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Commit 453dde29 authored by alexandresavard's avatar alexandresavard
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Merge branch 'master' into recording

parents 1e70d8cc 59232af1
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configure.ac
View file @ 453dde29
......@@ -2,7 +2,7 @@ dnl SFLPhone - configure.ac for automake 1.9 and autoconf 2.59
dnl
dnl Process this file with autoconf to produce a configure script.
AC_PREREQ(2.59)
AC_INIT([SFLPhone],[0.9.2-7],[sflphoneteam@savoirfairelinux.com],[sflphone])
AC_INIT([SFLPhone],[0.9.3],[sflphoneteam@savoirfairelinux.com],[sflphone])
AC_COPYRIGHT([[Copyright (c) Savoir-Faire Linux 2004-2009]])
AC_REVISION([$Revision$])
......
debian/changelog
View file @ 453dde29
sflphone (0.9.2-2ubuntu8) intrepid; urgency=low
[ Emmanuel Milou ]
* Update changelogs
* Fix bug in merge and in Makefile.am
* Terminate only one call
* Disable PJsip shutdown when changing STUN parameters
* Function terminateSIPCall added in sipvoiplink and managerimpl
* Add a timer to the alsa thread to not jam the CPU load
* Fix bug in sipvoiplink.cpp
* Clean shutdown of pulseaudio on quiting
* Fix DTMF at first start with Pulseaudio
* Remove zeroconf from the build system
* Add a library manager + exception handling
* Clean up in the working directory
* Better handling of capture XRUNs
* Restore mic adjust volume on ALSA layer
* Protect device ALSA operation if not opened
* Fix the switching layer bug
* Use dynamic_cast<> to use audiolayer-specific methods
* Open the audio devices only once at startup
* Refactoring of the ALSA part
* Functional plug-in manager
* Use a C++ thread to handle tones and DTMF in ALSA
* Restore IAXVoIPLink, restore Mutex
* Make the plugins registering against the plugin manager
* Migrate to 1->N relationship between voiplink and accounts
* API plugin for registration
* Use C++ thread in SIP, move everything in sipvoiplink
* Complete singleton pattern for the plugin manager
* Add -Wno-return-type compilation flag to remove warnings; Update
version number in configure.ac
* Add the dynamic loading for the plugin framework; integate unit
* Fix jaunty control file dependency problems
tests
[ Yun Liu ]
* Update rpm spec file
* modify build package script and spec file for suse
[ Alexandre Savard ]
* Add audiorecorder plugin and testaudiorecorder
* Add audio Recording class, edit global.h
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
sflphone (0.9.2-2ubuntu7) intrepid; urgency=low
[ Emmanuel Milou ]
* Update changelog to 0.9.2-6
* Fix some dbus-glib implementation details on the client side
* Init history after dbus initialization
* Add error checking in useragent; Clean sipvoiplink
* Prevent crash when trying to call an empty number
* Set the volume of the playback stream to PA_VOLUME_NORM at startup
* Fix GTK+ generic value double initialization
* Fix jaunty control file dependency problems
* Fix jaunty control file dependency problems
[ Yun Liu ]
* Fix bug ticket # 137
* Tolerant to gsm library of OpenSuse 11
[ Sven Werlen ]
* Update german translation
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
sflphone (0.9.2-2ubuntu6) intrepid; urgency=low
[ Emmanuel Milou ]
......
debian/changelog.hardy
View file @ 453dde29
sflphone (0.9.2-2ubuntu6) intrepid; urgency=low
sflphone (0.9.2-2ubuntu8) hardy; urgency=low
[ Emmanuel Milou ]
* Update changelogs
* Fix bug in merge and in Makefile.am
* Terminate only one call
* Disable PJsip shutdown when changing STUN parameters
* Function terminateSIPCall added in sipvoiplink and managerimpl
* Add a timer to the alsa thread to not jam the CPU load
* Fix bug in sipvoiplink.cpp
* Clean shutdown of pulseaudio on quiting
* Fix DTMF at first start with Pulseaudio
* Remove zeroconf from the build system
* Add a library manager + exception handling
* Clean up in the working directory
* Better handling of capture XRUNs
* Restore mic adjust volume on ALSA layer
* Protect device ALSA operation if not opened
* Fix the switching layer bug
* Use dynamic_cast<> to use audiolayer-specific methods
* Open the audio devices only once at startup
* Refactoring of the ALSA part
* Functional plug-in manager
* Use a C++ thread to handle tones and DTMF in ALSA
* Restore IAXVoIPLink, restore Mutex
* Make the plugins registering against the plugin manager
* Migrate to 1->N relationship between voiplink and accounts
* API plugin for registration
* Use C++ thread in SIP, move everything in sipvoiplink
* Complete singleton pattern for the plugin manager
* Add -Wno-return-type compilation flag to remove warnings; Update
version number in configure.ac
* Add the dynamic loading for the plugin framework; integate unit
* Fix jaunty control file dependency problems
tests
[ Yun Liu ]
* Update rpm spec file
* modify build package script and spec file for suse
[ Alexandre Savard ]
* Add audiorecorder plugin and testaudiorecorder
* Add audio Recording class, edit global.h
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
sflphone (0.9.2-2ubuntu7) hardy; urgency=low
[ Emmanuel Milou ]
* Update changelog to 0.9.2-6
* Fix some dbus-glib implementation details on the client side
* Init history after dbus initialization
* Add error checking in useragent; Clean sipvoiplink
* Prevent crash when trying to call an empty number
* Set the volume of the playback stream to PA_VOLUME_NORM at startup
* Fix GTK+ generic value double initialization
* Fix jaunty control file dependency problems
* Fix jaunty control file dependency problems
[ Yun Liu ]
* Fix bug ticket # 137
* Tolerant to gsm library of OpenSuse 11
[ Sven Werlen ]
* Update german translation
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
sflphone (0.9.2-2ubuntu6) hardy; urgency=low
[ Emmanuel Milou ]
* Migrate STUN configuration to the main config window
......
debian/changelog.intrepid
View file @ 453dde29
sflphone (0.9.2-2ubuntu8) intrepid; urgency=low
[ Emmanuel Milou ]
* Update changelogs
* Fix bug in merge and in Makefile.am
* Terminate only one call
* Disable PJsip shutdown when changing STUN parameters
* Function terminateSIPCall added in sipvoiplink and managerimpl
* Add a timer to the alsa thread to not jam the CPU load
* Fix bug in sipvoiplink.cpp
* Clean shutdown of pulseaudio on quiting
* Fix DTMF at first start with Pulseaudio
* Remove zeroconf from the build system
* Add a library manager + exception handling
* Clean up in the working directory
* Better handling of capture XRUNs
* Restore mic adjust volume on ALSA layer
* Protect device ALSA operation if not opened
* Fix the switching layer bug
* Use dynamic_cast<> to use audiolayer-specific methods
* Open the audio devices only once at startup
* Refactoring of the ALSA part
* Functional plug-in manager
* Use a C++ thread to handle tones and DTMF in ALSA
* Restore IAXVoIPLink, restore Mutex
* Make the plugins registering against the plugin manager
* Migrate to 1->N relationship between voiplink and accounts
* API plugin for registration
* Use C++ thread in SIP, move everything in sipvoiplink
* Complete singleton pattern for the plugin manager
* Add -Wno-return-type compilation flag to remove warnings; Update
version number in configure.ac
* Add the dynamic loading for the plugin framework; integate unit
* Fix jaunty control file dependency problems
tests
[ Yun Liu ]
* Update rpm spec file
* modify build package script and spec file for suse
[ Alexandre Savard ]
* Add audiorecorder plugin and testaudiorecorder
* Add audio Recording class, edit global.h
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
sflphone (0.9.2-2ubuntu7) intrepid; urgency=low
[ Emmanuel Milou ]
* Update changelog to 0.9.2-6
* Fix some dbus-glib implementation details on the client side
* Init history after dbus initialization
* Add error checking in useragent; Clean sipvoiplink
* Prevent crash when trying to call an empty number
* Set the volume of the playback stream to PA_VOLUME_NORM at startup
* Fix GTK+ generic value double initialization
* Fix jaunty control file dependency problems
* Fix jaunty control file dependency problems
[ Yun Liu ]
* Fix bug ticket # 137
* Tolerant to gsm library of OpenSuse 11
[ Sven Werlen ]
* Update german translation
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
sflphone (0.9.2-2ubuntu6) intrepid; urgency=low
[ Emmanuel Milou ]
......
debian/changelog.jaunty
View file @ 453dde29
sflphone (0.9.2-2ubuntu6) intrepid; urgency=low
sflphone (0.9.2-2ubuntu8) jaunty; urgency=low
[ Emmanuel Milou ]
* Update changelogs
* Fix bug in merge and in Makefile.am
* Terminate only one call
* Disable PJsip shutdown when changing STUN parameters
* Function terminateSIPCall added in sipvoiplink and managerimpl
* Add a timer to the alsa thread to not jam the CPU load
* Fix bug in sipvoiplink.cpp
* Clean shutdown of pulseaudio on quiting
* Fix DTMF at first start with Pulseaudio
* Remove zeroconf from the build system
* Add a library manager + exception handling
* Clean up in the working directory
* Better handling of capture XRUNs
* Restore mic adjust volume on ALSA layer
* Protect device ALSA operation if not opened
* Fix the switching layer bug
* Use dynamic_cast<> to use audiolayer-specific methods
* Open the audio devices only once at startup
* Refactoring of the ALSA part
* Functional plug-in manager
* Use a C++ thread to handle tones and DTMF in ALSA
* Restore IAXVoIPLink, restore Mutex
* Make the plugins registering against the plugin manager
* Migrate to 1->N relationship between voiplink and accounts
* API plugin for registration
* Use C++ thread in SIP, move everything in sipvoiplink
* Complete singleton pattern for the plugin manager
* Add -Wno-return-type compilation flag to remove warnings; Update
version number in configure.ac
* Add the dynamic loading for the plugin framework; integate unit
* Fix jaunty control file dependency problems
tests
[ Yun Liu ]
* Update rpm spec file
* modify build package script and spec file for suse
[ Alexandre Savard ]
* Add audiorecorder plugin and testaudiorecorder
* Add audio Recording class, edit global.h
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
sflphone (0.9.2-2ubuntu7) jaunty; urgency=low
[ Emmanuel Milou ]
* Update changelog to 0.9.2-6
* Fix some dbus-glib implementation details on the client side
* Init history after dbus initialization
* Add error checking in useragent; Clean sipvoiplink
* Prevent crash when trying to call an empty number
* Set the volume of the playback stream to PA_VOLUME_NORM at startup
* Fix GTK+ generic value double initialization
* Fix jaunty control file dependency problems
* Fix jaunty control file dependency problems
[ Yun Liu ]
* Fix bug ticket # 137
* Tolerant to gsm library of OpenSuse 11
[ Sven Werlen ]
* Update german translation
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
sflphone (0.9.2-2ubuntu6) jaunty; urgency=low
[ Emmanuel Milou ]
* Migrate STUN configuration to the main config window
......
sflphone-gtk/configure.ac
View file @ 453dde29
AC_INIT([SFLphone],[0.9.2-7],[sflphoneteam@savoirfairelinux.com],[sflphone])
AC_INIT([SFLphone],[0.9.3],[sflphoneteam@savoirfairelinux.com],[sflphone])
AM_CONFIG_HEADER(config.h)
PACKAGE=SFLphone
VERSION=0.9.2-7
VERSION=0.9.3
AM_INIT_AUTOMAKE($PACKAGE,$VERSION)
......@@ -15,7 +15,7 @@ AC_ARG_WITH(debug,
[with_debug=no]
)
if test "x$with_debug" = "xfull" -o "x$with_debug" = "xyes"; then
CFLAGS="$CFLAGS -g -DDEBUG -Wall -Wextra"
CFLAGS="$CFLAGS -g -O2 -DDEBUG -Wall -Wextra"
else
CFLAGS="$CFLAGS -g -Wall -O2"
fi
......
sflphone-gtk/src/Makefile.am
View file @ 453dde29
......@@ -43,8 +43,8 @@ EXTRA_DIST = marshaller.list
sflphone_gtk_LDADD = $(DEPS_LIBS) $(NOTIFY_LIBS) $(SEXY_LIBS)
AM_CPPFLAGS = $(DEPS_CFLAGS) \
-DICONS_DIR=\""/usr/share/sflphone"\" \
-DCODECS_DIR=\""/usr/lib/sflphone/codecs"\"
-DICONS_DIR=\""$(prefix)/share/sflphone"\" \
-DCODECS_DIR=\""$(prefix)/lib/sflphone/codecs"\"
# add symbolic link
install-exec-local:
......
sflphone-gtk/src/actions.h
View file @ 453dde29
......@@ -165,4 +165,7 @@ void sflphone_set_current_account();
*/
void sflphone_fill_codec_list();
void sflphone_record (call_t *c);
void sflphone_rec_call (void);
#endif
sflphone-gtk/src/configwindow.c
View file @ 453dde29
......@@ -184,8 +184,9 @@ clean_history( void )
select_account(GtkTreeSelection *selection, GtkTreeModel *model)
{
GtkTreeIter iter;
GValue val = {0};
GValue val;
memset (&val, 0, sizeof(val));
if (!gtk_tree_selection_get_selected(selection, &model, &iter))
{
selectedAccount = NULL;
......
sflphone-gtk/src/dbus.c
View file @ 453dde29
......@@ -178,7 +178,7 @@ error_alert(DBusGProxy *proxy UNUSED,
}
static void nameOwnerChanged(DBusGProxy *proxy, char *name, char *old_owner, char *new_owner, gpointer data )
static void nameOwnerChanged(DBusGProxy *proxy UNUSED, char *name , char *old_owner, char *new_owner, gpointer data UNUSED)
{
g_print("******************************************************************\n");
......@@ -1292,8 +1292,12 @@ dbus_set_record(const call_t * c)
org_sflphone_SFLphone_CallManager_set_recording (
callManagerProxy,
c->callID,
error);
g_print("called dbus_set_record on CallManager\n");
&error);
if(error)
{
g_error_free(error);
}
g_print("called dbus_set_record on CallManager\n");
}
......@@ -1588,10 +1592,10 @@ void dbus_set_stun_server( gchar* server)
}
}
guint dbus_stun_is_enabled (void)
gint dbus_stun_is_enabled (void)
{
GError* error = NULL;
guint stun;
gint stun;
org_sflphone_SFLphone_ConfigurationManager_is_stun_enabled(
configurationManagerProxy,
&stun,
......
sflphone-gtk/src/dbus.h
View file @ 453dde29
......@@ -432,7 +432,7 @@ guint dbus_get_sip_port();
gchar* dbus_get_stun_server (void);
void dbus_set_stun_server( gchar* server);
guint dbus_stun_is_enabled (void);
gint dbus_stun_is_enabled (void);
void dbus_enable_stun (void);
void dbus_set_record (const call_t * c);
......
sflphone-gtk/src/historyfilter.c
View file @ 453dde29
......@@ -17,6 +17,8 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include <string.h>
#include <historyfilter.h>
#include <calltree.h>
......@@ -35,9 +37,11 @@ is_visible(GtkTreeModel* model, GtkTreeIter* iter, gpointer data UNUSED)
{
if( SHOW_SEARCHBAR )
{
GValue val = {0,};
GValue val;
gchar* text = NULL;
gchar* search = (gchar*)gtk_entry_get_text(GTK_ENTRY(filter_entry));
memset (&val, 0, sizeof(val));
gtk_tree_model_get_value(GTK_TREE_MODEL(model), iter, 1, &val);
if(G_VALUE_HOLDS_STRING(&val)){
text = (gchar *)g_value_get_string(&val);
......@@ -45,6 +49,7 @@ is_visible(GtkTreeModel* model, GtkTreeIter* iter, gpointer data UNUSED)
if(text != NULL && g_ascii_strncasecmp(search, _("Search"), 6) != 0){
return g_regex_match_simple(search, text, G_REGEX_CASELESS, 0);
}
g_value_unset (&val);
return TRUE;
}
return TRUE;
......
sflphone-gtk/src/mainwindow.c
View file @ 453dde29
......@@ -59,25 +59,28 @@ on_delete (GtkWidget * widget UNUSED, gpointer data UNUSED)
/** Ask the user if he wants to hangup current calls */
gboolean
main_window_ask_quit(){
guint count = call_list_get_size(current_calls);
GtkWidget * dialog;
gint response;
gchar * question;
guint count = call_list_get_size(current_calls);
GtkWidget * dialog;
gint response;
gchar * question;
if(count == 1)
{
question = _("<b>There is one call in progress.</b>\nDo you still want to quit?");
question = _("There is one call in progress.");
}
else
{
question = _("<b>There are calls in progress.</b>\nDo you still want to quit?");
question = _("There are calls in progress.");
}
dialog = gtk_message_dialog_new_with_markup (GTK_WINDOW(window) ,
GTK_DIALOG_MODAL,
GTK_MESSAGE_QUESTION,
GTK_BUTTONS_YES_NO,
question);
"<b>%s<b>\n%s",
question,
_("Do you still want to quit?") );
response = gtk_dialog_run (GTK_DIALOG (dialog));
......@@ -209,6 +212,7 @@ main_window_message(GtkMessageType type, gchar * markup){
GTK_DIALOG_MODAL | GTK_DIALOG_DESTROY_WITH_PARENT,
type,
GTK_BUTTONS_CLOSE,
"%s\n",
markup);
gtk_dialog_run (GTK_DIALOG(dialog));
......
sflphone-gtk/src/menus.c
View file @ 453dde29
......@@ -492,25 +492,11 @@ edit_paste ( void * foo UNUSED)
}
static void
clear_history( void* foo UNUSED)
clear_history (void)
{
gchar *markup;
GtkWidget *dialog;
int response;
if( call_list_get_size( history ) == 0 ){
markup = g_markup_printf_escaped(_("History empty"));
dialog = gtk_message_dialog_new_with_markup ( GTK_WINDOW(get_main_window()),
GTK_DIALOG_DESTROY_WITH_PARENT,
GTK_MESSAGE_INFO,
GTK_BUTTONS_CLOSE,
markup);
response = gtk_dialog_run (GTK_DIALOG(dialog));
gtk_widget_destroy (GTK_WIDGET(dialog));
}
else{
if( call_list_get_size( history ) != 0 ){
call_list_clean_history();
}
}
}
GtkWidget *
......
src/Makefile.am
View file @ 453dde29
......@@ -47,7 +47,6 @@ sflphoned_LDADD = \
@CCRTP_LIBS@ \
@ALSA_LIBS@ \
@PULSEAUDIO_LIBS@ \
-luuid \
@SAMPLERATE_LIBS@
#sflphoned_LDFLAGS=-pg
......
src/audio/audiortp.cpp
View file @ 453dde29
......@@ -464,6 +464,7 @@ AudioRtpRTX::run () {
//_debugException("* ARTP Action: Stop");
//throw;
//}
}
......
src/managerimpl.cpp
View file @ 453dde29
......@@ -1990,32 +1990,6 @@ void ManagerImpl::setAccountDetails( const std::string& accountID, const std::ma
setConfig(accountID, HOSTNAME, (*details.find(HOSTNAME)).second);
setConfig(accountID, CONFIG_ACCOUNT_MAILBOX,(*details.find(CONFIG_ACCOUNT_MAILBOX)).second);
// SIP SPECIFIC
/*
if (accountType == "SIP") {
link = Manager::instance().getAccountLink( accountID );
if( link==0 )
{
_debug("Can not retrieve SIP link...\n");
return;
}
setConfig(accountID, SIP_STUN_SERVER,(*details.find(SIP_STUN_SERVER)).second);
setConfig(accountID, SIP_USE_STUN, (*details.find(SIP_USE_STUN)).second == "TRUE" ? "1" : "0");
if((*details.find(SIP_USE_STUN)).second == "TRUE")
{
link->setStunServer((*details.find(SIP_STUN_SERVER)).second.data());
}
else
{
link->setStunServer("");
}
restartPJSIP();
}*/
saveConfig();
acc = getAccount(accountID);
......@@ -2030,8 +2004,6 @@ void ManagerImpl::setAccountDetails( const std::string& accountID, const std::ma
// Update account details to the client side
if (_dbus) _dbus->getConfigurationManager()->accountsChanged();
//delete link; link=0;
}
void
......
src/sipvoiplink.cpp
View file @ 453dde29
......@@ -539,7 +539,6 @@ SIPVoIPLink::peerHungup(const CallID& id)
call->getInvSession()->mod_data[getModId()] = NULL;
// Release RTP thread
if (Manager::instance().isCurrentCall(id)) {
_debug("* SIP Info: Stopping AudioRTP for hangup\n");
......
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