From 8b770d0f5cf55275965a8d66344169c8845a3790 Mon Sep 17 00:00:00 2001
From: asavard <asavard@asavard-KT378AA-A2L-a6552f.(none)>
Date: Tue, 26 Oct 2010 18:56:04 -0400
Subject: [PATCH] [#4367] Fix AudioRtpclass using TRTPSessionBase
---
.../src/audio/audiortp/AudioRtpSession.cpp | 15 ++++++++++-----
.../src/audio/audiortp/AudioRtpSession.h | 3 ++-
2 files changed, 12 insertions(+), 6 deletions(-)
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
index d0199adf9b..f46471e8a1 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
@@ -42,7 +42,12 @@ namespace sfl
{
AudioRtpSession::AudioRtpSession (ManagerImpl * manager, SIPCall * sipcall) :
- ost::SymmetricRTPSession (ost::InetHostAddress (sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort()),
+ // ost::SymmetricRTPSession (ost::InetHostAddress (sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort()),
+ TRTPSessionBase<ost::DualRTPUDPIPv4Channel,ost::DualRTPUDPIPv4Channel,ost::AVPQueue>(ost::InetHostAddress (sipcall->getLocalIp().c_str()),
+ sipcall->getLocalAudioPort(),
+ 0,
+ ost::MembershipBookkeeping::defaultMembersHashSize,
+ ost::defaultApplication()),
AudioRtpRecordHandler(manager, sipcall)
, _time (new ost::Time())
, _mainloopSemaphore (0)
@@ -53,7 +58,7 @@ AudioRtpSession::AudioRtpSession (ManagerImpl * manager, SIPCall * sipcall) :
, _countNotificationTime (0)
, _ca (sipcall)
{
- this->setCancel (cancelDefault);
+ static_cast<ost::Thread *>(this)->setCancel (cancelDefault);
assert (_ca);
@@ -66,7 +71,7 @@ AudioRtpSession::~AudioRtpSession()
_debug ("AudioRtpSession: Delete AudioRtpSession instance");
try {
- this->terminate();
+ static_cast<ost::Thread *>(this)->terminate();
} catch (...) {
_debugException ("AudioRtpSession: Thread destructor didn't terminate correctly");
throw;
@@ -266,7 +271,7 @@ int AudioRtpSession::startRtpThread (AudioCodec* audiocodec)
setSessionTimeouts();
setSessionMedia (audiocodec);
initBuffers();
- int ret = this->start (_mainloopSemaphore);
+ int ret = static_cast<ost::Thread *>(this)->start (_mainloopSemaphore);
this->startRunning();
return ret;
}
@@ -299,7 +304,7 @@ void AudioRtpSession::run ()
_debug ("AudioRtpSession: Entering mainloop for call %s",_ca->getCallId().c_str());
- while (!testCancel()) {
+ while (!static_cast<ost::Thread *>(this)->testCancel()) {
// Reset timestamp to make sure the timing information are up to date
if (_timestampCount > RTP_TIMESTAMP_RESET_FREQ) {
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
index 272d2b68ca..0f3084b153 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -57,7 +57,8 @@ namespace sfl
{
// class AudioRtpSession : public ost::Thread, public ost::TimerPort, public AudioRtpRecordHandler, public ost::SymmetricRTPSession
-class AudioRtpSession : public ost::TimerPort, public ost::SymmetricRTPSession, public AudioRtpRecordHandler
+// class AudioRtpSession : public ost::Thread, public ost::TimerPort, public ost::SymmetricRTPSession, public AudioRtpRecordHandler
+class AudioRtpSession : public ost::Thread, public ost::TimerPort, public AudioRtpRecordHandler, public ost::TRTPSessionBase<ost::DualRTPUDPIPv4Channel,ost::DualRTPUDPIPv4Channel,ost::AVPQueue>
{
public:
/**
--
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