(true),
_donePlottingEvent(EventWrapper::Create())
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
_xlim[0] = 0;
_xlim[1] = 0;
@@ -419,7 +419,7 @@
int MatlabPlot::AddLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/)
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
if (!_enabled)
{
return -1;
@@ -435,7 +435,7 @@
int MatlabPlot::AddTimeLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/,
WebRtc_Word64 refTimeMs /*= -1*/)
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
if (!_enabled)
{
@@ -451,7 +451,7 @@
int MatlabPlot::GetLineIx(const char *name)
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
if (!_enabled)
{
@@ -485,7 +485,7 @@
void MatlabPlot::Append(int lineIndex, double x, double y)
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
if (!_enabled)
{
@@ -505,7 +505,7 @@
void MatlabPlot::Append(int lineIndex, double y)
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
if (!_enabled)
{
@@ -525,7 +525,7 @@
int MatlabPlot::Append(const char *name, double x, double y)
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
if (!_enabled)
{
@@ -548,7 +548,7 @@
int MatlabPlot::Append(const char *name, double y)
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
if (!_enabled)
{
@@ -571,7 +571,7 @@
int MatlabPlot::Length(char *name)
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
if (!_enabled)
{
@@ -592,7 +592,7 @@
void MatlabPlot::SetPlotAttribute(char *name, char *plotAttrib)
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
if (!_enabled)
{
@@ -789,7 +789,7 @@
void MatlabPlot::Plot()
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
_timeToPlot = true;
@@ -801,7 +801,7 @@
void MatlabPlot::Reset()
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
_enabled = true;
@@ -814,7 +814,7 @@
void MatlabPlot::SetFigHandle(int handle)
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
if (handle > 0)
_figHandle = handle;
@@ -823,21 +823,21 @@
bool
MatlabPlot::TimeToPlot()
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
return _enabled && _timeToPlot;
}
void
MatlabPlot::Plotting()
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
_plotting = true;
}
void
MatlabPlot::DonePlotting()
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
_timeToPlot = false;
_plotting = false;
_donePlottingEvent->Set();
@@ -858,7 +858,7 @@
int MatlabPlot::MakeTrend(const char *sourceName, const char *trendName, double slope, double offset, const char *plotAttrib)
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
int sourceIx;
int trendIx;
@@ -941,7 +941,7 @@
MatlabPlot * MatlabEngine::NewPlot(MatlabPlot *newPlot)
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
//MatlabPlot *newPlot = new MatlabPlot();
@@ -958,7 +958,7 @@
void MatlabEngine::DeletePlot(MatlabPlot *plot)
{
- CriticalSectionScoped cs(*_critSect);
+ CriticalSectionScoped cs(_critSect);
if (plot == NULL)
{
diff --git a/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc b/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc
index 20ffe60..d322242 100644
--- a/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc
+++ b/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc
@@ -33,7 +33,7 @@
TestLoadGenerator::TestLoadGenerator(TestSenderReceiver *sender, WebRtc_Word32 rtpSampleRate)
:
-_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+_critSect(CriticalSectionWrapper::CreateCriticalSection()),
_eventPtr(NULL),
_genThread(NULL),
_bitrateKbps(0),
@@ -50,7 +50,7 @@
Stop();
}
- delete &_critSect;
+ delete _critSect;
}
WebRtc_Word32 TestLoadGenerator::SetBitrate (WebRtc_Word32 newBitrateKbps)
diff --git a/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h b/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
index 4ab7519..c22591c 100644
--- a/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
+++ b/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
@@ -42,7 +42,7 @@
const WebRtc_UWord32 payloadSize,
const webrtc::FrameType frameType = webrtc::kVideoFrameDelta);
- webrtc::CriticalSectionWrapper& _critSect;
+ webrtc::CriticalSectionWrapper* _critSect;
webrtc::EventWrapper *_eventPtr;
webrtc::ThreadWrapper* _genThread;
WebRtc_Word32 _bitrateKbps;
diff --git a/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc b/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
index 2e8270a..1fc0fd3 100644
--- a/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
+++ b/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
@@ -37,7 +37,7 @@
TestSenderReceiver::TestSenderReceiver (void)
:
-_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+_critSect(CriticalSectionWrapper::CreateCriticalSection()),
_eventPtr(NULL),
_procThread(NULL),
_running(false),
@@ -85,7 +85,7 @@
Stop(); // N.B. without critSect
- _critSect.Enter();
+ _critSect->Enter();
if (_rtp)
{
@@ -99,7 +99,7 @@
_transport = NULL;
}
- delete &_critSect;
+ delete _critSect;
}
@@ -440,5 +440,3 @@
bwEstimateKbitMax);
}
}
-
-
diff --git a/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h b/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
index cc9ccdd..7f7f2f0 100644
--- a/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
+++ b/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
@@ -148,7 +148,7 @@
private:
RtpRtcp* _rtp;
UdpTransport* _transport;
- webrtc::CriticalSectionWrapper& _critSect;
+ webrtc::CriticalSectionWrapper* _critSect;
webrtc::EventWrapper *_eventPtr;
webrtc::ThreadWrapper* _procThread;
bool _running;