From fb2205a84eb2dcf619c1967d17c2a5b645942f13 Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandre.savard@savoirfairelinux.net>
Date: Mon, 17 Aug 2009 08:59:49 -0400
Subject: [PATCH] [#1883] Fix mem leaks in audio rtp
---
sflphone-common/src/audio/audiortp.cpp | 37 +++++++++++++--------
sflphone-common/src/audio/audiortp.h | 5 ++-
sflphone-common/src/audio/mainbuffer.cpp | 41 ++++++++++++++++++++----
sflphone-common/src/audio/pulselayer.cpp | 4 +++
sflphone-common/src/audio/ringbuffer.cpp | 18 +++++++----
sflphone-common/src/conference.cpp | 30 +++++++++++++++--
sflphone-common/src/conference.h | 2 ++
sflphone-common/src/managerimpl.cpp | 30 ++++++++++++++---
sflphone-common/src/managerimpl.h | 4 +++
sflphone-common/src/sipcall.cpp | 3 +-
sflphone-common/src/sipvoiplink.cpp | 2 +-
11 files changed, 141 insertions(+), 35 deletions(-)
diff --git a/sflphone-common/src/audio/audiortp.cpp b/sflphone-common/src/audio/audiortp.cpp
index 4109f03efe..c744424f69 100644
--- a/sflphone-common/src/audio/audiortp.cpp
+++ b/sflphone-common/src/audio/audiortp.cpp
@@ -43,23 +43,26 @@
////////////////////////////////////////////////////////////////////////////////
// AudioRtp
////////////////////////////////////////////////////////////////////////////////
-AudioRtp::AudioRtp() :_RTXThread (0), _symmetric(), _threadMutex()
+AudioRtp::AudioRtp() :_RTXThread (0), _symmetric(), _rtpMutex()
{
}
AudioRtp::~AudioRtp (void)
{
- ost::MutexLock m (_threadMutex);
+ ost::MutexLock m (_rtpMutex);
- delete _RTXThread;
- _RTXThread = 0;
+ if (_RTXThread != _RTXThread)
+ {
+ delete _RTXThread;
+ _RTXThread = 0;
+ }
}
void
AudioRtp::createNewSession (SIPCall *ca)
{
- ost::MutexLock m (_threadMutex);
+ ost::MutexLock m (_rtpMutex);
_debug ("AudioRtp::Create new rtp session\n");
@@ -83,7 +86,7 @@ AudioRtp::createNewSession (SIPCall *ca)
int
AudioRtp::start (void)
{
- ost::MutexLock m (_threadMutex);
+ ost::MutexLock m (_rtpMutex);
if (_RTXThread == 0) {
_debug ("! ARTP Failure: Cannot start audiortp thread since not yet created\n");
@@ -109,7 +112,7 @@ bool
AudioRtp::closeRtpSession ()
{
- ost::MutexLock m (_threadMutex);
+ ost::MutexLock m (_rtpMutex);
// This will make RTP threads finish.
_debug ("AudioRtp::Stopping rtp session\n");
@@ -174,9 +177,9 @@ AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym) : time (new ost::Time()),
AudioRtpRTX::~AudioRtpRTX ()
{
- // ost::MutexLock m (_threadMutex);
+ ost::MutexLock m (_rtpRtxMutex);
- _debug ("Delete AudioRtpRTX instance\n");
+ _debug ("Delete AudioRtpRTX instance in callid %s\n", _ca->getCallId().c_str());
try {
this->terminate();
@@ -188,8 +191,7 @@ AudioRtpRTX::~AudioRtpRTX ()
_debug("Unbind call id %s from all participants\n", _ca->getCallId().c_str());
_audiolayer->getMainBuffer()->unBindAll(_ca->getCallId());
- _ca = 0;
-
+ _debug("DELETE print micData address %p\n", micData);
delete [] micData;
micData = NULL;
delete [] micDataConverted;
@@ -208,6 +210,7 @@ AudioRtpRTX::~AudioRtpRTX ()
delete converter;
converter = NULL;
+ _ca = 0;
// _session->terminate();
delete _session;
@@ -221,17 +224,24 @@ AudioRtpRTX::~AudioRtpRTX ()
void
AudioRtpRTX::initBuffers()
{
+ ost::MutexLock m (_rtpRtxMutex);
+
+ _debug("AudioRtpRTX::initBuffers Init RTP buffers for %s\n", _ca->getCallId().c_str());
+
converter = new SamplerateConverter (_layerSampleRate , _layerFrameSize);
int nbSamplesMax = (int) (_layerSampleRate * _layerFrameSize /1000);
+ _debug("AudioRtpRTX::initBuffers NBSAMPLEMAX %i\n", nbSamplesMax);
micData = new SFLDataFormat[nbSamplesMax];
+ _debug("CREATE print micData address %p\n", micData);
micDataConverted = new SFLDataFormat[nbSamplesMax];
micDataEncoded = new unsigned char[nbSamplesMax];
spkrDataConverted = new SFLDataFormat[nbSamplesMax];
spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
+ Manager::instance().addStream(_ca->getCallId());
// _audiolayer->getMainBuffer()->bindCallID(_ca->getCallId());
}
@@ -356,10 +366,10 @@ AudioRtpRTX::processDataEncode()
// available bytes inside ringbuffer
int availBytesFromMic = _audiolayer->getMainBuffer()->availForGet(_ca->getCallId());
-
+
// set available byte to maxByteToGet
int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
-
+ // _debug("bytesAvail %i\n", bytesAvail);
if (bytesAvail == 0)
return 0;
@@ -573,6 +583,7 @@ AudioRtpRTX::run ()
while (!testCancel()) {
+ // _debug("Main while loop for call: %s\n", _ca->getCallId().c_str());
// Send session
sessionWaiting = _session->isWaiting();
diff --git a/sflphone-common/src/audio/audiortp.h b/sflphone-common/src/audio/audiortp.h
index 49330c2fd8..8335ef9d0d 100644
--- a/sflphone-common/src/audio/audiortp.h
+++ b/sflphone-common/src/audio/audiortp.h
@@ -222,6 +222,9 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
/** The audio codec used during the session */
AudioCodec* _audiocodec;
+
+ /** Mutex */
+ ost::Mutex _rtpRtxMutex;
};
@@ -281,7 +284,7 @@ class AudioRtp {
bool _symmetric;
/** Mutex */
- ost::Mutex _threadMutex;
+ ost::Mutex _rtpMutex;
};
#endif // __AUDIO_RTP_H__
diff --git a/sflphone-common/src/audio/mainbuffer.cpp b/sflphone-common/src/audio/mainbuffer.cpp
index e2bbe9e132..244d323fd7 100644
--- a/sflphone-common/src/audio/mainbuffer.cpp
+++ b/sflphone-common/src/audio/mainbuffer.cpp
@@ -301,7 +301,12 @@ int MainBuffer::availForPut(CallID call_id)
ost::MutexLock guard (_mutex);
- return getRingBuffer(call_id)->AvailForPut();
+ RingBuffer* ringbuffer = getRingBuffer(call_id);
+
+ if (ringbuffer == NULL)
+ return 0;
+ else
+ return ringbuffer->AvailForPut();
}
@@ -314,6 +319,9 @@ int MainBuffer::getData(void *buffer, int toCopy, unsigned short volume, CallID
CallIDSet* callid_set = getCallIDSet(call_id);
+ if(callid_set == NULL)
+ return 0;
+
if(callid_set->empty())
{
// _debug("CallIDSet with ID: \"%s\" is empty!\n", call_id.c_str());
@@ -391,15 +399,17 @@ int MainBuffer::getDataByID(void *buffer, int toCopy, unsigned short volume, Cal
int MainBuffer::availForGet(CallID call_id)
{
- // _debug("MainBuffer::availForGet\n");
ost::MutexLock guard (_mutex);
CallIDSet* callid_set = getCallIDSet(call_id);
+ if (callid_set == NULL)
+ return 0;
+
if (callid_set->empty())
{
- // _debug("CallIDSet with ID: \"%s\" is empty!\n", call_id.c_str());
+ _debug("CallIDSet with ID: \"%s\" is empty!\n", call_id.c_str());
return 0;
}
@@ -429,7 +439,12 @@ int MainBuffer::availForGet(CallID call_id)
int MainBuffer::availForGetByID(CallID call_id, CallID reader_id)
{
- return getRingBuffer(call_id)->AvailForGet(reader_id);
+ RingBuffer* ringbuffer = getRingBuffer(call_id);
+
+ if (ringbuffer == NULL)
+ return 0;
+ else
+ return ringbuffer->AvailForGet(reader_id);
}
@@ -442,6 +457,9 @@ int MainBuffer::discard(int toDiscard, CallID call_id)
CallIDSet* callid_set = getCallIDSet(call_id);
+ if (callid_set == NULL)
+ return 0;
+
if(callid_set->empty())
{
// _debug("CallIDSet with ID: \"%s\" is empty!\n", call_id.c_str());
@@ -472,7 +490,12 @@ int MainBuffer::discard(int toDiscard, CallID call_id)
int MainBuffer::discardByID(int toDiscard, CallID call_id, CallID reader_id)
{
- return getRingBuffer(call_id)->Discard(toDiscard, reader_id);
+ RingBuffer* ringbuffer = getRingBuffer(call_id);
+
+ if(ringbuffer == NULL)
+ return 0;
+ else
+ return ringbuffer->Discard(toDiscard, reader_id);
}
@@ -486,6 +509,9 @@ void MainBuffer::flush(CallID call_id)
CallIDSet* callid_set = getCallIDSet(call_id);
+ if (callid_set == NULL)
+ return;
+
if(callid_set->empty())
{
// _debug("CallIDSet with ID: \"%s\" is empty!\n", call_id.c_str());
@@ -520,5 +546,8 @@ void MainBuffer::flushDefault()
void MainBuffer::flushByID(CallID call_id, CallID reader_id)
{
- getRingBuffer(call_id)->flush(reader_id);
+ RingBuffer* ringbuffer = getRingBuffer(call_id);
+
+ if(ringbuffer != NULL)
+ ringbuffer->flush(reader_id);
}
diff --git a/sflphone-common/src/audio/pulselayer.cpp b/sflphone-common/src/audio/pulselayer.cpp
index c8c28c33ee..93f7dacb91 100644
--- a/sflphone-common/src/audio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulselayer.cpp
@@ -40,6 +40,8 @@ PulseLayer::PulseLayer (ManagerImpl* manager)
_debug ("PulseLayer::Pulse audio constructor: Create context\n");
out_buffer = new SFLDataFormat[STATIC_BUFSIZE];
+ _urgentRingBuffer.createReadPointer();
+
}
// Destructor
@@ -361,6 +363,7 @@ void PulseLayer::writeToSpeaker (void)
out_buffer[k] = 0;
if (urgentAvail > 0) {
+
// Urgent data (dtmf, incoming call signal) come first.
//_debug("Play urgent!: %i\e" , urgentAvail);
@@ -396,6 +399,7 @@ void PulseLayer::writeToSpeaker (void)
} else {
// out = (SFLDataFormat*) pa_xmalloc (framesPerBuffer * sizeof (SFLDataFormat));
// _debug("PulseLayer::writeToSpeaker _mainBuffer.getData() toGet %i\n", toGet);
+
normalAvail = _mainBuffer.availForGet();
toGet = (normalAvail < (int) (framesPerBuffer * sizeof (SFLDataFormat))) ? normalAvail : framesPerBuffer * sizeof (SFLDataFormat);
diff --git a/sflphone-common/src/audio/ringbuffer.cpp b/sflphone-common/src/audio/ringbuffer.cpp
index c8198df9a2..c342669f64 100644
--- a/sflphone-common/src/audio/ringbuffer.cpp
+++ b/sflphone-common/src/audio/ringbuffer.cpp
@@ -98,11 +98,19 @@ RingBuffer::getReadPointer(CallID call_id)
if(getNbReadPointer() == 0)
return 0;
+ // _debug("RingBuffer::getReadPointer() id %s\n", call_id.c_str());
+
ReadPointer::iterator iter = _readpointer.find(call_id);
if (iter == _readpointer.end())
{
- _debug("RingBuffer::getReadPointer Error read pointer is null\n");
- return NULL;
+ // _debug(" RingBuffer::getReadPointer Error read pointer size: %i\n", _readpointer.size());
+ // _debug(" RingBuffer::getReadPointer Error read pointer \"%s\" is null\n", call_id.c_str());
+ ReadPointer::iterator iter2;
+ for( iter2 = _readpointer.begin(); iter2 != _readpointer.end(); iter2++)
+ {
+ // x_debug(" RingBuffer::getReadPointer list avail pointer \"%s\"\n", iter2->first.c_str());
+ }
+ return 0;
}
else
{
@@ -137,7 +145,7 @@ RingBuffer::storeReadPointer(int pointer_value, CallID call_id)
if(iter != _readpointer.end())
{
iter->second = pointer_value;
- _debug("store read pointer call_id %s, size: %i \n",call_id.c_str(), _readpointer.size());
+ // _debug("store read pointer call_id %s, size: %i \n",call_id.c_str(), _readpointer.size());
}
else
{
@@ -295,10 +303,8 @@ RingBuffer::Get (void *buffer, int toCopy, unsigned short volume, CallID call_id
dest = (samplePtr) buffer;
copied = 0;
-
- int mStart = getReadPointer(call_id);
-
+ int mStart = getReadPointer(call_id);
//fprintf(stderr, "G");
while (toCopy) {
diff --git a/sflphone-common/src/conference.cpp b/sflphone-common/src/conference.cpp
index 4bb252ea67..50b2ff4116 100644
--- a/sflphone-common/src/conference.cpp
+++ b/sflphone-common/src/conference.cpp
@@ -43,6 +43,7 @@ void Conference::add(CallID participant_id)
_debug("---- Conference:: add participant %s\n", participant_id.c_str());
+ /*
if(_nbParticipant >= 1)
{
ParticipantSet::iterator iter;
@@ -54,8 +55,8 @@ void Conference::add(CallID participant_id)
}
}
- // Manager::instance().getAudioDriver()->getMainBuffer()->bindCallID(participant_id);
-
+ Manager::instance().getAudioDriver()->getMainBuffer()->bindCallID(participant_id);
+ */
_participants.insert(participant_id);
_nbParticipant++;
@@ -67,6 +68,7 @@ void Conference::remove(CallID participant_id)
_debug("---- Conference:: remove participant %s\n", participant_id.c_str());
+ /*
if(_nbParticipant >= 1)
{
ParticipantSet::iterator iter = _participants.begin();
@@ -77,9 +79,33 @@ void Conference::remove(CallID participant_id)
Manager::instance().getAudioDriver()->getMainBuffer()->unBindCallID(participant_id, *iter);
}
}
+ */
_participants.erase(participant_id);
_nbParticipant--;
}
+
+void Conference::bindParticipant(CallID participant_id)
+{
+
+ if(_nbParticipant >= 1)
+ {
+ ParticipantSet::iterator iter;
+
+ for(iter = _participants.begin(); iter != _participants.end(); iter++)
+ {
+ if (participant_id != (*iter))
+ {
+ _debug("---- Conference:: bind callid %s with %s in conference add\n", participant_id.c_str(), (*iter).c_str());
+ Manager::instance().getAudioDriver()->getMainBuffer()->bindCallID(participant_id, *iter);
+ }
+ }
+ }
+
+ _debug("---- Conference:: bind callid %s with default_id in conference add\n", participant_id.c_str());
+
+ Manager::instance().getAudioDriver()->getMainBuffer()->bindCallID(participant_id);
+
+}
diff --git a/sflphone-common/src/conference.h b/sflphone-common/src/conference.h
index 8340997470..b1e0cb2c0e 100644
--- a/sflphone-common/src/conference.h
+++ b/sflphone-common/src/conference.h
@@ -43,6 +43,8 @@ class Conference{
void remove(CallID participant_id);
+ void bindParticipant(CallID participant_id);
+
private:
/** Unique ID of the call */
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index 486ef37fcf..b0f901cd86 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -369,9 +369,6 @@ ManagerImpl::hangupCall (const CallID& id)
if (! (nbCalls > 1))
audiolayer->stopStream();
- if(participToConference(id))
- removeParticipant(id);
-
/* Direct IP to IP call */
if (getConfigFromCall (id) == Call::IPtoIP) {
returnValue = SIPVoIPLink::instance (AccountNULL)->hangup (id);
@@ -392,6 +389,9 @@ ManagerImpl::hangupCall (const CallID& id)
removeCallAccount (id);
}
+ if(participToConference(id))
+ removeParticipant(id);
+
switchCall ("");
if (_audiodriver->getLayerType() == PULSEAUDIO && getConfigInt (PREFERENCES , CONFIG_PA_VOLUME_CTRL)) {
@@ -650,8 +650,6 @@ ManagerImpl::createConference(const CallID& id)
conf->add(getCurrentCallId());
conf->add(id);
- answerCall(id);
-
}
void
@@ -705,6 +703,9 @@ ManagerImpl::addParticipant(const CallID& call_id)
{
_debug("NO CONFERENCE YET, CREATE ONE\n");
createConference(call_id);
+
+ answerCall(call_id);
+
}
else
{
@@ -747,6 +748,25 @@ ManagerImpl::removeParticipant(const CallID& call_id)
}
}
+void
+ManagerImpl::addStream(const CallID& call_id)
+{
+ _debug("ManagerImpl::addStream %s\n", call_id.c_str());
+
+ if(participToConference(call_id))
+ {
+ ConferenceMap::iterator iter = _conferencemap.find(default_conf);
+
+ Conference* conf = iter->second;
+
+ conf->bindParticipant(call_id);
+ }
+ else
+ {
+ getAudioDriver()->getMainBuffer()->bindCallID(call_id);
+ }
+}
+
//THREAD=Main
bool
ManagerImpl::saveConfig (void)
diff --git a/sflphone-common/src/managerimpl.h b/sflphone-common/src/managerimpl.h
index a3ca34541a..f6b6d9184c 100644
--- a/sflphone-common/src/managerimpl.h
+++ b/sflphone-common/src/managerimpl.h
@@ -196,6 +196,8 @@ class ManagerImpl {
void removeParticipant(const CallID& call_id);
+ void addStream(const CallID& call_id);
+
/**
* Save config to file
* @return true on success
@@ -1162,8 +1164,10 @@ class ManagerImpl {
int isStunEnabled (void);
void enableStun (void);
+ // Map
ConferenceCallMap _conferencecall;
+ //
ConferenceMap _conferencemap;
private:
diff --git a/sflphone-common/src/sipcall.cpp b/sflphone-common/src/sipcall.cpp
index fdfd2c3294..68d2eb47bc 100644
--- a/sflphone-common/src/sipcall.cpp
+++ b/sflphone-common/src/sipcall.cpp
@@ -39,7 +39,8 @@ SIPCall::SIPCall (const CallID& id, Call::CallType type, pj_pool_t *pool) : Call
SIPCall::~SIPCall()
{
-
+ delete _audiortp;
+ _audiortp = 0;
delete _local_sdp;
_local_sdp = 0;
_debug ("SIPCALL::Destructor for this class is called \n");
diff --git a/sflphone-common/src/sipvoiplink.cpp b/sflphone-common/src/sipvoiplink.cpp
index fd9915f79e..2023c3b6c1 100644
--- a/sflphone-common/src/sipvoiplink.cpp
+++ b/sflphone-common/src/sipvoiplink.cpp
@@ -734,7 +734,7 @@ SIPVoIPLink::hangup (const CallID& id)
// Release RTP thread
if (Manager::instance().isCurrentCall (id)) {
- _debug ("* SIP Info: Stopping AudioRTP for hangup\n");
+ _debug ("* SIP Info: Stopping AudioRTP for hangup %s\n", call->getCallId().c_str());
call->getAudioRtp()->closeRtpSession();
}
--
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