diff --git a/daemon/src/audio/audiortp/audio_rtp_factory.cpp b/daemon/src/audio/audiortp/audio_rtp_factory.cpp
index 4b4b3c5c0e1ad676e37e0e1dde0d30e0038184d9..a153e47f0c45ca7c36b50e402046c3d209ad2500 100644
--- a/daemon/src/audio/audiortp/audio_rtp_factory.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_factory.cpp
@@ -91,7 +91,7 @@ void AudioRtpFactory::initSession()
switch (keyExchangeProtocol_) {
case ZRTP:
- rtpSession_ = new AudioZrtpSession(ca_, zidFilename);
+ rtpSession_ = new AudioZrtpSession(*ca_, zidFilename);
// TODO: be careful with that. The hello hash is computed asynchronously. Maybe it's
// not even available at that point.
if (helloHashEnabled_)
@@ -99,14 +99,14 @@ void AudioRtpFactory::initSession()
break;
case SDES:
- rtpSession_ = new AudioSrtpSession(ca_);
+ rtpSession_ = new AudioSrtpSession(*ca_);
break;
default:
throw UnsupportedRtpSessionType("Unsupported Rtp Session Exception Type!");
}
} else
- rtpSession_ = new AudioSymmetricRtpSession(ca_);
+ rtpSession_ = new AudioSymmetricRtpSession(*ca_);
}
void AudioRtpFactory::start(AudioCodec* audiocodec)
@@ -117,7 +117,7 @@ void AudioRtpFactory::start(AudioCodec* audiocodec)
if (keyExchangeProtocol_ == SDES and localContext_ and remoteContext_)
static_cast<AudioSrtpSession *>(rtpSession_)->restoreCryptoContext(localContext_, remoteContext_);
- if (rtpSession_->startRtpThread(audiocodec) != 0)
+ if (rtpSession_->startRtpThread(*audiocodec) != 0)
throw AudioRtpFactoryException("AudioRtpFactory: Error: Failed to start AudioRtpSession thread");
}
@@ -150,7 +150,7 @@ void AudioRtpFactory::updateSessionMedia(AudioCodec *audiocodec)
if (rtpSession_ == NULL)
throw AudioRtpFactoryException("AudioRtpFactory: Error: rtpSession_ was null when trying to update IP address");
- rtpSession_->updateSessionMedia(audiocodec);
+ rtpSession_->updateSessionMedia(*audiocodec);
}
void AudioRtpFactory::updateDestinationIpAddress()
diff --git a/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp b/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
index 8ce03be48914e6902196162e5b23d51cba85c0d1..99c5064d94a00d36bcfba5c137f3a9ff1848e312 100644
--- a/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
@@ -67,13 +67,17 @@ AudioRtpRecord::~AudioRtpRecord()
}
-AudioRtpRecordHandler::AudioRtpRecordHandler(SIPCall *ca) : audioRtpRecord_(), id_(ca->getCallId()), echoCanceller(ca->getMemoryPool()), gainController(8000, -10.0)
+AudioRtpRecordHandler::AudioRtpRecordHandler(SIPCall &call) :
+ audioRtpRecord_(),
+ id_(call.getCallId()),
+ echoCanceller(call.getMemoryPool()),
+ gainController(8000, -10.0)
{}
AudioRtpRecordHandler::~AudioRtpRecordHandler() {}
-void AudioRtpRecordHandler::setRtpMedia(AudioCodec* audioCodec)
+void AudioRtpRecordHandler::setRtpMedia(AudioCodec *audioCodec)
{
ost::MutexLock lock(audioRtpRecord_.audioCodecMutex_);
diff --git a/daemon/src/audio/audiortp/audio_rtp_record_handler.h b/daemon/src/audio/audiortp/audio_rtp_record_handler.h
index 9012049a133d1b279448835ba31e3c5d7bcb4a75..ac630e2872aac936f7aa05f8c73897c80e6b1296 100644
--- a/daemon/src/audio/audiortp/audio_rtp_record_handler.h
+++ b/daemon/src/audio/audiortp/audio_rtp_record_handler.h
@@ -103,7 +103,7 @@ class AudioRtpRecord {
class AudioRtpRecordHandler {
public:
- AudioRtpRecordHandler(SIPCall *);
+ AudioRtpRecordHandler(SIPCall &);
virtual ~AudioRtpRecordHandler();
/**
diff --git a/daemon/src/audio/audiortp/audio_rtp_session.cpp b/daemon/src/audio/audiortp/audio_rtp_session.cpp
index 0df3c691dae301ebd7118866453910a22e3f3f03..70efc7860d0c2c61825bc62830b2572cab79f027 100644
--- a/daemon/src/audio/audiortp/audio_rtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_session.cpp
@@ -43,9 +43,9 @@
#include "manager.h"
namespace sfl {
-AudioRtpSession::AudioRtpSession(SIPCall * sipcall, ost::RTPDataQueue *queue, ost::Thread *thread) :
- AudioRtpRecordHandler(sipcall)
- , ca_(sipcall)
+AudioRtpSession::AudioRtpSession(SIPCall &call, ost::RTPDataQueue &queue, ost::Thread &thread) :
+ AudioRtpRecordHandler(call)
+ , call_(call)
, timestamp_(0)
, timestampIncrement_(0)
, queue_(queue)
@@ -55,16 +55,15 @@ AudioRtpSession::AudioRtpSession(SIPCall * sipcall, ost::RTPDataQueue *queue, os
, timestampCount_(0)
, thread_(thread)
{
- assert(ca_);
- queue_->setTypeOfService(ost::RTPDataQueue::tosEnhanced);
+ queue_.setTypeOfService(ost::RTPDataQueue::tosEnhanced);
}
AudioRtpSession::~AudioRtpSession()
{
- queue_->disableStack();
+ queue_.disableStack();
}
-void AudioRtpSession::updateSessionMedia(AudioCodec *audioCodec)
+void AudioRtpSession::updateSessionMedia(AudioCodec &audioCodec)
{
int lastSamplingRate = audioRtpRecord_.codecSampleRate_;
@@ -79,9 +78,9 @@ void AudioRtpSession::updateSessionMedia(AudioCodec *audioCodec)
}
-void AudioRtpSession::setSessionMedia(AudioCodec *audioCodec)
+void AudioRtpSession::setSessionMedia(AudioCodec &audioCodec)
{
- setRtpMedia(audioCodec);
+ setRtpMedia(&audioCodec);
// store codec info locally
int payloadType = getCodecPayloadType();
@@ -102,14 +101,14 @@ void AudioRtpSession::setSessionMedia(AudioCodec *audioCodec)
if (payloadType == g722PayloadType) {
DEBUG("AudioRtpSession: Setting G722 payload format");
- queue_->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, g722RtpClockRate));
+ queue_.setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, g722RtpClockRate));
} else {
if (dynamic) {
DEBUG("AudioRtpSession: Setting dynamic payload format");
- queue_->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, smplRate));
+ queue_.setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) payloadType, smplRate));
} else {
DEBUG("AudioRtpSession: Setting static payload format");
- queue_->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) payloadType));
+ queue_.setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) payloadType));
}
}
}
@@ -138,20 +137,20 @@ void AudioRtpSession::sendDtmfEvent()
processDataEncode();
// change Payload type for DTMF payload
- queue_->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) getDtmfPayloadType(), 8000));
+ queue_.setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) getDtmfPayloadType(), 8000));
- queue_->setMark(true);
- queue_->sendImmediate(timestamp_, (const unsigned char *)(&payload), sizeof(payload));
- queue_->setMark(false);
+ queue_.setMark(true);
+ queue_.sendImmediate(timestamp_, (const unsigned char *)(&payload), sizeof(payload));
+ queue_.setMark(false);
// get back the payload to audio
- queue_->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) getCodecPayloadType()));
+ queue_.setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) getCodecPayloadType()));
}
void AudioRtpSession::receiveSpeakerData()
{
- const ost::AppDataUnit* adu = queue_->getData(queue_->getFirstTimestamp());
+ const ost::AppDataUnit* adu = queue_.getData(queue_.getFirstTimestamp());
if (!adu)
return;
@@ -179,7 +178,7 @@ void AudioRtpSession::sendMicData()
timestamp_ += timestampIncrement_;
// putData puts the data on RTP queue, sendImmediate bypass this queue
- queue_->sendImmediate(timestamp_, getMicDataEncoded(), compSize);
+ queue_.sendImmediate(timestamp_, getMicDataEncoded(), compSize);
}
@@ -187,28 +186,28 @@ void AudioRtpSession::setSessionTimeouts()
{
DEBUG("AudioRtpSession: Set session scheduling timeout (%d) and expireTimeout (%d)", sfl::schedulingTimeout, sfl::expireTimeout);
- queue_->setSchedulingTimeout(sfl::schedulingTimeout);
- queue_->setExpireTimeout(sfl::expireTimeout);
+ queue_.setSchedulingTimeout(sfl::schedulingTimeout);
+ queue_.setExpireTimeout(sfl::expireTimeout);
}
void AudioRtpSession::setDestinationIpAddress()
{
// Store remote ip in case we would need to forget current destination
- remote_ip_ = ost::InetHostAddress(ca_->getLocalSDP()->getRemoteIP().c_str());
+ remote_ip_ = ost::InetHostAddress(call_.getLocalSDP()->getRemoteIP().c_str());
if (!remote_ip_) {
WARN("AudioRtpSession: Target IP address (%s) is not correct!",
- ca_->getLocalSDP()->getRemoteIP().data());
+ call_.getLocalSDP()->getRemoteIP().data());
return;
}
// Store remote port in case we would need to forget current destination
- remote_port_ = (unsigned short) ca_->getLocalSDP()->getRemoteAudioPort();
+ remote_port_ = (unsigned short) call_.getLocalSDP()->getRemoteAudioPort();
DEBUG("AudioRtpSession: New remote address for session: %s:%d",
- ca_->getLocalSDP()->getRemoteIP().data(), remote_port_);
+ call_.getLocalSDP()->getRemoteIP().data(), remote_port_);
- if (!queue_->addDestination(remote_ip_, remote_port_)) {
+ if (!queue_.addDestination(remote_ip_, remote_port_)) {
WARN("AudioRtpSession: Can't add new destination to session!");
return;
}
@@ -221,7 +220,7 @@ void AudioRtpSession::updateDestinationIpAddress()
// Destination address are stored in a list in ccrtp
// This method remove the current destination entry
- if (!queue_->forgetDestination(remote_ip_, remote_port_, remote_port_ + 1))
+ if (!queue_.forgetDestination(remote_ip_, remote_port_, remote_port_ + 1))
DEBUG("AudioRtpSession: Did not remove previous destination");
// new destination is stored in call
@@ -230,7 +229,7 @@ void AudioRtpSession::updateDestinationIpAddress()
}
-int AudioRtpSession::startRtpThread(AudioCodec* audiocodec)
+int AudioRtpSession::startRtpThread(AudioCodec &audiocodec)
{
if (isStarted_)
return 0;
@@ -243,8 +242,8 @@ int AudioRtpSession::startRtpThread(AudioCodec* audiocodec)
initBuffers();
initNoiseSuppress();
- queue_->enableStack();
- return thread_->start();
+ queue_.enableStack();
+ return thread_.start();
}
diff --git a/daemon/src/audio/audiortp/audio_rtp_session.h b/daemon/src/audio/audiortp/audio_rtp_session.h
index f52f6472ef4cadc3fa7364db3556cef6c76e97db..c6baca3f29fd81b9899922e6b1aef87a767a0478 100644
--- a/daemon/src/audio/audiortp/audio_rtp_session.h
+++ b/daemon/src/audio/audiortp/audio_rtp_session.h
@@ -31,16 +31,18 @@
* shall include the source code for the parts of OpenSSL used as well
* as that of the covered work.
*/
-#ifndef SFL_AUDIO_RTP_SESSION_H_
-#define SFL_AUDIO_RTP_SESSION_H_
+#ifndef AUDIO_RTP_SESSION_H_
+#define AUDIO_RTP_SESSION_H_
#include "audio_rtp_record_handler.h"
-#include <audio/codecs/audiocodec.h>
#include <ccrtp/rtp.h>
#include <ccrtp/formats.h>
#include "noncopyable.h"
class SIPCall;
+namespace ost {
+ class Thread;
+}
namespace sfl {
@@ -52,12 +54,12 @@ class AudioRtpSession : public AudioRtpRecordHandler {
* Constructor
* @param sipcall The pointer on the SIP call
*/
- AudioRtpSession(SIPCall* sipcall, ost::RTPDataQueue *queue, ost::Thread *thread);
+ AudioRtpSession(SIPCall &sipcall, ost::RTPDataQueue &queue, ost::Thread &thread);
virtual ~AudioRtpSession();
- void updateSessionMedia(AudioCodec *audioCodec);
+ void updateSessionMedia(AudioCodec &audioCodec);
- virtual int startRtpThread(AudioCodec*);
+ virtual int startRtpThread(AudioCodec&);
/**
* Used mostly when receiving a reinvite
@@ -68,7 +70,7 @@ class AudioRtpSession : public AudioRtpRecordHandler {
/**
* Set the audio codec for this RTP session
*/
- virtual void setSessionMedia(AudioCodec *codec) = 0;
+ virtual void setSessionMedia(AudioCodec &codec) = 0;
bool onRTPPacketRecv(ost::IncomingRTPPkt&);
@@ -86,7 +88,7 @@ class AudioRtpSession : public AudioRtpRecordHandler {
*/
virtual void sendMicData();
- SIPCall *ca_;
+ SIPCall &call_;
/**
* Timestamp for this session
@@ -99,7 +101,7 @@ class AudioRtpSession : public AudioRtpRecordHandler {
*/
int timestampIncrement_;
- ost::RTPDataQueue *queue_;
+ ost::RTPDataQueue &queue_;
bool isStarted_;
private:
@@ -140,7 +142,7 @@ class AudioRtpSession : public AudioRtpRecordHandler {
*/
short timestampCount_;
- ost::Thread *thread_;
+ ost::Thread &thread_;
};
}
diff --git a/daemon/src/audio/audiortp/audio_srtp_session.cpp b/daemon/src/audio/audiortp/audio_srtp_session.cpp
index 60e173ba62a63da2528abd7b638a51f789885c52..500d03ba7dc3ac89986a72a5a0915482350418df 100644
--- a/daemon/src/audio/audiortp/audio_srtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_srtp_session.cpp
@@ -29,8 +29,6 @@
*/
#include "audio_srtp_session.h"
-#include "sip/sipcall.h"
-
#include <openssl/sha.h>
#include <openssl/hmac.h>
#include <openssl/evp.h>
@@ -44,8 +42,8 @@
namespace sfl {
-AudioSrtpSession::AudioSrtpSession(SIPCall * sipcall) :
- AudioSymmetricRtpSession(sipcall),
+AudioSrtpSession::AudioSrtpSession(SIPCall &call) :
+ AudioSymmetricRtpSession(call),
remoteCryptoCtx_(NULL),
localCryptoCtx_(NULL),
localCryptoSuite_(0),
diff --git a/daemon/src/audio/audiortp/audio_srtp_session.h b/daemon/src/audio/audiortp/audio_srtp_session.h
index 02429d7b193d63a8777f7dfc441bdb1130f2a200..5cce01a60a21f6ad249115ebe7d04e2bc21626e8 100644
--- a/daemon/src/audio/audiortp/audio_srtp_session.h
+++ b/daemon/src/audio/audiortp/audio_srtp_session.h
@@ -27,10 +27,9 @@
* shall include the source code for the parts of OpenSSL used as well
* as that of the covered work.
*/
-#ifndef __AUDIO_SRTP_SESSION_H__
-#define __AUDIO_SRTP_SESSION_H__
+#ifndef AUDIO_SRTP_SESSION_H_
+#define AUDIO_SRTP_SESSION_H_
-#include "audio_rtp_session.h"
#include "audio_symmetric_rtp_session.h"
#include "sip/sdes_negotiator.h"
#include "noncopyable.h"
@@ -73,7 +72,7 @@ class AudioSrtpSession : public AudioSymmetricRtpSession {
/**
* Constructor for this rtp session
*/
- AudioSrtpSession(SIPCall * sipcall);
+ AudioSrtpSession(SIPCall &call);
/**
* Used to get sdp crypto header to be included in sdp session. This
@@ -85,7 +84,7 @@ class AudioSrtpSession : public AudioSymmetricRtpSession {
/**
* Set remote crypto header from incoming sdp offer
*/
- void setRemoteCryptoInfo(sfl::SdesNegotiator& nego);
+ void setRemoteCryptoInfo(sfl::SdesNegotiator &nego);
/**
* Init local crypto context for outgoing data
diff --git a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
index 5169ce97eb44be937033f2eb918b460e823c4d88..3324f22ff4358f7713cf8e024decb2eff9f4abc1 100644
--- a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
@@ -41,26 +41,26 @@
namespace sfl {
-AudioSymmetricRtpSession::AudioSymmetricRtpSession(SIPCall * sipcall) :
+AudioSymmetricRtpSession::AudioSymmetricRtpSession(SIPCall &call) :
ost::TimerPort()
- , ost::SymmetricRTPSession(ost::InetHostAddress(sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort())
- , AudioRtpSession(sipcall, this, this)
- , rtpThread_(new AudioRtpThread(this))
+ , ost::SymmetricRTPSession(ost::InetHostAddress(call.getLocalIp().c_str()), call.getLocalAudioPort())
+ , AudioRtpSession(call, *this, *this)
+ , rtpThread_(*this)
{
- DEBUG("AudioSymmetricRtpSession: Setting new RTP session with destination %s:%d", ca_->getLocalIp().c_str(), ca_->getLocalAudioPort());
- audioRtpRecord_.callId_ = ca_->getCallId();
+ DEBUG("AudioSymmetricRtpSession: Setting new RTP session with destination %s:%d", call_.getLocalIp().c_str(), call_.getLocalAudioPort());
+ audioRtpRecord_.callId_ = call_.getCallId();
}
AudioSymmetricRtpSession::~AudioSymmetricRtpSession()
{
- rtpThread_->running_ = false;
- delete rtpThread_;
+ if (rtpThread_.running_) {
+ rtpThread_.running_ = false;
+ rtpThread_.join();
+ }
}
-AudioSymmetricRtpSession::AudioRtpThread::AudioRtpThread(AudioSymmetricRtpSession *session) : running_(true), rtpSession_(session)
-{
- assert(rtpSession_);
-}
+AudioSymmetricRtpSession::AudioRtpThread::AudioRtpThread(AudioSymmetricRtpSession &session) : running_(true), rtpSession_(session)
+{}
void AudioSymmetricRtpSession::AudioRtpThread::run()
{
@@ -72,10 +72,10 @@ void AudioSymmetricRtpSession::AudioRtpThread::run()
while (running_) {
// Send session
- if (rtpSession_->DtmfPending())
- rtpSession_->sendDtmfEvent();
+ if (rtpSession_.DtmfPending())
+ rtpSession_.sendDtmfEvent();
else
- rtpSession_->sendMicData();
+ rtpSession_.sendMicData();
Thread::sleep(TimerPort::getTimer());
@@ -85,13 +85,13 @@ void AudioSymmetricRtpSession::AudioRtpThread::run()
DEBUG("AudioRtpThread: Leaving audio rtp thread loop");
}
-void AudioSymmetricRtpSession::setSessionMedia(AudioCodec *audioCodec)
+void AudioSymmetricRtpSession::setSessionMedia(AudioCodec &audioCodec)
{
AudioRtpSession::setSessionMedia(audioCodec);
- ca_->setRecordingSmplRate(getCodecSampleRate());
+ call_.setRecordingSmplRate(getCodecSampleRate());
}
-int AudioSymmetricRtpSession::startRtpThread(AudioCodec* audiocodec)
+int AudioSymmetricRtpSession::startRtpThread(AudioCodec &audiocodec)
{
DEBUG("AudioSymmetricRtpSession: Starting main thread");
if (isStarted_)
diff --git a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
index 96270012d262ecedeacbbf632a75a9160417f85a..b0fda2be3d926de11341e7c546ce5f69327c03f6 100644
--- a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
+++ b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
@@ -56,9 +56,9 @@ class AudioSymmetricRtpSession : public ost::TimerPort, public ost::SymmetricRTP
public:
/**
* Constructor
- * @param sipcall The pointer on the SIP call
+ * @param call The SIP call
*/
- AudioSymmetricRtpSession(SIPCall* sipcall);
+ AudioSymmetricRtpSession(SIPCall &call);
~AudioSymmetricRtpSession();
virtual bool onRTPPacketRecv(ost::IncomingRTPPkt& pkt) {
@@ -66,7 +66,7 @@ class AudioSymmetricRtpSession : public ost::TimerPort, public ost::SymmetricRTP
}
int startSymmetricRtpThread() {
- return rtpThread_->start();
+ return rtpThread_.start();
}
private:
@@ -74,7 +74,7 @@ class AudioSymmetricRtpSession : public ost::TimerPort, public ost::SymmetricRTP
class AudioRtpThread : public ost::Thread, public ost::TimerPort {
public:
- AudioRtpThread(AudioSymmetricRtpSession *session);
+ AudioRtpThread(AudioSymmetricRtpSession &session);
virtual void run();
@@ -82,12 +82,12 @@ class AudioSymmetricRtpSession : public ost::TimerPort, public ost::SymmetricRTP
private:
NON_COPYABLE(AudioRtpThread);
- AudioSymmetricRtpSession *rtpSession_;
+ AudioSymmetricRtpSession &rtpSession_;
};
- void setSessionMedia(AudioCodec *codec);
- int startRtpThread(AudioCodec* audiocodec);
+ void setSessionMedia(AudioCodec &codec);
+ int startRtpThread(AudioCodec &audiocodec);
- AudioRtpThread *rtpThread_;
+ AudioRtpThread rtpThread_;
};
}
diff --git a/daemon/src/audio/audiortp/audio_zrtp_session.cpp b/daemon/src/audio/audiortp/audio_zrtp_session.cpp
index 81903a9aae11eb9b2a5878d68064b3d15419798e..222adb89c1f3c84440fe6363002043d5e0e800ed 100644
--- a/daemon/src/audio/audiortp/audio_zrtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_zrtp_session.cpp
@@ -49,13 +49,13 @@
namespace sfl {
-AudioZrtpSession::AudioZrtpSession(SIPCall * sipcall, const std::string& zidFilename) :
- AudioRtpSession(sipcall, this, this),
- ost::TRTPSessionBase<ost::SymmetricRTPChannel, ost::SymmetricRTPChannel, ost::ZrtpQueue>(ost::InetHostAddress(sipcall->getLocalIp().c_str()),
- sipcall->getLocalAudioPort(),
- 0,
- ost::MembershipBookkeeping::defaultMembersHashSize,
- ost::defaultApplication()),
+AudioZrtpSession::AudioZrtpSession(SIPCall &call, const std::string& zidFilename) :
+ AudioRtpSession(call, *this, *this),
+ ost::TRTPSessionBase<ost::SymmetricRTPChannel, ost::SymmetricRTPChannel, ost::ZrtpQueue>(ost::InetHostAddress(call_.getLocalIp().c_str()),
+ call_.getLocalAudioPort(),
+ 0,
+ ost::MembershipBookkeeping::defaultMembersHashSize,
+ ost::defaultApplication()),
zidFilename_(zidFilename)
{
DEBUG("AudioZrtpSession initialized");
@@ -63,13 +63,14 @@ AudioZrtpSession::AudioZrtpSession(SIPCall * sipcall, const std::string& zidFile
setCancel(cancelDefault);
- DEBUG("AudioZrtpSession: Setting new RTP session with destination %s:%d", ca_->getLocalIp().c_str(), ca_->getLocalAudioPort());
+ DEBUG("AudioZrtpSession: Setting new RTP session with destination %s:%d",
+ call_.getLocalIp().c_str(), call_.getLocalAudioPort());
}
AudioZrtpSession::~AudioZrtpSession()
{
ost::Thread::terminate();
- Manager::instance().getMainBuffer()->unBindAll(ca_->getCallId());
+ Manager::instance().getMainBuffer()->unBindAll(call_.getCallId());
}
void AudioZrtpSession::final()
@@ -102,7 +103,7 @@ void AudioZrtpSession::initializeZid()
if (initialize(zidCompleteFilename.c_str()) >= 0) {
DEBUG("Register callbacks");
setEnableZrtp(true);
- setUserCallback(new ZrtpSessionCallback(ca_));
+ setUserCallback(new ZrtpSessionCallback(call_));
return;
}
@@ -129,17 +130,17 @@ void AudioZrtpSession::sendMicData()
timestamp_ += timestampIncrement_;
// this step is only needed for ZRTP
- queue_->putData(timestamp_, getMicDataEncoded(), compSize);
+ queue_.putData(timestamp_, getMicDataEncoded(), compSize);
// putData puts the data on RTP queue, sendImmediate bypasses this queue
- queue_->sendImmediate(timestamp_, getMicDataEncoded(), compSize);
+ queue_.sendImmediate(timestamp_, getMicDataEncoded(), compSize);
}
void AudioZrtpSession::run()
{
// Set recording sampling rate
- ca_->setRecordingSmplRate(getCodecSampleRate());
- DEBUG("AudioZrtpSession: Entering mainloop for call %s", ca_->getCallId().c_str());
+ call_.setRecordingSmplRate(getCodecSampleRate());
+ DEBUG("AudioZrtpSession: Entering mainloop for call %s", call_.getCallId().c_str());
uint32 timeout = 0;
@@ -169,7 +170,7 @@ void AudioZrtpSession::run()
setCancel(cancelImmediate);
timerTick();
} else {
- if (isPendingData(timeout/1000)) {
+ if (isPendingData(timeout / 1000)) {
setCancel(cancelDeferred);
if (isActive())
@@ -181,17 +182,15 @@ void AudioZrtpSession::run()
}
}
- DEBUG("AudioZrtpSession: Left main loop for call %s", ca_->getCallId().c_str());
+ DEBUG("AudioZrtpSession: Left main loop for call %s", call_.getCallId().c_str());
}
-
void AudioZrtpSession::incrementTimestampForDTMF()
{
timestamp_ += 160;
}
-
-void AudioZrtpSession::setSessionMedia(AudioCodec *audioCodec)
+void AudioZrtpSession::setSessionMedia(AudioCodec &audioCodec)
{
AudioRtpSession::setSessionMedia(audioCodec);
}
diff --git a/daemon/src/audio/audiortp/audio_zrtp_session.h b/daemon/src/audio/audiortp/audio_zrtp_session.h
index 00d9dad819d875b9b8854bcaf2c164a5ea214e7d..9e7c196e3dcf91646c91ac61ce8957cfe176995d 100644
--- a/daemon/src/audio/audiortp/audio_zrtp_session.h
+++ b/daemon/src/audio/audiortp/audio_zrtp_session.h
@@ -47,16 +47,17 @@ class AudioCodec;
namespace sfl {
-class ZrtpZidException: public std::runtime_error {
+class ZrtpZidException : public std::runtime_error {
public:
ZrtpZidException(const std::string& str = "") :
std::runtime_error("ZRTP ZID initialization failed." + str) {}
};
-// class AudioZrtpSession : public ost::TimerPort, public ost::SymmetricZRTPSession, public AudioRtpRecordHandler
-class AudioZrtpSession : public AudioRtpSession, protected ost::Thread, public ost::TRTPSessionBase<ost::SymmetricRTPChannel, ost::SymmetricRTPChannel, ost::ZrtpQueue> {
+class AudioZrtpSession :
+ public AudioRtpSession, protected ost::Thread,
+ public ost::TRTPSessionBase<ost::SymmetricRTPChannel, ost::SymmetricRTPChannel, ost::ZrtpQueue> {
public:
- AudioZrtpSession(SIPCall * sipcall, const std::string& zidFilename);
+ AudioZrtpSession(SIPCall &call, const std::string& zidFilename);
~AudioZrtpSession();
virtual void final();
@@ -64,7 +65,7 @@ class AudioZrtpSession : public AudioRtpSession, protected ost::Thread, public o
// Thread associated method
virtual void run();
- virtual bool onRTPPacketRecv(ost::IncomingRTPPkt& pkt) {
+ virtual bool onRTPPacketRecv(ost::IncomingRTPPkt &pkt) {
return AudioRtpSession::onRTPPacketRecv(pkt);
}
@@ -73,7 +74,7 @@ class AudioZrtpSession : public AudioRtpSession, protected ost::Thread, public o
void initializeZid();
std::string zidFilename_;
void incrementTimestampForDTMF();
- void setSessionMedia(AudioCodec *codec);
+ void setSessionMedia(AudioCodec &codec);
};
}
diff --git a/daemon/src/audio/audiortp/zrtp_session_callback.cpp b/daemon/src/audio/audiortp/zrtp_session_callback.cpp
index 270468a80ea2a8fc8e683d515610d31dce7d0a17..061cf1c64c22b66f270e80302085e149d5328de8 100644
--- a/daemon/src/audio/audiortp/zrtp_session_callback.cpp
+++ b/daemon/src/audio/audiortp/zrtp_session_callback.cpp
@@ -45,8 +45,8 @@ using namespace ost;
namespace sfl {
-ZrtpSessionCallback::ZrtpSessionCallback(SIPCall *sipcall) :
- sipcall_(sipcall)
+ZrtpSessionCallback::ZrtpSessionCallback(SIPCall &call) :
+ call_(call)
{
using std::pair;
using std::string;
@@ -112,21 +112,21 @@ void
ZrtpSessionCallback::secureOn(std::string cipher)
{
DEBUG("Zrtp: Secure mode is on with cipher %s", cipher.c_str());
- Manager::instance().getDbusManager()->getCallManager()->secureZrtpOn(sipcall_->getCallId(), cipher);
+ Manager::instance().getDbusManager()->getCallManager()->secureZrtpOn(call_.getCallId(), cipher);
}
void
ZrtpSessionCallback::secureOff()
{
DEBUG("Zrtp: Secure mode is off");
- Manager::instance().getDbusManager()->getCallManager()->secureZrtpOff(sipcall_->getCallId());
+ Manager::instance().getDbusManager()->getCallManager()->secureZrtpOff(call_.getCallId());
}
void
ZrtpSessionCallback::showSAS(std::string sas, bool verified)
{
DEBUG("Zrtp: SAS is: %s", sas.c_str());
- Manager::instance().getDbusManager()->getCallManager()->showSAS(sipcall_->getCallId(), sas, verified);
+ Manager::instance().getDbusManager()->getCallManager()->showSAS(call_.getCallId(), sas, verified);
}
@@ -134,7 +134,7 @@ void
ZrtpSessionCallback::zrtpNotSuppOther()
{
DEBUG("Zrtp: Callee does not support ZRTP");
- Manager::instance().getDbusManager()->getCallManager()->zrtpNotSuppOther(sipcall_->getCallId());
+ Manager::instance().getDbusManager()->getCallManager()->zrtpNotSuppOther(call_.getCallId());
}
@@ -163,13 +163,13 @@ ZrtpSessionCallback::zrtpNegotiationFailed(MessageSeverity severity, int subCode
std::map<int32, std::string>::const_iterator iter = zrtpMap_.find(subCode);
if (iter != zrtpMap_.end()) {
DEBUG("%s", iter->second.c_str());
- Manager::instance().getDbusManager()->getCallManager()->zrtpNegotiationFailed(sipcall_->getCallId(), iter->second, "ZRTP");
+ Manager::instance().getDbusManager()->getCallManager()->zrtpNegotiationFailed(call_.getCallId(), iter->second, "ZRTP");
}
} else {
std::map<int32, std::string>::const_iterator iter = severeMap_.find(subCode);
if (iter != severeMap_.end()) {
DEBUG("%s", iter->second.c_str());
- Manager::instance().getDbusManager()->getCallManager()->zrtpNegotiationFailed(sipcall_->getCallId(), iter->second, "severe");
+ Manager::instance().getDbusManager()->getCallManager()->zrtpNegotiationFailed(call_.getCallId(), iter->second, "severe");
}
}
}
@@ -178,7 +178,7 @@ void
ZrtpSessionCallback::confirmGoClear()
{
DEBUG("Zrtp: Received go clear message. Until confirmation, ZRTP won't send any data");
- Manager::instance().getDbusManager()->getCallManager()->zrtpNotSuppOther(sipcall_->getCallId());
+ Manager::instance().getDbusManager()->getCallManager()->zrtpNotSuppOther(call_.getCallId());
}
std::map<int32, std::string> ZrtpSessionCallback::infoMap_;
diff --git a/daemon/src/audio/audiortp/zrtp_session_callback.h b/daemon/src/audio/audiortp/zrtp_session_callback.h
index 08bb9cf4fb4c9bffd2f9fbc49d50520dda88a6c3..e9e92a9c2cda8c02dec10a562bb9831dd410cef4 100644
--- a/daemon/src/audio/audiortp/zrtp_session_callback.h
+++ b/daemon/src/audio/audiortp/zrtp_session_callback.h
@@ -28,8 +28,8 @@
* as that of the covered work.
*/
-#ifndef __ZRTP_SESSION_CALLBACK_H__
-#define __ZRTP_SESSION_CALLBACK_H__
+#ifndef ZRTP_SESSION_CALLBACK_H_
+#define ZRTP_SESSION_CALLBACK_H_
#include <cstddef>
using std::ptrdiff_t;
@@ -39,7 +39,6 @@ using std::ptrdiff_t;
#include <libzrtpcpp/ZrtpUserCallback.h>
#include <exception>
#include <map>
-#include "noncopyable.h"
class SIPCall;
@@ -47,7 +46,7 @@ namespace sfl {
class ZrtpSessionCallback: public ZrtpUserCallback {
public:
- ZrtpSessionCallback(SIPCall *sipcall);
+ ZrtpSessionCallback(SIPCall &call);
void secureOn(std::string cipher);
void secureOff();
@@ -58,12 +57,11 @@ class ZrtpSessionCallback: public ZrtpUserCallback {
void confirmGoClear();
private:
- NON_COPYABLE(ZrtpSessionCallback);
- SIPCall* sipcall_;
+ SIPCall &call_;
static std::map<int32, std::string> infoMap_;
static std::map<int32, std::string> warningMap_;
static std::map<int32, std::string> severeMap_;
static std::map<int32, std::string> zrtpMap_;
};
}
-#endif // __ZRTP_SESSION_CALLBACK_H__
+#endif // ZRTP_SESSION_CALLBACK_H_