Refactor the internal API to the rtp/rtcp module.
Combination of previous CLs in revisions 2211, 2212, 2214, 2215, 2216.
Review URL: https://webrtc-codereview.appspot.com/570008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2231 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 8906e20..e2ab811 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -36,222 +36,103 @@
const WebRtc_UWord16 kDefaultRtt = 200;
-RtpRtcp* RtpRtcp::CreateRtpRtcp(const WebRtc_Word32 id,
- bool audio) {
- if(audio) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id, "CreateRtpRtcp(audio)");
+RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
+ if (configuration.clock) {
+ return new ModuleRtpRtcpImpl(configuration);
} else {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id, "CreateRtpRtcp(video)");
- }
- // ModuleRTPUtility::GetSystemClock() creates a new instance of a system
- // clock implementation. The OwnsClock() function informs the module that
- // it is responsible for deleting the instance.
- ModuleRtpRtcpImpl* rtp_rtcp_instance = new ModuleRtpRtcpImpl(id,
- audio, ModuleRTPUtility::GetSystemClock());
- rtp_rtcp_instance->OwnsClock();
- return rtp_rtcp_instance;
-}
-
-RtpRtcp* RtpRtcp::CreateRtpRtcp(const WebRtc_Word32 id,
- const bool audio,
- RtpRtcpClock* clock) {
- if (audio) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- id,
- "CreateRtpRtcp(audio)");
- } else {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- id,
- "CreateRtpRtcp(video)");
- }
- return new ModuleRtpRtcpImpl(id, audio, clock);
-}
-
-void RtpRtcp::DestroyRtpRtcp(RtpRtcp* module) {
- if (module) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- static_cast<ModuleRtpRtcpImpl*>(module)->Id(),
- "DestroyRtpRtcp()");
- delete static_cast<ModuleRtpRtcpImpl*>(module);
+ RtpRtcp::Configuration configuration_copy;
+ memcpy(&configuration_copy, &configuration,
+ sizeof(RtpRtcp::Configuration));
+ configuration_copy.clock = ModuleRTPUtility::GetSystemClock();
+ ModuleRtpRtcpImpl* rtp_rtcp_instance =
+ new ModuleRtpRtcpImpl(configuration_copy);
+ rtp_rtcp_instance->OwnsClock();
+ return rtp_rtcp_instance;
}
}
-ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const WebRtc_Word32 id,
- const bool audio,
- RtpRtcpClock* clock):
- _rtpSender(id, audio, clock),
- _rtpReceiver(id, audio, clock, this),
- _rtcpSender(id, audio, clock, this),
- _rtcpReceiver(id, clock, this),
- _owns_clock(false),
- _clock(*clock),
- _id(id),
- _audio(audio),
- _collisionDetected(false),
- _lastProcessTime(clock->GetTimeInMS()),
- _lastBitrateProcessTime(clock->GetTimeInMS()),
- _lastPacketTimeoutProcessTime(clock->GetTimeInMS()),
- _packetOverHead(28), // IPV4 UDP
- _criticalSectionModulePtrs(CriticalSectionWrapper::CreateCriticalSection()),
- _criticalSectionModulePtrsFeedback(
- CriticalSectionWrapper::CreateCriticalSection()),
- _defaultModule(NULL),
- _audioModule(NULL),
- _videoModule(NULL),
- _deadOrAliveActive(false),
- _deadOrAliveTimeoutMS(0),
- _deadOrAliveLastTimer(0),
- _receivedNTPsecsAudio(0),
- _receivedNTPfracAudio(0),
- _RTCPArrivalTimeSecsAudio(0),
- _RTCPArrivalTimeFracAudio(0),
- _nackMethod(kNackOff),
- _nackLastTimeSent(0),
- _nackLastSeqNumberSent(0),
- _simulcast(false),
- _keyFrameReqMethod(kKeyFrameReqFirRtp)
+ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
+ : _rtpSender(configuration.id, configuration.audio, configuration.clock),
+ _rtpReceiver(configuration.id, configuration.audio, configuration.clock,
+ this),
+ _rtcpSender(configuration.id, configuration.audio, configuration.clock,
+ this),
+ _rtcpReceiver(configuration.id, configuration.clock, this),
+ _owns_clock(false),
+ _clock(*configuration.clock),
+ _id(configuration.id),
+ _audio(configuration.audio),
+ _collisionDetected(false),
+ _lastProcessTime(configuration.clock->GetTimeInMS()),
+ _lastBitrateProcessTime(configuration.clock->GetTimeInMS()),
+ _lastPacketTimeoutProcessTime(configuration.clock->GetTimeInMS()),
+ _packetOverHead(28), // IPV4 UDP
+ _criticalSectionModulePtrs(
+ CriticalSectionWrapper::CreateCriticalSection()),
+ _criticalSectionModulePtrsFeedback(
+ CriticalSectionWrapper::CreateCriticalSection()),
+ _defaultModule(
+ static_cast<ModuleRtpRtcpImpl*>(configuration.default_module)),
+ _deadOrAliveActive(false),
+ _deadOrAliveTimeoutMS(0),
+ _deadOrAliveLastTimer(0),
+ _nackMethod(kNackOff),
+ _nackLastTimeSent(0),
+ _nackLastSeqNumberSent(0),
+ _simulcast(false),
+ _keyFrameReqMethod(kKeyFrameReqFirRtp)
#ifdef MATLAB
- , _plot1(NULL)
+ , _plot1(NULL)
#endif
{
_sendVideoCodec.codecType = kVideoCodecUnknown;
+
+ if (_defaultModule) {
+ _defaultModule->RegisterChildModule(this);
+ }
+ // TODO(pwestin) move to constructors of each rtp/rtcp sender/receiver object.
+ _rtpReceiver.RegisterIncomingDataCallback(configuration.incoming_data);
+ _rtpReceiver.RegisterIncomingRTPCallback(configuration.incoming_messages);
+ _rtcpReceiver.RegisterRtcpObservers(configuration.intra_frame_callback,
+ configuration.bandwidth_callback,
+ configuration.rtcp_feedback);
+ _rtpSender.RegisterAudioCallback(configuration.audio_messages);
+ _rtpReceiver.RegisterIncomingAudioCallback(configuration.audio_messages);
+
+ _rtpSender.RegisterSendTransport(configuration.outgoing_transport);
+ _rtcpSender.RegisterSendTransport(configuration.outgoing_transport);
+
+ _rtcpSender.SetRemoteBitrateObserver(configuration.bitrate_observer);
+
// make sure that RTCP objects are aware of our SSRC
WebRtc_UWord32 SSRC = _rtpSender.SSRC();
_rtcpSender.SetSSRC(SSRC);
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
+ WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s created", __FUNCTION__);
}
ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() {
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
- // make sure to unregister this module from other modules
+ // All child modules MUST be deleted before deleting the default.
+ assert(_childModules.empty());
- const bool defaultInstance(_childModules.empty() ? false : true);
-
- if (defaultInstance) {
- // deregister for the default module
- // will go in to the child modules and remove it self
- std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- while (it != _childModules.end()) {
- RtpRtcp* module = *it;
- _childModules.erase(it);
- if (module) {
- module->DeRegisterDefaultModule();
- }
- it = _childModules.begin();
- }
- } else {
- // deregister for the child modules
- // will go in to the default and remove it self
- DeRegisterDefaultModule();
+ // Deregister for the child modules
+ // will go in to the default and remove it self
+ if (_defaultModule) {
+ _defaultModule->DeRegisterChildModule(this);
}
-
- if (_audio) {
- DeRegisterVideoModule();
- } else {
- DeRegisterSyncModule();
- }
-
#ifdef MATLAB
if (_plot1) {
eng.DeletePlot(_plot1);
_plot1 = NULL;
}
#endif
-
- delete _criticalSectionModulePtrs;
- delete _criticalSectionModulePtrsFeedback;
if (_owns_clock) {
delete &_clock;
}
}
-WebRtc_Word32 ModuleRtpRtcpImpl::ChangeUniqueId(const WebRtc_Word32 id) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "ChangeUniqueId(new id:%d)", id);
-
- _id = id;
-
- _rtpReceiver.ChangeUniqueId(id);
- _rtcpReceiver.ChangeUniqueId(id);
- _rtpSender.ChangeUniqueId(id);
- _rtcpSender.ChangeUniqueId(id);
- return 0;
-}
-
-// default encoder that we need to multiplex out
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterDefaultModule(RtpRtcp* module) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterDefaultModule(module:0x%x)", module);
-
- if (module == NULL) {
- return -1;
- }
- if (module == this) {
- WEBRTC_TRACE(kTraceError,
- kTraceRtpRtcp,
- _id,
- "RegisterDefaultModule can't register self as default");
- return -1;
- }
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
-
- if (_defaultModule) {
- _defaultModule->DeRegisterChildModule(this);
- }
- _defaultModule = (ModuleRtpRtcpImpl*)module;
- _defaultModule->RegisterChildModule(this);
- return 0;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterDefaultModule() {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "DeRegisterDefaultModule()");
-
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- if (_defaultModule) {
- _defaultModule->DeRegisterChildModule(this);
- _defaultModule = NULL;
- }
- return 0;
-}
-
-bool ModuleRtpRtcpImpl::DefaultModuleRegistered() {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "DefaultModuleRegistered()");
-
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- if (_defaultModule) {
- return true;
- }
- return false;
-}
-
-WebRtc_UWord32 ModuleRtpRtcpImpl::NumberChildModules() {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "NumberChildModules");
-
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- CriticalSectionScoped doubleLock(_criticalSectionModulePtrsFeedback);
- // we use two locks for protecting _childModules one
- // (_criticalSectionModulePtrsFeedback) for incoming messages
- // (BitrateSent and UpdateTMMBR) and _criticalSectionModulePtrs for
- // all outgoing messages sending packets etc
-
- return _childModules.size();
-}
-
void ModuleRtpRtcpImpl::RegisterChildModule(RtpRtcp* module) {
WEBRTC_TRACE(kTraceModuleCall,
kTraceRtpRtcp,
@@ -259,9 +140,9 @@
"RegisterChildModule(module:0x%x)",
module);
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ CriticalSectionScoped lock(_criticalSectionModulePtrs.get());
- CriticalSectionScoped doubleLock(_criticalSectionModulePtrsFeedback);
+ CriticalSectionScoped doubleLock(_criticalSectionModulePtrsFeedback.get());
// we use two locks for protecting _childModules one
// (_criticalSectionModulePtrsFeedback) for incoming
// messages (BitrateSent) and _criticalSectionModulePtrs
@@ -275,9 +156,9 @@
_id,
"DeRegisterChildModule(module:0x%x)", removeModule);
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ CriticalSectionScoped lock(_criticalSectionModulePtrs.get());
- CriticalSectionScoped doubleLock(_criticalSectionModulePtrsFeedback);
+ CriticalSectionScoped doubleLock(_criticalSectionModulePtrsFeedback.get());
std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
while (it != _childModules.end()) {
@@ -290,76 +171,6 @@
}
}
-// Lip-sync between voice-video engine,
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSyncModule(RtpRtcp* audioModule) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterSyncModule(module:0x%x)",
- audioModule);
-
- if (audioModule == NULL) {
- return -1;
- }
- if (_audio) {
- return -1;
- }
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- _audioModule = (ModuleRtpRtcpImpl*)audioModule;
- return _audioModule->RegisterVideoModule(this);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterSyncModule() {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "DeRegisterSyncModule()");
-
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- if (_audioModule) {
- ModuleRtpRtcpImpl* audioModule = _audioModule;
- _audioModule = NULL;
- _receivedNTPsecsAudio = 0;
- _receivedNTPfracAudio = 0;
- _RTCPArrivalTimeSecsAudio = 0;
- _RTCPArrivalTimeFracAudio = 0;
- audioModule->DeRegisterVideoModule();
- }
- return 0;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterVideoModule(RtpRtcp* videoModule) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterVideoModule(module:0x%x)",
- videoModule);
-
- if (videoModule == NULL) {
- return -1;
- }
- if (!_audio) {
- return -1;
- }
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- _videoModule = (ModuleRtpRtcpImpl*)videoModule;
- return 0;
-}
-
-void ModuleRtpRtcpImpl::DeRegisterVideoModule() {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "DeRegisterVideoModule()");
-
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- if (_videoModule) {
- ModuleRtpRtcpImpl* videoModule = _videoModule;
- _videoModule = NULL;
- videoModule->DeRegisterSyncModule();
- }
-}
-
// returns the number of milliseconds until the module want a worker thread
// to call Process
WebRtc_Word32 ModuleRtpRtcpImpl::TimeUntilNextProcess() {
@@ -436,20 +247,6 @@
* Receiver
*/
-WebRtc_Word32 ModuleRtpRtcpImpl::InitReceiver() {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "InitReceiver()");
-
- _packetOverHead = 28; // default is IPV4 UDP
- _receivedNTPsecsAudio = 0;
- _receivedNTPfracAudio = 0;
- _RTCPArrivalTimeSecsAudio = 0;
- _RTCPArrivalTimeFracAudio = 0;
-
- _rtpReceiver.Init();
- _rtpReceiver.SetPacketOverHead(_packetOverHead);
- return 0;
-}
-
void ModuleRtpRtcpImpl::ProcessDeadOrAliveTimer() {
if (_deadOrAliveActive) {
const WebRtc_UWord32 now = _clock.GetTimeInMS();
@@ -687,7 +484,6 @@
_id,
"IncomingPacket(packetLength:%u)",
incomingPacketLength);
-
// minimum RTP is 12 bytes
// minimum RTCP is 8 bytes (RTCP BYE)
if (incomingPacketLength < 8 || incomingPacket == NULL) {
@@ -754,85 +550,10 @@
}
}
-WebRtc_Word32 ModuleRtpRtcpImpl::IncomingAudioNTP(
- const WebRtc_UWord32 audioReceivedNTPsecs,
- const WebRtc_UWord32 audioReceivedNTPfrac,
- const WebRtc_UWord32 audioRTCPArrivalTimeSecs,
- const WebRtc_UWord32 audioRTCPArrivalTimeFrac) {
- _receivedNTPsecsAudio = audioReceivedNTPsecs;
- _receivedNTPfracAudio = audioReceivedNTPfrac;
- _RTCPArrivalTimeSecsAudio = audioRTCPArrivalTimeSecs;
- _RTCPArrivalTimeFracAudio = audioRTCPArrivalTimeFrac;
- return 0;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterIncomingDataCallback(
- RtpData* incomingDataCallback) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterIncomingDataCallback(incomingDataCallback:0x%x)",
- incomingDataCallback);
-
- return _rtpReceiver.RegisterIncomingDataCallback(incomingDataCallback);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterIncomingRTPCallback(
- RtpFeedback* incomingMessagesCallback) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterIncomingRTPCallback(incomingMessagesCallback:0x%x)",
- incomingMessagesCallback);
-
- return _rtpReceiver.RegisterIncomingRTPCallback(incomingMessagesCallback);
-}
-
-void ModuleRtpRtcpImpl::RegisterRtcpObservers(
- RtcpIntraFrameObserver* intra_frame_callback,
- RtcpBandwidthObserver* bandwidth_callback,
- RtcpFeedback* feedback_callback) {
- _rtcpReceiver.RegisterRtcpObservers(intra_frame_callback, bandwidth_callback,
- feedback_callback);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterAudioCallback(
- RtpAudioFeedback* messagesCallback) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterAudioCallback(messagesCallback:0x%x)",
- messagesCallback);
-
- if (_rtpSender.RegisterAudioCallback(messagesCallback) == 0) {
- return _rtpReceiver.RegisterIncomingAudioCallback(messagesCallback);
- }
- return -1;
-}
-
/**
* Sender
*/
-WebRtc_Word32 ModuleRtpRtcpImpl::InitSender() {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "InitSender()");
-
- _collisionDetected = false;
-
- // if we are already receiving inform our sender to avoid collision
- if (_rtpSender.Init(_rtpReceiver.SSRC()) != 0) {
- return -1;
- }
- WebRtc_Word32 retVal = _rtcpSender.Init();
-
- // make sure that RTCP objects are aware of our SSRC
- // (it could have changed due to collision)
- WebRtc_UWord32 SSRC = _rtpSender.SSRC();
- _rtcpReceiver.SetSSRC(SSRC);
- _rtcpSender.SetSSRC(SSRC);
- return retVal;
-}
-
WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSendPayload(
const CodecInst& voiceCodec) {
WEBRTC_TRACE(kTraceModuleCall,
@@ -962,7 +683,7 @@
if (defaultInstance) {
// for default we need to update all child modules too
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ CriticalSectionScoped lock(_criticalSectionModulePtrs.get());
std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
while (it != _childModules.end()) {
@@ -1060,7 +781,7 @@
return _rtpSender.SendingMedia();
}
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ CriticalSectionScoped lock(_criticalSectionModulePtrs.get());
std::list<ModuleRtpRtcpImpl*>::const_iterator it = _childModules.begin();
while (it != _childModules.end()) {
RTPSender& rtpSender = (*it)->_rtpSender;
@@ -1072,19 +793,6 @@
return false;
}
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSendTransport(
- Transport* outgoingTransport) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterSendTransport(0x%x)", outgoingTransport);
-
- if (_rtpSender.RegisterSendTransport(outgoingTransport) == 0) {
- return _rtcpSender.RegisterSendTransport(outgoingTransport);
- }
- return -1;
-}
-
WebRtc_Word32 ModuleRtpRtcpImpl::SendOutgoingData(
FrameType frameType,
WebRtc_Word8 payloadType,
@@ -1121,7 +829,7 @@
return -1;
}
int idx = 0;
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ CriticalSectionScoped lock(_criticalSectionModulePtrs.get());
std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
for (; idx < rtpVideoHdr->simulcastIdx; idx++) {
it++;
@@ -1144,7 +852,7 @@
NULL,
&(rtpVideoHdr->codecHeader));
} else {
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ CriticalSectionScoped lock(_criticalSectionModulePtrs.get());
// TODO(pwestin) remove codecInfo from SendOutgoingData
VideoCodecInformation* codecInfo = NULL;
@@ -1198,7 +906,7 @@
const bool defaultInstance(_childModules.empty() ? false : true);
if (defaultInstance) {
// for default we need to update all child modules too
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ CriticalSectionScoped lock(_criticalSectionModulePtrs.get());
std::list<ModuleRtpRtcpImpl*>::const_iterator it =
_childModules.begin();
while (it != _childModules.end()) {
@@ -1563,11 +1271,6 @@
return _rtcpSender.SetREMBData(bitrate, numberOfSSRC, SSRC);
}
-bool ModuleRtpRtcpImpl::SetRemoteBitrateObserver(
- RtpRemoteBitrateObserver* observer) {
- return _rtcpSender.SetRemoteBitrateObserver(observer);
-}
-
/*
* (IJ) Extended jitter report.
*/
@@ -1655,7 +1358,7 @@
const bool defaultInstance(_childModules.empty() ? false : true);
if (defaultInstance) {
// for default we need to check all child modules too
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ CriticalSectionScoped lock(_criticalSectionModulePtrs.get());
std::list<ModuleRtpRtcpImpl*>::const_iterator it =
_childModules.begin();
while (it != _childModules.end()) {
@@ -1918,7 +1621,7 @@
const bool haveChildModules(_childModules.empty() ? false : true);
if (haveChildModules) {
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ CriticalSectionScoped lock(_criticalSectionModulePtrs.get());
if (_simulcast) {
uint32_t bitrate_remainder = bitrate;
std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
@@ -1994,7 +1697,7 @@
const bool defaultInstance(_childModules.empty() ? false : true);
if (defaultInstance) {
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ CriticalSectionScoped lock(_criticalSectionModulePtrs.get());
std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
while (it != _childModules.end()) {
@@ -2041,7 +1744,7 @@
const bool defaultInstance(_childModules.empty() ? false : true);
if (defaultInstance) {
// for default we need to check all child modules too
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ CriticalSectionScoped lock(_criticalSectionModulePtrs.get());
std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
while (it != _childModules.end()) {
RtpRtcp* module = *it;
@@ -2075,7 +1778,7 @@
const bool defaultInstance(_childModules.empty() ? false : true);
if (defaultInstance) {
// for default we need to update all child modules too
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
+ CriticalSectionScoped lock(_criticalSectionModulePtrs.get());
std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
while (it != _childModules.end()) {
@@ -2128,7 +1831,7 @@
if (defaultInstance) {
// for default we need to update the send bitrate
- CriticalSectionScoped lock(_criticalSectionModulePtrsFeedback);
+ CriticalSectionScoped lock(_criticalSectionModulePtrsFeedback.get());
if (totalRate != NULL)
*totalRate = 0;
@@ -2183,66 +1886,6 @@
return 0;
}
-// for lip sync
-void ModuleRtpRtcpImpl::OnReceivedNTP() {
- // don't do anything if we are the audio module
- // video module is responsible for sync
- if (!_audio) {
- WebRtc_Word32 diff = 0;
- WebRtc_UWord32 receivedNTPsecs = 0;
- WebRtc_UWord32 receivedNTPfrac = 0;
- WebRtc_UWord32 RTCPArrivalTimeSecs = 0;
- WebRtc_UWord32 RTCPArrivalTimeFrac = 0;
-
- if (0 == _rtcpReceiver.NTP(&receivedNTPsecs,
- &receivedNTPfrac,
- &RTCPArrivalTimeSecs,
- &RTCPArrivalTimeFrac)) {
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
-
- if (_audioModule) {
- if (0 != _audioModule->RemoteNTP(&_receivedNTPsecsAudio,
- &_receivedNTPfracAudio,
- &_RTCPArrivalTimeSecsAudio,
- &_RTCPArrivalTimeFracAudio)) {
- // failed ot get audio NTP
- return;
- }
- }
- if (_receivedNTPfracAudio != 0) {
- // ReceivedNTPxxx is NTP at sender side when sent.
- // RTCPArrivalTimexxx is NTP at receiver side when received.
- // can't use ConvertNTPTimeToMS since calculation can be
- // negative
-
- WebRtc_Word32 NTPdiff = (WebRtc_Word32)
- ((_receivedNTPsecsAudio - receivedNTPsecs) *
- 1000); // ms
- NTPdiff += (WebRtc_Word32)
- (_receivedNTPfracAudio / FracMS - receivedNTPfrac / FracMS);
-
- WebRtc_Word32 RTCPdiff =
- static_cast<WebRtc_Word32> ((_RTCPArrivalTimeSecsAudio -
- RTCPArrivalTimeSecs) * 1000);
- RTCPdiff += (WebRtc_Word32)
- (_RTCPArrivalTimeFracAudio / FracMS -
- RTCPArrivalTimeFrac / FracMS);
-
- diff = NTPdiff - RTCPdiff;
- // if diff is + video is behind
- if (diff < -1000 || diff > 1000) {
- // unresonable ignore value.
- diff = 0;
- return;
- }
- }
- }
- // export via callback
- // after release of critsect
- _rtcpReceiver.UpdateLipSync(diff);
- }
-}
-
RateControlRegion ModuleRtpRtcpImpl::OnOverUseStateUpdate(
const RateControlInput& rateControlInput) {
