From f8bc445a28ed8698e9dab9c4efb744a3ece4e660 Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
Date: Fri, 25 Feb 2011 22:20:07 -0500
Subject: [PATCH] [#5211] Move mainbuffer unbind call in rtp final method
---
.../src/audio/audiortp/AudioRtpFactory.cpp | 2 +-
.../src/audio/audiortp/AudioRtpSession.cpp | 10 +++++-----
.../src/audio/audiortp/AudioRtpSession.h | 4 +++-
sflphone-common/src/audio/pulseaudio/pulselayer.cpp | 11 -----------
sflphone-common/src/global.h | 2 +-
sflphone-common/src/sip/sipcall.cpp | 4 ++--
sflphone-common/src/sip/sipvoiplink.cpp | 13 +++++++------
7 files changed, 19 insertions(+), 27 deletions(-)
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
index ab02df4ecc..84d44d91ea 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpFactory.cpp
@@ -203,7 +203,7 @@ void AudioRtpFactory::stop (void)
break;
case Symmetric:
- static_cast<AudioSrtpSession *> (_rtpSession)->stopRtpThread();
+ static_cast<AudioRtpSession *> (_rtpSession)->stopRtpThread();
break;
case Zrtp:
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
index a61a6165ee..954e406c7e 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
@@ -81,13 +81,15 @@ AudioRtpSession::~AudioRtpSession()
_debugException ("AudioRtpSession: Thread destructor didn't terminate correctly");
throw;
}
-
- Manager::instance().getMainBuffer()->unBindAll (_audioRtpRecord._callId);
}
void AudioRtpSession::final()
{
- delete this;
+ _debug ("AudioRtpSession: Finalize AudioRtpSession instance");
+
+ Manager::instance().getMainBuffer()->unBindAll (_audioRtpRecord._callId);
+
+ delete static_cast<AudioRtpSession *> (this);
}
void AudioRtpSession::setSessionTimeouts (void)
@@ -403,8 +405,6 @@ void AudioRtpSession::run ()
_debug ("AudioRtpSession: Left main loop for call %s", _audioRtpRecord._callId.c_str());
- Thread::exit();
-
}
}
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
index 6e495d29f8..7461d7027b 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -71,6 +71,8 @@ class AudioRtpSession : protected ost::Thread, public ost::TimerPort, public Aud
virtual void final ();
+ void terminateRtpSession();
+
// Thread associated method
virtual void run ();
@@ -137,7 +139,7 @@ class AudioRtpSession : protected ost::Thread, public ost::TimerPort, public Aud
// optional in ost::thread, then
// it amounts to the same as doing
// start() with no semaphore at all.
- ost::Semaphore * _mainloopSemaphore;
+ ost::Semaphore *_mainloopSemaphore;
// Main destination address for this rtp session.
// Stored in case or reINVITE, which may require to forget
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index c360756ee2..6aa3b226e1 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -249,10 +249,6 @@ PulseLayer::PulseLayer (ManagerImpl* manager)
byteCounter = 0;
-
- // captureFile = new ofstream ("probeCaptureFile", ofstream::binary);
- // spkrFile = new ofstream ("probeSpkrFile", ofstream::binary);
-
openLayer();
}
@@ -271,13 +267,6 @@ PulseLayer::~PulseLayer (void)
delete AudioLayer::_audiofilter;
AudioLayer::_audiofilter = NULL;
-
-
- // captureFile->close();
- // spkrFile->close();
-
- // delete captureFile;
- // delete spkrFile;
}
void
diff --git a/sflphone-common/src/global.h b/sflphone-common/src/global.h
index f69f1898ac..d69aea955e 100644
--- a/sflphone-common/src/global.h
+++ b/sflphone-common/src/global.h
@@ -44,7 +44,7 @@
#include <vector>
#include "logger.h"
-#define SFLPHONED_VERSION "0.9.8" /** Version number */
+#define SFLPHONED_VERSION "0.9.12" /** Version number */
#define HOMEDIR (getenv ("HOME")) /** Home directory */
#define XDG_DATA_HOME (getenv ("XDG_DATA_HOME"))
diff --git a/sflphone-common/src/sip/sipcall.cpp b/sflphone-common/src/sip/sipcall.cpp
index 8e88ee7edd..17476ad1e4 100644
--- a/sflphone-common/src/sip/sipcall.cpp
+++ b/sflphone-common/src/sip/sipcall.cpp
@@ -65,8 +65,8 @@ SIPCall::~SIPCall()
delete _local_sdp;
_local_sdp = NULL;
- _error ("SDP: pool capacity %d", pj_pool_get_capacity (_pool));
- _error ("SDP: pool size %d", pj_pool_get_used_size (_pool));
+ _debug ("SDP: pool capacity %d", pj_pool_get_capacity (_pool));
+ _debug ("SDP: pool size %d", pj_pool_get_used_size (_pool));
// Release memory allocated for SDP only once
pj_pool_release (_pool);
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index b727efac33..31aac9ac76 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -473,7 +473,6 @@ int SIPVoIPLink::sendRegister (AccountID id)
int expire_value;
pj_status_t status;
- pj_str_t useragent;
pjsip_tx_data *tdata;
pjsip_host_info destination;
@@ -636,7 +635,9 @@ int SIPVoIPLink::sendRegister (AccountID id)
// Add User-Agent Header
pj_list_init (&hdr_list);
- useragent = pj_str ( (char*) get_useragent_name (id).c_str());
+ const char *useragent_name = get_useragent_name (id).c_str();
+ pj_str_t useragent = pj_str ( (char *) useragent_name);
+ // pj_str_t useragent = pj_str ( (char *) "SFLphone"); // { (char *) "SFLphone", 8};
h = pjsip_generic_string_hdr_create (_pool, &STR_USER_AGENT, &useragent);
@@ -771,8 +772,8 @@ SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl) throw
_debug ("UserAgent: New outgoing call %s to %s", id.c_str(), toUrl.c_str());
- _error ("UserAgent: pool capacity %d", pj_pool_get_capacity (_pool));
- _error ("UserAgent: pool size %d", pj_pool_get_used_size (_pool));
+ _debug ("UserAgent: pool capacity %d", pj_pool_get_capacity (_pool));
+ _debug ("UserAgent: pool size %d", pj_pool_get_used_size (_pool));
SIPCall* call = new SIPCall (id, Call::Outgoing, _cp);
@@ -864,8 +865,8 @@ SIPVoIPLink::answer (const CallID& id)
SIPCall *call = getSIPCall (id);
- _error ("UserAgent: pool capacity %d", pj_pool_get_capacity (_pool));
- _error ("UserAgent: pool size %d", pj_pool_get_used_size (_pool));
+ _debug ("UserAgent: pool capacity %d", pj_pool_get_capacity (_pool));
+ _debug ("UserAgent: pool size %d", pj_pool_get_used_size (_pool));
if (call==0) {
--
GitLab