diff --git a/autogen.sh b/autogen.sh
index 47c9baa2215201ef4b04c53cd94067e69b7e784b..e63f71ddd78137734c54388664397300f863b2bd 100755
--- a/autogen.sh
+++ b/autogen.sh
@@ -6,4 +6,3 @@ libtoolize --force
autoheader
autoconf -f
automake -a
-./configure --prefix=/usr --with-debug
diff --git a/configure.ac b/configure.ac
index 30a2ac659b51a66e3f869079e3c196dd7f4bc81f..e58406ac18f0d74aa236e904d60132de1d0b24eb 100644
--- a/configure.ac
+++ b/configure.ac
@@ -135,8 +135,6 @@ AC_CHECK_FUNCS([bzero floor gethostbyname gethrtime gettimeofday \
dnl *****************************************
dnl Check for header files
dnl *****************************************
-dnl Check for exosip2
-LP_CHECK_EXOSIP2
SFLPHONE_LIBS="$SFLPHONE_LIBS"
dnl Check for GNU ccRTP
diff --git a/globals.mak b/globals.mak
index 41cc38fb63fc2017f0b12a6a8b67ce723fcaef3f..3ed71e1808ed32ba21f56c777fc33a92964c9e2a 100644
--- a/globals.mak
+++ b/globals.mak
@@ -8,6 +8,7 @@ PJSIP_LIBS = -lpjnath -lpjsua -lpjsip -lpjmedia -lpjsip-simple -lpjsip-ua -lpjme
AM_CPPFLAGS = \
-I$(src)/libs \
-I$(src)/src \
+ -I$(src)/test \
-DPREFIX=\"$(prefix)\" \
-DPROGSHAREDIR=\"${datadir}/sflphone\" \
$(ZEROCONFFLAGS) \
diff --git a/sflphone-gtk/autogen.sh b/sflphone-gtk/autogen.sh
index 47c9baa2215201ef4b04c53cd94067e69b7e784b..e63f71ddd78137734c54388664397300f863b2bd 100755
--- a/sflphone-gtk/autogen.sh
+++ b/sflphone-gtk/autogen.sh
@@ -6,4 +6,3 @@ libtoolize --force
autoheader
autoconf -f
automake -a
-./configure --prefix=/usr --with-debug
diff --git a/sflphone-gtk/src/accountlist.c b/sflphone-gtk/src/accountlist.c
index 8e9654eb8661b2bfc612f9a30d0c205b89739094..893d057d24ed2776acbc99cc65c8e33617b0ed53 100644
--- a/sflphone-gtk/src/accountlist.c
+++ b/sflphone-gtk/src/accountlist.c
@@ -223,3 +223,44 @@ account_list_get_registered_accounts( void )
g_print(" %d registered accounts \n" , res );
return res;
}
+
+gchar* account_list_get_current_id( void ){
+ if( __CURRENT_ACCOUNT_ID == NULL )
+ return "";
+ else
+ return __CURRENT_ACCOUNT_ID;
+}
+
+int account_list_get_sip_account_number( void ){
+
+ int n;
+ guint size, i;
+ account_t *current;
+
+ size = account_list_get_size();
+ n = 0;
+ for( i=0; i<size ;i++ ){
+ current = account_list_get_nth( i );
+ if( strcmp(g_hash_table_lookup(current->properties, ACCOUNT_TYPE), "SIP" ) == 0 )
+ n++;
+ }
+
+ return n;
+}
+
+int account_list_get_iax_account_number( void ){
+
+ int n;
+ guint size, i;
+ account_t *current;
+
+ size = account_list_get_size();
+ n = 0;
+ for( i=0; i<size ;i++ ){
+ current = account_list_get_nth( i );
+ if( strcmp(g_hash_table_lookup(current->properties, ACCOUNT_TYPE), "IAX" ) == 0 )
+ n++;
+ }
+
+ return n;
+}
diff --git a/sflphone-gtk/src/accountlist.h b/sflphone-gtk/src/accountlist.h
index 70756bfad130458da41f4cfdad4917f3b65daf52..18b5575d87504af68f280cdc15c35db0cbada95c 100644
--- a/sflphone-gtk/src/accountlist.h
+++ b/sflphone-gtk/src/accountlist.h
@@ -161,4 +161,22 @@ void account_list_move_up( guint index );
*/
void account_list_move_down( guint index );
+/**
+ * Return the ID of the current default account
+ * @return gchar* The id
+ */
+gchar* account_list_get_current_id( void );
+
+/**
+ * Returns the number of SIP accounts that have been configured
+ */
+int account_list_get_sip_account_number( void );
+
+/**
+ * Returns the number of IAX accounts that have been configured
+ */
+int account_list_get_iax_account_number( void );
+
+
+
#endif
diff --git a/sflphone-gtk/src/actions.c b/sflphone-gtk/src/actions.c
index 39cbbf2ca37676d0809c21c06bd09524fac43b9e..6b1aaf7c3fcad92b30537b8902025cd593704ba7 100644
--- a/sflphone-gtk/src/actions.c
+++ b/sflphone-gtk/src/actions.c
@@ -39,25 +39,35 @@ guint voice_mails;
void
sflphone_notify_voice_mail ( const gchar* accountID , guint count )
{
- voice_mails = count ;
- gchar* id = g_strdup( accountID );
- if(count > 0)
- {
- gchar * message = g_new0(gchar, 50);
- if( count > 1)
- g_sprintf(message, _("%d voice mails"), count);
- else
- g_sprintf(message, _("%d voice mail"), count);
- statusbar_push_message(message, __MSG_VOICE_MAILS);
- g_free(message);
- }
- // TODO: add ifdef
- if( account_list_get_size() > 0 )
- {
- account_t* acc = account_list_get_by_id( id );
- if( acc != NULL )
- notify_voice_mails( count , acc );
- }
+ gchar *id;
+ gchar *current;
+
+ // We want to notify only for the default current account; ie the first in the list
+ id = g_strdup( accountID );
+ current = account_list_get_current_id();
+ if( strcmp( id, current ) != 0 )
+ return;
+
+ voice_mails = count ;
+
+ if(count > 0)
+ {
+ gchar * message = g_new0(gchar, 50);
+ if( count > 1)
+ g_sprintf(message, _("%d voice mails"), count);
+ else
+ g_sprintf(message, _("%d voice mail"), count);
+ statusbar_push_message(message, __MSG_VOICE_MAILS);
+ g_free(message);
+ }
+
+ // TODO: add ifdef
+ if( account_list_get_size() > 0 )
+ {
+ account_t* acc = account_list_get_by_id( id );
+ if( acc != NULL )
+ notify_voice_mails( count , acc );
+ }
}
void
diff --git a/sflphone-gtk/src/configwindow.c b/sflphone-gtk/src/configwindow.c
index 9081ef06e9a7434d5b4dfb899b5b8b80828aa89f..2c2c731bc37e8f1d3453b2cf7662b833e2cca495 100644
--- a/sflphone-gtk/src/configwindow.c
+++ b/sflphone-gtk/src/configwindow.c
@@ -314,7 +314,7 @@ set_pulse_app_volume_control( void )
dbus_set_pulse_app_volume_control();
}
-static void update_port( GtkSpinButton *button, void *ptr )
+static void update_port( GtkSpinButton *button UNUSED, void *ptr )
{
dbus_set_sip_port(gtk_spin_button_get_value_as_int((GtkSpinButton *)(ptr)));
}
@@ -439,6 +439,10 @@ create_accounts_tab()
GtkWidget*
create_general_settings ()
{
+
+ int curPort;
+ int n;
+
GtkWidget *ret;
GtkWidget *notifFrame;
@@ -542,14 +546,18 @@ create_general_settings ()
gtk_box_pack_start( GTK_BOX(vbox) , widg , TRUE , TRUE , 1);
g_signal_connect(G_OBJECT( widg ) , "clicked" , G_CALLBACK( set_pulse_app_volume_control ) , NULL);
+ n = account_list_get_sip_account_number();
+ printf("sip account number = %i\n", n);
+
/** SIP port information */
- int curPort = dbus_get_sip_port();
+ curPort = dbus_get_sip_port();
if(curPort <= 0 || curPort > 65535)
curPort = 5060;
-
+
frame = gtk_frame_new( _("SIP Port"));
gtk_box_pack_start(GTK_BOX(ret), frame, FALSE, FALSE, 0);
gtk_widget_show( frame );
+ gtk_widget_set_sensitive( GTK_WIDGET(frame), (n==0)?FALSE:TRUE );
hbox = gtk_hbox_new(FALSE, 10);
gtk_widget_show( hbox );
@@ -557,6 +565,7 @@ create_general_settings ()
GtkWidget *applyButton = gtk_button_new_with_label(_("Apply"));
gtk_widget_set_size_request(applyButton, 60, 35);
+ //gtk_widget_set_sensitive( GTK_WIDGET(applyButton), (n==0)?FALSE:TRUE );
label = gtk_label_new(_("Port:"));
diff --git a/src/audio/alsalayer.cpp b/src/audio/alsalayer.cpp
index e1ba97a377118d7bcb8b8b0b6bb4a9a05c07b3e7..72673ec566442a3cd146b7d19423019e90a8cffa 100644
--- a/src/audio/alsalayer.cpp
+++ b/src/audio/alsalayer.cpp
@@ -296,8 +296,8 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
snd_pcm_uframes_t period_size_out = getFrameSize() * getSampleRate() / 1000 * 2;//1024 ;
snd_pcm_uframes_t buffer_size_out = period_size_out * 4 ;
- unsigned int buffer_time = 80000; //80ms
- unsigned int period_time = buffer_time / 4 ; //20ms
+ unsigned int buffer_time = 100000; //100ms
+ unsigned int period_time = buffer_time / 4 ; //25ms
if(flag == SFL_PCM_BOTH || flag == SFL_PCM_CAPTURE)
{
diff --git a/src/audio/audiostream.cpp b/src/audio/audiostream.cpp
index 520443d15741cb267021788087fc17d9f2e99305..1f7df13ba887c2c66f334f3cd1039567850680a1 100644
--- a/src/audio/audiostream.cpp
+++ b/src/audio/audiostream.cpp
@@ -21,14 +21,14 @@
static pa_channel_map channel_map ;
-AudioStream::AudioStream( pa_context* context, int type, std::string desc, double vol )
+AudioStream::AudioStream( pa_context* context, int type, std::string desc, double vol UNUSED )
: _audiostream(NULL), _streamType(type), _streamDescription(desc), flag(PA_STREAM_AUTO_TIMING_UPDATE), sample_spec(), _volume()
{
sample_spec.format = PA_SAMPLE_S16LE;
sample_spec.rate = 44100;
sample_spec.channels = 1;
channel_map.channels = 1;
- pa_cvolume_set( &_volume , 1 , PA_VOLUME_MUTED ) ; // * vol / 100 ;
+ pa_cvolume_set( &_volume , 1 , PA_VOLUME_NORM ) ; // * vol / 100 ;
_audiostream = createStream( context );
}
@@ -49,7 +49,7 @@ AudioStream::disconnect( void )
}
void
-AudioStream::stream_state_callback( pa_stream* s, void* user_data )
+AudioStream::stream_state_callback( pa_stream* s, void* user_data UNUSED )
{
_debug("The state of the stream changed\n");
assert(s);
diff --git a/src/iaxcall.cpp b/src/iaxcall.cpp
index 04c5928152a01f2fd7315283169502299bd98cf1..05327d77d6a15719bf26ce6b394a08ba33a992a1 100644
--- a/src/iaxcall.cpp
+++ b/src/iaxcall.cpp
@@ -55,9 +55,11 @@ IAXCall::setFormat(int format)
int
IAXCall::getSupportedFormat()
{
- CodecOrder map = getCodecMap().getActiveCodecs();
- int format = 0;
- int iter;
+ CodecOrder map;
+ int format = 0;
+ unsigned int iter;
+
+ map = getCodecMap().getActiveCodecs();
for(iter=0 ; iter < map.size() ; iter++){
switch(map[iter]) {
@@ -85,7 +87,7 @@ IAXCall::getFirstMatchingFormat(int needles)
{
CodecOrder map = getCodecMap().getActiveCodecs();
int format = 0;
- int iter;
+ unsigned int iter;
for(iter=0 ; iter < map.size() ; iter++) {
switch(map[iter]) {
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index 4da2f14b5c11c01ec4739b24576a95c94d0f9ef2..f8bf510607a42a1085f57e2c6addc8b2014a85a5 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -216,16 +216,18 @@ IAXVoIPLink::getEvent()
void
IAXVoIPLink::sendAudioFromMic(void)
{
- IAXCall* currentCall = getIAXCall(Manager::instance().getCurrentCallId());
-
- if (!currentCall) {
- // Let's mind our own business.
- return;
- }
-
- if( currentCall -> getAudioCodec() < 0 )
- return;
+ int maxBytesToGet, availBytesFromMic, bytesAvail, nbSample, compSize;
+ AudioCodec *ac;
+
+ IAXCall* currentCall = getIAXCall(Manager::instance().getCurrentCallId());
+
+ if (!currentCall) {
+ // Let's mind our own business.
+ return;
+ }
+ if( currentCall -> getAudioCodec() < 0 )
+ return;
// Just make sure the currentCall is in state to receive audio right now.
//_debug("Here we get: connectionState: %d state: %d \n",
@@ -237,7 +239,7 @@ IAXVoIPLink::sendAudioFromMic(void)
return;
}
- AudioCodec* ac = currentCall -> getCodecMap().getCodec( currentCall -> getAudioCodec() );
+ ac = currentCall -> getCodecMap().getCodec( currentCall -> getAudioCodec() );
if (!ac) {
// Audio codec still not determined.
if (audiolayer) {
@@ -252,10 +254,15 @@ IAXVoIPLink::sendAudioFromMic(void)
// we have to get 20ms of data from the mic *20/1000 = /50
// rate/50 shall be lower than IAX__20S_48KHZ_MAX
- int maxBytesToGet = audiolayer->getSampleRate()* audiolayer->getFrameSize() / 1000 * sizeof(SFLDataFormat);
+ maxBytesToGet = audiolayer->getSampleRate()* audiolayer->getFrameSize() / 1000 * sizeof(SFLDataFormat);
+
+ // We have to update the audio layer type in case we switched
+ // TODO Find out a better way to do it
+ updateAudiolayer();
// available bytes inside ringbuffer
- int availBytesFromMic = audiolayer->canGetMic();
+ availBytesFromMic = audiolayer->canGetMic();
+
if (availBytesFromMic < maxBytesToGet) {
// We need packets full!
@@ -263,17 +270,17 @@ IAXVoIPLink::sendAudioFromMic(void)
}
// take the lowest
- int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
+ bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
//_debug("available = %d, maxBytesToGet = %d\n", availBytesFromMic, maxBytesToGet);
// Get bytes from micRingBuffer to data_from_mic
- int nbSample = audiolayer->getMic( micData, bytesAvail ) / sizeof(SFLDataFormat);
+ nbSample = audiolayer->getMic( micData, bytesAvail ) / sizeof(SFLDataFormat);
// resample
nbSample = converter->downsampleData( micData , micDataConverted , (int)ac ->getClockRate() , (int)audiolayer->getSampleRate() , nbSample );
// for the mono: range = 0 to IAX_FRAME2SEND * sizeof(int16)
- int compSize = ac->codecEncode( micDataEncoded, micDataConverted , nbSample*sizeof(int16));
+ compSize = ac->codecEncode( micDataEncoded, micDataConverted , nbSample*sizeof(int16));
// Send it out!
_mutexIAX.enterMutex();
@@ -482,6 +489,8 @@ IAXVoIPLink::transfer(const CallID& id, const std::string& to)
iax_transfer(call->getSession(), callto);
_mutexIAX.leaveMutex();
+ return true;
+
// should we remove it?
// removeCall(id);
}
@@ -498,6 +507,8 @@ IAXVoIPLink::refuse(const CallID& id)
iax_reject(call->getSession(), (char*) reason.c_str());
_mutexIAX.leaveMutex();
removeCall(id);
+
+ return true;
}
bool
@@ -510,6 +521,8 @@ IAXVoIPLink::carryingDTMFdigits(const CallID& id, char code)
_mutexIAX.enterMutex();
iax_send_dtmf(call->getSession(), code);
_mutexIAX.leaveMutex();
+
+ return true;
}
@@ -575,7 +588,6 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
// note activity?
//
CallID id = call->getCallId();
- int16* output = 0; // for audio output
switch(event->etype) {
case IAX_EVENT_HANGUP:
@@ -675,6 +687,13 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
void
IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
{
+
+
+ unsigned char *data;
+ unsigned int size, max, nbInt16;
+ int expandedSize, nbSample;
+ AudioCodec *ac;
+
// If we receive datalen == 0, some things of the jitter buffer in libiax2/iax.c
// were triggered
if (!event->datalen) {
@@ -691,28 +710,28 @@ IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
call->setFormat(event->subclass);
}
//_debug("Receive: len=%d, format=%d, _receiveDataDecoded=%p\n", event->datalen, call->getFormat(), _receiveDataDecoded);
- AudioCodec* ac = call->getCodecMap().getCodec( call -> getAudioCodec() );
+ ac = call->getCodecMap().getCodec( call -> getAudioCodec() );
- unsigned char* data = (unsigned char*)event->data;
- unsigned int size = event->datalen;
+ data = (unsigned char*)event->data;
+ size = event->datalen;
// Decode data with relevant codec
- int max = (int)( ac->getClockRate() * audiolayer->getFrameSize() / 1000 );
+ max = (int)( ac->getClockRate() * audiolayer->getFrameSize() / 1000 );
if (size > max) {
_debug("The size %d is bigger than expected %d. Packet cropped. Ouch!\n", size, max);
size = max;
}
- int expandedSize = ac->codecDecode( spkrDataDecoded , data , size );
- int nbInt16 = expandedSize/sizeof(int16);
+ expandedSize = ac->codecDecode( spkrDataDecoded , data , size );
+ nbInt16 = expandedSize/sizeof(int16);
if (nbInt16 > max) {
- _debug("We have decoded an IAX VOICE packet larger than expected: %s VS %s. Cropping.\n", nbInt16, max);
+ _debug("We have decoded an IAX VOICE packet larger than expected: %i VS %i. Cropping.\n", nbInt16, max);
nbInt16 = max;
}
- int nbSample = nbInt16;
+ nbSample = nbInt16;
// resample
nbInt16 = converter->upsampleData( spkrDataDecoded , spkrDataConverted , ac->getClockRate() , audiolayer->getSampleRate() , nbSample);
@@ -730,26 +749,40 @@ IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
void
IAXVoIPLink::iaxHandleRegReply(iax_event* event)
{
- if (event->etype == IAX_EVENT_REGREJ) {
- /* Authentication failed! */
- _mutexIAX.enterMutex();
- iax_destroy(_regSession);
- _mutexIAX.leaveMutex();
- _regSession = NULL;
- setRegistrationState(ErrorAuth);
- }
- else if (event->etype == IAX_EVENT_REGACK) {
- /* Authentication succeeded */
- _mutexIAX.enterMutex();
- iax_destroy(_regSession);
- _mutexIAX.leaveMutex();
- _regSession = NULL;
- // I mean, save the timestamp, so that we re-register again in the REFRESH time.
- // Defaults to 60, as per draft-guy-iax-03.
- _nextRefreshStamp = time(NULL) + (event->ies.refresh ? event->ies.refresh : 60);
- setRegistrationState(Registered);
- }
+ int voicemail;
+ std::string account_id;
+
+ if (event->etype == IAX_EVENT_REGREJ) {
+ /* Authentication failed! */
+ _mutexIAX.enterMutex();
+ iax_destroy(_regSession);
+ _mutexIAX.leaveMutex();
+ _regSession = NULL;
+ setRegistrationState(ErrorAuth);
+ }
+
+ else if (event->etype == IAX_EVENT_REGACK) {
+ /* Authentication succeeded */
+ _mutexIAX.enterMutex();
+
+ // Looking for the voicemail information
+ //if( event->ies != 0 )
+ voicemail = event->ies.msgcount;
+ _debug("iax voicemail number notification: %i\n", voicemail);
+ // Notify the client if new voicemail waiting for the current account
+ account_id = getAccountID();
+ Manager::instance().startVoiceMessageNotification(account_id.c_str(), voicemail);
+
+ iax_destroy(_regSession);
+ _mutexIAX.leaveMutex();
+ _regSession = NULL;
+
+ // I mean, save the timestamp, so that we re-register again in the REFRESH time.
+ // Defaults to 60, as per draft-guy-iax-03.
+ _nextRefreshStamp = time(NULL) + (event->ies.refresh ? event->ies.refresh : 60);
+ setRegistrationState(Registered);
+ }
}
void
@@ -858,3 +891,8 @@ IAXVoIPLink::iaxCodecMapToFormat(IAXCall* call)
}
+void IAXVoIPLink::updateAudiolayer( void )
+{
+ audiolayer = NULL;
+ audiolayer = Manager::instance().getAudioDriver();
+}
diff --git a/src/iaxvoiplink.h b/src/iaxvoiplink.h
index 9f1525afe28cffdb6c22a1947b7e81652d5e48f8..569ac88298964b0462cb4e4ef5945b3dd72cffdb 100644
--- a/src/iaxvoiplink.h
+++ b/src/iaxvoiplink.h
@@ -124,7 +124,7 @@ class IAXVoIPLink : public VoIPLink
* @return bool true on success
* false otherwise
*/
- bool cancel(const CallID& id) { return false; }
+ bool cancel(const CallID& id UNUSED ) { return false; }
/**
* Put a call on hold
@@ -168,7 +168,7 @@ class IAXVoIPLink : public VoIPLink
*/
bool carryingDTMFdigits(const CallID& id, char code);
- bool sendMessage(const std::string& to, const std::string& body) { return false; }
+ bool sendMessage(const std::string& to UNUSED, const std::string& body UNUSED) { return false; }
bool isContactPresenceSupported() { return false; }
@@ -188,6 +188,8 @@ class IAXVoIPLink : public VoIPLink
*/
void setPass(const std::string& pass) { _pass = pass; }
+ void updateAudiolayer( void );
+
private:
/**
* Get IAX Call from an id
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index c6107423007acc01fb6d76646421f9e8ab80b9e2..82f6bae0a4698d1db62a9488c7f8030bfbc7a7ac 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -1032,7 +1032,7 @@ ManagerImpl::createSettingsPath (void) {
* Initialization: Main Thread
*/
void
-ManagerImpl::initConfigFile (void)
+ManagerImpl::initConfigFile ( bool load_user_value )
{
std::string mes = gettext("Init config file\n");
_debug("%s",mes.c_str());
@@ -1079,7 +1079,7 @@ ManagerImpl::initConfigFile (void)
fill_config_int(CONFIG_SIP_PORT, DFT_SIP_PORT);
// Loads config from ~/.sflphone/sflphonedrc or so..
- if (createSettingsPath() == 1) {
+ if (createSettingsPath() == 1 && load_user_value) {
_exist = _config.populateFromFile(_path);
}
@@ -1161,6 +1161,7 @@ ManagerImpl::getActiveCodecList( void )
std::stringstream ss;
ss << active[i];
v.push_back((ss.str()).data());
+ _debug("%s\n", ss.str().data());
i++;
}
return v;
@@ -1576,8 +1577,11 @@ ManagerImpl::getCurrentAudioOutputPlugin( void )
void
ManagerImpl::initAudioDriver(void)
{
- _debugInit("AudioLayer Creation");
+ int error;
+
+ _debugInit("AudioLayer Creation");
+
if( getConfigInt( PREFERENCES , CONFIG_AUDIO ) == ALSA )
_audiodriver = new AlsaLayer( this );
else if( getConfigInt( PREFERENCES , CONFIG_AUDIO ) == PULSEAUDIO )
@@ -1588,7 +1592,7 @@ ManagerImpl::initAudioDriver(void)
if (_audiodriver == 0) {
_debug("Init audio driver error\n");
} else {
- int error = getAudioDriver()->getErrorMessage();
+ error = getAudioDriver()->getErrorMessage();
if (error == -1) {
_debug("Init audio driver: %i\n", error);
}
@@ -2401,7 +2405,13 @@ UserAgent *ManagerImpl::getUserAgent()
int
ManagerImpl::getSipPort()
{
- return _userAgent->getSipPort();
+ if( _userAgent )
+ return _userAgent->getSipPort();
+ else
+ {
+ // It means that no SIP accounts are configured, so return a default value
+ return 0;
+ }
}
void
diff --git a/src/managerimpl.h b/src/managerimpl.h
index bb16b7134ba0f72d5436f7efad8b375614a64e95..b97edb59d6de79e94137ccbc14f1658584427a95 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -784,7 +784,7 @@ class ManagerImpl {
* Fills the local _config (Conf::ConfigTree) with the default contents.
* Called in main.cpp, just before Manager::init().
*/
- void initConfigFile (void);
+ void initConfigFile ( bool load_user_value = true );
/**
* Tell if the setup was already loaded
@@ -1058,6 +1058,8 @@ private:
bool testAccountMap();
#endif
+ friend class ConfigurationTest;
+
public:
/**
* Retuun the instance of sip manager
diff --git a/test/Makefile.am b/test/Makefile.am
index aab435ef9c922e4e59e616d90cdf30b3d953b404..14411aa140bbc3264308b27f0cd418287544821c 100644
--- a/test/Makefile.am
+++ b/test/Makefile.am
@@ -20,6 +20,7 @@ OBJECT_FILES= \
configurationTester_SOURCES = \
configurationTest.cpp \
+ configurationTest.h \
TestMain.cpp
configurationTester_LDADD = \
diff --git a/test/configurationTest.cpp b/test/configurationTest.cpp
index 02225a54aaf540b9673c7e72f82205af493433b4..8208c9944fa6699baa92d5aabf63ff621af995c5 100644
--- a/test/configurationTest.cpp
+++ b/test/configurationTest.cpp
@@ -1,38 +1,137 @@
-#include <cppunit/extensions/HelperMacros.h>
-#include <cppunit/TestCaller.h>
-#include <cppunit/TestCase.h>
-#include <cppunit/TestSuite.h>
-#include <assert.h>
+/*
+ * Copyright (C) 2008 Savoir-Faire Linux inc.
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
-#include "manager.h"
-#include "global.h"
+#include <stdio.h>
+#include <sstream>
-// Cppunit import
+#include "configurationTest.h"
-class ConfigurationTest : public CppUnit::TestCase {
+using std::cout;
+using std::endl;
- CPPUNIT_TEST_SUITE( ConfigurationTest );
- CPPUNIT_TEST( testDefaultValueAudio );
- CPPUNIT_TEST( testTheTest );
- CPPUNIT_TEST_SUITE_END();
+void ConfigurationTest::setUp(){
+ // Load the default configuration
+ Manager::instance().initConfigFile();
+}
- public:
- ConfigurationTest() : CppUnit::TestCase("Configuration Tests") {}
-
- void testDefaultValueAudio(){
- CPPUNIT_ASSERT( Manager::instance().getConfigString( AUDIO, ALSA_PLUGIN ) == "default" );
- }
+void ConfigurationTest::testDefaultValueAudio(){
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( AUDIO, ALSA_CARD_ID_IN ) == ALSA_DFT_CARD) ;
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( AUDIO, ALSA_CARD_ID_OUT ) == ALSA_DFT_CARD );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( AUDIO, ALSA_SAMPLE_RATE ) == DFT_SAMPLE_RATE);
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( AUDIO, ALSA_FRAME_SIZE ) == DFT_FRAME_SIZE) ;
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( AUDIO, ALSA_PLUGIN ) == PCM_DEFAULT );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( AUDIO, VOLUME_SPKR ) == DFT_VOL_SPKR_STR);
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( AUDIO, VOLUME_MICRO ) == DFT_VOL_MICRO_STR);
+}
- void testTheTest(){
- CPPUNIT_ASSERT( 3 == 2 );
- }
+void ConfigurationTest::testDefaultValuePreferences(){
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( PREFERENCES, ZONE_TONE ) == DFT_ZONE );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( PREFERENCES, CONFIG_ZEROCONF ) == CONFIG_ZEROCONF_DEFAULT_STR );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( PREFERENCES, CONFIG_DIALPAD ) == YES_STR );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( PREFERENCES, CONFIG_RINGTONE ) == YES_STR );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( PREFERENCES, CONFIG_SEARCHBAR ) == YES_STR );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( PREFERENCES, CONFIG_START ) == NO_STR );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( PREFERENCES, CONFIG_POPUP ) == YES_STR );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( PREFERENCES, CONFIG_NOTIFY ) == YES_STR );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( PREFERENCES, CONFIG_MAIL_NOTIFY ) == NO_STR );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( PREFERENCES, CONFIG_VOLUME ) == YES_STR );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( PREFERENCES, REGISTRATION_EXPIRE ) == DFT_EXPIRE_VALUE );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( PREFERENCES, CONFIG_AUDIO ) == DFT_AUDIO_MANAGER );
- void setUp(){
- }
+}
- void tearDown(){
- }
+void ConfigurationTest::testDefaultValueSignalisation(){
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( SIGNALISATION , SYMMETRIC ) == YES_STR );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( SIGNALISATION , PLAY_DTMF ) == YES_STR );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( SIGNALISATION , PLAY_TONES ) == YES_STR );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( SIGNALISATION , PULSE_LENGTH ) == DFT_PULSE_LENGTH_STR );
+ CPPUNIT_ASSERT( Manager::instance().getConfigString( SIGNALISATION , SEND_DTMF_AS ) == SIP_INFO_STR );
+}
-};
+void ConfigurationTest::testLoadSIPAccount(){
-CPPUNIT_TEST_SUITE_REGISTRATION( ConfigurationTest );
+ AccountMap accounts;
+ Account *current;
+ std::ostringstream ss;
+ int nb_account; // Must be 1
+
+ // Load the account from the user file
+ nb_account = Manager::instance().loadAccountMap();
+ CPPUNIT_ASSERT_EQUAL( 1, nb_account );
+ // Save the account information
+ accounts = Manager::instance()._accountMap;
+
+ AccountMap::iterator iter = accounts.begin();
+ CPPUNIT_ASSERT( Manager::instance().accountExists( iter->first ) == true );
+
+ while( iter != accounts.end() ){
+ current = iter->second;
+ CPPUNIT_ASSERT( iter->first == current->getAccountID() );
+ CPPUNIT_ASSERT( 0 == current->getVoIPLink() );
+ iter++;
+ }
+}
+
+void ConfigurationTest::testUnloadSIPAccount(){
+
+ AccountMap accounts;
+
+ // Load the accounts from the user file
+ Manager::instance().loadAccountMap();
+ // Unload the accounts
+ Manager::instance().unloadAccountMap();
+ // Save the account information
+ accounts = Manager::instance()._accountMap;
+
+ AccountMap::iterator iter = accounts.begin();
+ CPPUNIT_ASSERT( Manager::instance().accountExists( iter->first ) == false );
+
+ if( iter != accounts.end() ){
+ CPPUNIT_FAIL("Unload account map failed\n");
+ }
+}
+
+void ConfigurationTest::testInitVolume(){
+
+ Manager::instance().initVolume();
+
+ CPPUNIT_ASSERT( Manager::instance().getConfigInt( AUDIO, VOLUME_SPKR) == Manager::instance().getSpkrVolume() );
+ CPPUNIT_ASSERT( Manager::instance().getConfigInt( AUDIO, VOLUME_MICRO) == Manager::instance().getMicVolume() );
+}
+
+void ConfigurationTest::testInitAudioDriver(){
+
+ // Load the audio driver
+ Manager::instance().initAudioDriver();
+
+ // Check the creation
+ if( Manager::instance().getAudioDriver() == NULL )
+ CPPUNIT_FAIL("Error while loading audio layer");
+
+ // Check if it has been created with the right type
+ if( Manager::instance().getConfigInt( PREFERENCES, CONFIG_AUDIO ) == ALSA )
+ CPPUNIT_ASSERT_EQUAL( Manager::instance().getAudioDriver()->getLayerType(), ALSA );
+ else if( Manager::instance().getConfigInt( PREFERENCES, CONFIG_AUDIO ) == PULSEAUDIO )
+ CPPUNIT_ASSERT_EQUAL( Manager::instance().getAudioDriver()->getLayerType(), PULSEAUDIO );
+ else
+ CPPUNIT_FAIL("Wrong audio layer type");
+}
+
+void ConfigurationTest::testSelectAudioDriver(){
+}
diff --git a/test/configurationTest.h b/test/configurationTest.h
new file mode 100644
index 0000000000000000000000000000000000000000..87828521eed303cbe2b195684f0ffbe45048db74
--- /dev/null
+++ b/test/configurationTest.h
@@ -0,0 +1,108 @@
+/*
+ * Copyright (C) 2008 Savoir-Faire Linux inc.
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+// Cppunit import
+#include <cppunit/extensions/HelperMacros.h>
+#include <cppunit/TestCaller.h>
+#include <cppunit/TestCase.h>
+#include <cppunit/TestSuite.h>
+
+#include <assert.h>
+
+// Application import
+#include "manager.h"
+#include "audio/audiolayer.h"
+#include "global.h"
+#include "user_cfg.h"
+
+/*
+ * @file configurationTest.cpp
+ * @brief Regroups unitary tests related to the user configuration.
+ * Check if the default configuration has been successfully loaded
+ */
+
+#ifndef _CONFIGURATION_TEST_
+#define _CONFIGURATION_TEST_
+
+class ConfigurationTest : public CppUnit::TestCase {
+
+ /*
+ * Use cppunit library macros to add unit test the factory
+ */
+ CPPUNIT_TEST_SUITE( ConfigurationTest );
+ CPPUNIT_TEST( testDefaultValueAudio );
+ CPPUNIT_TEST( testDefaultValuePreferences );
+ CPPUNIT_TEST( testDefaultValueSignalisation );
+ CPPUNIT_TEST( testLoadSIPAccount );
+ CPPUNIT_TEST( testUnloadSIPAccount );
+ CPPUNIT_TEST( testInitVolume );
+ CPPUNIT_TEST( testInitAudioDriver );
+ CPPUNIT_TEST( testSelectAudioDriver );
+ CPPUNIT_TEST_SUITE_END();
+
+ public:
+ ConfigurationTest() : CppUnit::TestCase("Configuration Tests") {}
+
+ /*
+ * Code factoring - Common resources can be initialized here.
+ * This method is called by unitcpp before each test
+ */
+ void setUp();
+
+ /*
+ * Code factoring - Common resources can be released here.
+ * This method is called by unitcpp after each test
+ */
+ inline void tearDown(){
+ // Not much to do
+ }
+
+ /*
+ * Unit tests related to the audio preferences
+ */
+ void testDefaultValueAudio();
+
+ /*
+ * Unit tests related to the global settings
+ */
+ void testDefaultValuePreferences();
+
+ /*
+ * Unit tests related to the global settings
+ */
+ void testDefaultValueSignalisation();
+
+ /*
+ * Try to load one SIP account.
+ * So be sure to have only one SIP account so that the test could succeed
+ */
+ void testLoadSIPAccount();
+ void testUnloadSIPAccount();
+
+ void testInitVolume();
+
+ void testInitAudioDriver();
+ void testSelectAudioDriver();
+
+};
+
+/* Register our test module */
+CPPUNIT_TEST_SUITE_REGISTRATION( ConfigurationTest );
+
+#endif