diff --git a/sflphone-gtk/src/actions.c b/sflphone-gtk/src/actions.c
index 161935df460a4759343761a1d3ede8832d9d49a3..6d6bbe50b07bdde12073bf4567ab5dd476d2f2f5 100644
--- a/sflphone-gtk/src/actions.c
+++ b/sflphone-gtk/src/actions.c
@@ -135,6 +135,8 @@ sflphone_hung_up( call_t * c)
#if GTK_CHECK_VERSION(2,10,0)
status_tray_icon_blink( FALSE );
#endif
+
+ statusbar_pop_message(__MSG_ACCOUNT_DEFAULT);
}
/** Internal to actions: Fill account list */
@@ -449,6 +451,10 @@ process_dialing(call_t * c, guint keyval, gchar * key)
dbus_start_tone( FALSE , 0 );
//dbus_play_dtmf( key );
}
+
+ g_print("process_dialing : keyval : %i \n",keyval);
+ g_print("process_dialing : key : %s \n",key);
+
switch (keyval)
{
case 65293: /* ENTER */
@@ -462,9 +468,10 @@ process_dialing(call_t * c, guint keyval, gchar * key)
{ /* Brackets mandatory because of local vars */
gchar * before = c->to;
if(strlen(c->to) >= 1){
+
c->to = g_strndup(c->to, strlen(c->to) -1);
g_free(before);
- g_print("TO: %s\n", c->to);
+ g_print("TO: backspace %s\n", c->to);
if(c->state == CALL_STATE_DIALING)
{
@@ -487,14 +494,16 @@ process_dialing(call_t * c, guint keyval, gchar * key)
case 65509: /* CAPS */
break;
default:
- if (keyval < 255 || (keyval >65453 && keyval < 65466))
+ // if (keyval < 255 || (keyval >65453 && keyval < 65466))
+ if (keyval < 127)
{
- if(c->state != CALL_STATE_TRANSFERT)
+
+ if(c->state != CALL_STATE_TRANSFERT)
dbus_play_dtmf( key );
gchar * before = c->to;
c->to = g_strconcat(c->to, key, NULL);
g_free(before);
- g_print("TO: %s\n", c->to);
+ g_print("TO:default %s\n", c->to);
if(c->state == CALL_STATE_DIALING)
{
@@ -512,6 +521,9 @@ process_dialing(call_t * c, guint keyval, gchar * key)
call_t *
sflphone_new_call()
{
+
+ sflphone_on_hold();
+
// Play a tone when creating a new call
if( call_list_get_size(current_calls) == 0 )
dbus_start_tone( TRUE , ( voice_mails > 0 )? TONE_WITH_MESSAGE : TONE_WITHOUT_MESSAGE) ;
@@ -539,7 +551,7 @@ sflphone_new_call()
void
sflphone_keypad( guint keyval, gchar * key)
{
-
+
call_t * c = call_get_selected(current_calls);
if(c)
{
@@ -548,6 +560,7 @@ sflphone_keypad( guint keyval, gchar * key)
case CALL_STATE_DIALING: // Currently dialing => edit number
process_dialing(c, keyval, key);
break;
+ case CALL_STATE_RECORD:
case CALL_STATE_CURRENT:
switch (keyval)
{
@@ -681,7 +694,17 @@ sflphone_place_call ( call_t * c )
else
{
- account_t * current = account_list_get_current();
+ account_t * current;
+
+ if(c->accountID != 0)
+ current = account_list_get_by_id(c->accountID);
+ else
+ current = account_list_get_current();
+
+ // printf("sflphone_place_call :: c->accountID : %i \n",c->accountID);
+
+ // account_t * current = c->accountID;
+
if( current )
{
if(g_strcasecmp(g_hash_table_lookup( current->properties, "Status"),"REGISTERED")==0)
@@ -724,6 +747,31 @@ sflphone_place_call ( call_t * c )
}
+void
+sflphone_display_selected_codec (const gchar* codecName)
+{
+ call_t * selectedCall = call_get_selected(current_calls);
+ gchar* msg;
+ account_t* acc;
+ if(selectedCall->accountID != NULL){
+ acc = account_list_get_by_id(selectedCall->accountID);
+ msg = g_markup_printf_escaped(_("%s account- %s %s") ,
+ (gchar*)g_hash_table_lookup( acc->properties , ACCOUNT_TYPE),
+ (gchar*)g_hash_table_lookup( acc->properties , ACCOUNT_ALIAS),
+ codecName);
+ statusbar_push_message( msg , __MSG_ACCOUNT_DEFAULT);
+ g_free(msg);
+ }
+
+}
+
+gchar*
+sflphone_get_current_codec_name()
+{
+ call_t * selectedCall = call_get_selected(current_calls);
+ return dbus_get_current_codec_name(selectedCall);
+}
+
void
sflphone_rec_call()
{
@@ -745,6 +793,9 @@ sflphone_rec_call()
}
update_call_tree(current_calls,selectedCall);
update_menus();
+
+ // gchar* codname = sflphone_get_current_codec_name();
+ // printf("sflphone_get_current_codec_name: %s \n",codname);
}
/* Internal to action - set the __CURRENT_ACCOUNT variable */
@@ -807,3 +858,4 @@ sflphone_fill_codec_list()
exit(0);
}
}
+
diff --git a/sflphone-gtk/src/actions.h b/sflphone-gtk/src/actions.h
index ebe0157fe885f0ba6293c45bf42cdf1d44ebfdc4..7c22eeb1b9cc7d9a57fb930ff14580bbc8294cf1 100644
--- a/sflphone-gtk/src/actions.h
+++ b/sflphone-gtk/src/actions.h
@@ -168,4 +168,8 @@ void sflphone_fill_codec_list();
void sflphone_record (call_t *c);
void sflphone_rec_call (void);
+
+gchar* sflphone_get_current_codec_name();
+
+void sflphone_display_selected_codec (const gchar* codecName);
#endif
diff --git a/sflphone-gtk/src/callmanager-glue.h b/sflphone-gtk/src/callmanager-glue.h
index c4123cd1e39732f6931696431c5de88d78df6c31..1921fbb0e0b630413699dc5b7c71d2e308b4ba5c 100644
--- a/sflphone-gtk/src/callmanager-glue.h
+++ b/sflphone-gtk/src/callmanager-glue.h
@@ -529,6 +529,44 @@ org_sflphone_SFLphone_CallManager_get_current_call_id_async (DBusGProxy *proxy,
stuff->userdata = userdata;
return dbus_g_proxy_begin_call (proxy, "getCurrentCallID", org_sflphone_SFLphone_CallManager_get_current_call_id_async_callback, stuff, g_free, G_TYPE_INVALID);
}
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+gboolean
+org_sflphone_SFLphone_CallManager_get_current_codec_name (DBusGProxy *proxy, const char * IN_callID, char ** OUT_codecName, GError **error)
+
+{
+ return dbus_g_proxy_call (proxy, "getCurrentCodecName", error, G_TYPE_STRING, IN_callID, G_TYPE_INVALID, G_TYPE_STRING, OUT_codecName, G_TYPE_INVALID);
+}
+
+typedef void (*org_sflphone_SFLphone_CallManager_get_current_codec_name_reply) (DBusGProxy *proxy, char * OUT_codecName, GError *error, gpointer userdata);
+
+static void
+org_sflphone_SFLphone_CallManager_get_current_codec_name_async_callback (DBusGProxy *proxy, DBusGProxyCall *call, void *user_data)
+{
+ DBusGAsyncData *data = (DBusGAsyncData*) user_data;
+ GError *error = NULL;
+ char * OUT_codecName;
+ dbus_g_proxy_end_call (proxy, call, &error, G_TYPE_STRING, &OUT_codecName, G_TYPE_INVALID);
+ (*(org_sflphone_SFLphone_CallManager_get_current_codec_name_reply)data->cb) (proxy, OUT_codecName, error, data->userdata);
+ return;
+}
+
+static
+#ifdef G_HAVE_INLINE
+inline
+#endif
+DBusGProxyCall*
+org_sflphone_SFLphone_CallManager_get_current_codec_name_async (DBusGProxy *proxy, const char * IN_callID, org_sflphone_SFLphone_CallManager_get_current_codec_name_reply callback, gpointer userdata)
+
+{
+ DBusGAsyncData *stuff;
+ stuff = g_new (DBusGAsyncData, 1);
+ stuff->cb = G_CALLBACK (callback);
+ stuff->userdata = userdata;
+ return dbus_g_proxy_begin_call (proxy, "getCurrentCodecName", org_sflphone_SFLphone_CallManager_get_current_codec_name_async_callback, stuff, g_free, G_TYPE_STRING, IN_callID, G_TYPE_INVALID);
+}
#endif /* defined DBUS_GLIB_CLIENT_WRAPPERS_org_sflphone_SFLphone_CallManager */
G_END_DECLS
diff --git a/sflphone-gtk/src/calltree.c b/sflphone-gtk/src/calltree.c
index 9b56a004f7abb92acc019f3d55b6b783adbe1809..dd205d3f099a39f380bcec71775097e706393aa0 100644
--- a/sflphone-gtk/src/calltree.c
+++ b/sflphone-gtk/src/calltree.c
@@ -37,7 +37,7 @@ GtkToolItem * hangupButton;
GtkToolItem * holdButton;
GtkToolItem * transfertButton;
GtkToolItem * unholdButton;
-GtkToolItem * historyButton;
+// GtkToolItem * historyButton;
GtkToolItem * mailboxButton;
GtkToolItem * recButton;
guint transfertButtonConnId; //The button toggled signal connection ID
@@ -395,6 +395,7 @@ void row_activated(GtkTreeView *tree_view UNUSED,
break;
}
}
+ // if history
else
{
newCall = g_new0( call_t, 1 );
@@ -405,7 +406,8 @@ void row_activated(GtkTreeView *tree_view UNUSED,
g_sprintf(newCall->callID, "%d", rand());
newCall->_start = 0;
newCall->_stop = 0;
-
+ printf("call : account : %s \n", selectedCall->accountID);
+ newCall->accountID = selectedCall->accountID;
printf("call : from : %s to %s\n", newCall->from, newCall->to);
call_list_add(current_calls, newCall);
update_call_tree_add(current_calls, newCall);
@@ -689,6 +691,7 @@ update_call_tree (calltab_t* tab, call_t * c)
}
else
{
+ g_print("Stuff to be printed %s %s \n",call_get_number(c),call_get_name(c));
description = g_markup_printf_escaped("<b>%s</b> <i>%s</i>",
call_get_number(c),
call_get_name(c));
diff --git a/sflphone-gtk/src/dbus.c b/sflphone-gtk/src/dbus.c
index e56170f64f09f6f8824060f4ddacb015b7bcec1b..b58b4d667374ec1ca284dda6f251aaa1c64b7ab6 100644
--- a/sflphone-gtk/src/dbus.c
+++ b/sflphone-gtk/src/dbus.c
@@ -60,6 +60,15 @@ incoming_call_cb (DBusGProxy *proxy UNUSED,
sflphone_incoming_call (c);
}
+static void
+curent_selected_codec (DBusGProxy *proxy UNUSED,
+ const gchar* callID,
+ const gchar* codecName,
+ void * foo UNUSED )
+{
+ g_print ("Codec decided! %s\n",codecName);
+ sflphone_display_selected_codec (codecName);
+}
static void
volume_changed_cb (DBusGProxy *proxy UNUSED,
@@ -232,6 +241,14 @@ dbus_connect ()
dbus_g_proxy_connect_signal (callManagerProxy,
"incomingCall", G_CALLBACK(incoming_call_cb), NULL, NULL);
+ /* Current codec */
+ dbus_g_object_register_marshaller(g_cclosure_user_marshal_VOID__STRING_STRING_STRING,
+ G_TYPE_NONE, G_TYPE_STRING, G_TYPE_STRING, G_TYPE_INVALID);
+ dbus_g_proxy_add_signal (callManagerProxy,
+ "currentSelectedCodec", G_TYPE_STRING, G_TYPE_STRING, G_TYPE_INVALID);
+ dbus_g_proxy_connect_signal (callManagerProxy,
+ "currentSelectedCodec", G_CALLBACK(curent_selected_codec), NULL, NULL);
+
/* Register a marshaller for STRING,STRING */
dbus_g_object_register_marshaller(g_cclosure_user_marshal_VOID__STRING_STRING,
G_TYPE_NONE, G_TYPE_STRING, G_TYPE_STRING, G_TYPE_INVALID);
@@ -649,6 +666,30 @@ dbus_codec_details( int payload )
return array;
}
+gchar*
+dbus_get_current_codec_name(const call_t * c)
+{
+
+ printf("dbus_get_current_codec_name : CallID : %s \n", c->callID);
+
+ gchar* codecName;
+ GError* error = NULL;
+
+ org_sflphone_SFLphone_CallManager_get_current_codec_name (
+ callManagerProxy,
+ c->callID,
+ &codecName,
+ &error);
+ if(error)
+ {
+ g_error_free(error);
+ }
+
+ printf("dbus_get_current_codec_name : codecName : %s \n", codecName);
+
+ return codecName;
+}
+
gchar**
diff --git a/sflphone-gtk/src/dbus.h b/sflphone-gtk/src/dbus.h
index b6c1ebddfd468a6baba24276a02d9aa19ce2d198..f347a4a47e4395900d8b4fb4edb9417e9b82a421 100644
--- a/sflphone-gtk/src/dbus.h
+++ b/sflphone-gtk/src/dbus.h
@@ -177,6 +177,12 @@ gchar** dbus_get_active_codec_list( void );
*/
void dbus_set_active_codec_list( const gchar** list );
+/**
+ * CallManager - return the codec name
+ * @param call_t* current call
+ */
+gchar* dbus_get_current_codec_name(const call_t * c);
+
/**
* ConfigurationManager - Get the list of available input audio plugins
* @return gchar** The list of plugins
diff --git a/sflphone-gtk/src/mainwindow.c b/sflphone-gtk/src/mainwindow.c
index 9eb1a34fcc1d2b93831e84f630c6579fa0ffe5f3..6eaa0c0c723bed27ae928c77d4360422a6da2017 100644
--- a/sflphone-gtk/src/mainwindow.c
+++ b/sflphone-gtk/src/mainwindow.c
@@ -104,7 +104,7 @@ create_main_window ()
window = gtk_window_new (GTK_WINDOW_TOPLEVEL);
gtk_container_set_border_width (GTK_CONTAINER (window), 0);
gtk_window_set_title (GTK_WINDOW (window), PACKAGE);
- gtk_window_set_default_size (GTK_WINDOW (window), 260, 320);
+ gtk_window_set_default_size (GTK_WINDOW (window), 258, 320);
gtk_window_set_default_icon_from_file (ICONS_DIR "/sflphone.png",
NULL);
gtk_window_set_position( GTK_WINDOW( window ) , GTK_WIN_POS_MOUSE);
diff --git a/sflphone-gtk/src/menus.c b/sflphone-gtk/src/menus.c
index fcd371ac0a430e6dada126b7339c3586be4eede3..c0a21e40f1238f66570060ba8f3a51c9d43c714a 100644
--- a/sflphone-gtk/src/menus.c
+++ b/sflphone-gtk/src/menus.c
@@ -38,6 +38,7 @@ GtkWidget * newCallMenu;
GtkWidget * holdMenu;
GtkWidget * copyMenu;
GtkWidget * pasteMenu;
+GtkWidget * recordMenu;
guint holdConnId; //The hold_menu signal connection ID
@@ -55,6 +56,7 @@ void update_menus()
gtk_widget_set_sensitive( GTK_WIDGET(hangUpMenu), FALSE);
gtk_widget_set_sensitive( GTK_WIDGET(newCallMenu),FALSE);
gtk_widget_set_sensitive( GTK_WIDGET(holdMenu), FALSE);
+ gtk_widget_set_sensitive( GTK_WIDGET(recordMenu), FALSE);
gtk_widget_set_sensitive( GTK_WIDGET(copyMenu), FALSE);
call_t * selectedCall = call_get_selected(active_calltree);
@@ -83,20 +85,17 @@ void update_menus()
gtk_widget_set_sensitive( GTK_WIDGET(newCallMenu),TRUE);
break;
case CALL_STATE_CURRENT:
+ case CALL_STATE_RECORD:
gtk_widget_set_sensitive( GTK_WIDGET(hangUpMenu), TRUE);
gtk_widget_set_sensitive( GTK_WIDGET(holdMenu), TRUE);
gtk_widget_set_sensitive( GTK_WIDGET(newCallMenu),TRUE);
+ gtk_widget_set_sensitive( GTK_WIDGET(recordMenu), TRUE);
gtk_image_menu_item_set_image( GTK_IMAGE_MENU_ITEM ( holdMenu ), gtk_image_new_from_file( ICONS_DIR "/icon_hold.svg"));
break;
case CALL_STATE_BUSY:
case CALL_STATE_FAILURE:
gtk_widget_set_sensitive( GTK_WIDGET(hangUpMenu), TRUE);
break;
- case CALL_STATE_RECORD:
- gtk_widget_set_sensitive( GTK_WIDGET(hangUpMenu), TRUE);
- gtk_widget_set_sensitive( GTK_WIDGET(holdMenu), TRUE);
- gtk_widget_set_sensitive( GTK_WIDGET(newCallMenu),TRUE);
- break;
default:
g_warning("Should not happen in update_menus()!");
break;
@@ -225,6 +224,12 @@ call_hang_up ( void * foo UNUSED)
sflphone_hang_up();
}
+static void
+call_record ( void * foo UNUSED)
+{
+ sflphone_rec_call();
+}
+
static void
call_wizard ( void * foo UNUSED)
{
@@ -318,6 +323,16 @@ create_call_menu()
NULL);
gtk_widget_show (menu_items);
+ image = gtk_tool_button_new_from_stock (GTK_STOCK_MEDIA_RECORD);
+ recordMenu = gtk_image_menu_item_new_with_mnemonic(_("_Record"));
+ gtk_image_menu_item_set_image( GTK_IMAGE_MENU_ITEM ( recordMenu ), image );
+ gtk_menu_shell_append (GTK_MENU_SHELL (menu), recordMenu);
+ gtk_widget_set_sensitive( GTK_WIDGET(recordMenu), FALSE);
+ g_signal_connect_swapped (G_OBJECT (recordMenu), "activate",
+ G_CALLBACK (call_record),
+ NULL);
+ gtk_widget_show (recordMenu);
+
// Separator
menu_items = gtk_separator_menu_item_new ();
gtk_menu_shell_append (GTK_MENU_SHELL (menu), menu_items);
@@ -682,7 +697,7 @@ show_popup_menu (GtkWidget *my_widget, GdkEventButton *event)
{
// TODO update the selection to make sure the call under the mouse is the call selected
- gboolean pickup = FALSE, hangup = FALSE, hold = FALSE, copy = FALSE;
+ gboolean pickup = FALSE, hangup = FALSE, hold = FALSE, copy = FALSE, record = FALSE;
gboolean accounts = FALSE;
call_t * selectedCall = call_get_selected(current_calls);
@@ -707,9 +722,11 @@ show_popup_menu (GtkWidget *my_widget, GdkEventButton *event)
hangup = TRUE;
accounts = TRUE;
break;
+ case CALL_STATE_RECORD:
case CALL_STATE_CURRENT:
hangup = TRUE;
hold = TRUE;
+ record = TRUE;
break;
case CALL_STATE_BUSY:
case CALL_STATE_FAILURE:
@@ -791,6 +808,18 @@ show_popup_menu (GtkWidget *my_widget, GdkEventButton *event)
gtk_widget_show (menu_items);
}
+ if(record)
+ {
+ menu_items = gtk_image_menu_item_new_with_mnemonic(_("_Record"));
+ image = gtk_tool_button_new_from_stock (GTK_STOCK_MEDIA_RECORD);
+ gtk_image_menu_item_set_image(GTK_IMAGE_MENU_ITEM(menu_items), image);
+ gtk_menu_shell_append (GTK_MENU_SHELL (menu), menu_items);
+ g_signal_connect (G_OBJECT (menu_items), "activate",
+ G_CALLBACK (call_record),
+ NULL);
+ gtk_widget_show (menu_items);
+ }
+
if(accounts)
{
menu_items = gtk_separator_menu_item_new ();
diff --git a/sflphone-gtk/src/sflnotify.c b/sflphone-gtk/src/sflnotify.c
index a71ba9ea0e17b49a45934995aad9972ff4fa095a..b00123dfda655cc4da441d78b46ce37bdb59695e 100644
--- a/sflphone-gtk/src/sflnotify.c
+++ b/sflphone-gtk/src/sflnotify.c
@@ -205,8 +205,10 @@ notify_no_registered_accounts( )
gchar* body="";
notify_init("sflphone");
+
body = g_markup_printf_escaped(_("You have no registered accounts"));
+
title = g_markup_printf_escaped(_("Error"));
pixbuf = gdk_pixbuf_new_from_file(ICONS_DIR "/sflphone.png", NULL);
@@ -223,9 +225,10 @@ notify_no_registered_accounts( )
notify_notification_set_timeout( notification , NOTIFY_EXPIRES_DEFAULT );
notify_notification_add_action( notification , "setup" , _("Setup Accounts") , (NotifyActionCallback) setup_accounts_cb , NULL , NULL );
- if (!notify_notification_show (notification, NULL)) {
- g_print("notify(), failed to send notification\n");
- }
+ if (!notify_notification_show (notification, NULL)) {
+ g_print("notify(), failed to send notification\n");
+ }
+
}
void
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 319ba0d92626666004fe4da8aa5509e599129746..4a35ebca25329ed9d07b9bd6ffa8e9b42db9a34f 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -116,6 +116,7 @@ AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym) : time(new ost::Time()), _
_sym(sym), micData(NULL), micDataConverted(NULL), micDataEncoded(NULL), spkrDataDecoded(NULL), spkrDataConverted(NULL),
converter(NULL), _layerSampleRate(),_codecSampleRate(), _layerFrameSize(), _audiocodec(NULL)
{
+
setCancel(cancelDefault);
// AudioRtpRTX should be close if we change sample rate
// TODO: Change bind address according to user settings.
@@ -150,6 +151,7 @@ AudioRtpRTX::~AudioRtpRTX () {
delete _session; _session = NULL;
}
+
delete [] micData; micData = NULL;
delete [] micDataConverted; micDataConverted = NULL;
delete [] micDataEncoded; micDataEncoded = NULL;
@@ -163,6 +165,8 @@ AudioRtpRTX::~AudioRtpRTX () {
}
+
+
void
AudioRtpRTX::initBuffers()
{
@@ -248,6 +252,7 @@ AudioRtpRTX::initAudioRtpSession (void)
_debugException("! ARTP Failure: initialisation failed");
throw;
}
+
}
void
@@ -312,14 +317,18 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
{
+
if (_ca == 0) { return; }
//try {
+
AudioLayer* audiolayer = Manager::instance().getAudioDriver();
if (!audiolayer) { return; }
const ost::AppDataUnit* adu = NULL;
// Get audio data stream
+ // printf("AudioRtpRTX::receiveSessionForSpkr() %i \n",_session->getFirstTimestamp());
+
if (!_sym) {
adu = _sessionRecv->getData(_sessionRecv->getFirstTimestamp());
} else {
@@ -330,10 +339,14 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
return;
}
+
+
//int payload = adu->getType(); // codec type
unsigned char* spkrData = (unsigned char*)adu->getData(); // data in char
unsigned int size = adu->getSize(); // size in char
+ // printf("AudioRtpRTX::receiveSessionForSpkr() Size of data from %i \n",size);
+
// Decode data with relevant codec
unsigned int max = (unsigned int)(_codecSampleRate * _layerFrameSize / 1000);
@@ -359,6 +372,7 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
// Do sample rate conversion
int nb_sample_down = nbSample;
nbSample = reSampleData(_codecSampleRate , nb_sample_down, UP_SAMPLING);
+
#ifdef DATAFORMAT_IS_FLOAT
#else
#endif
@@ -403,17 +417,23 @@ AudioRtpRTX::reSampleData(int sampleRate_codec, int nbSamples, int status)
return 0;
}
+
+
void
AudioRtpRTX::run () {
- //mic, we receive from soundcard in stereo, and we send encoded
- //encoding before sending
- AudioLayer *audiolayer = Manager::instance().getAudioDriver();
- _layerFrameSize = audiolayer->getFrameSize(); // en ms
- _layerSampleRate = audiolayer->getSampleRate();
- initBuffers();
- int step;
- //try {
+ //mic, we receive from soundcard in stereo, and we send encoded
+ //encoding before sending
+ AudioLayer *audiolayer = Manager::instance().getAudioDriver();
+ _layerFrameSize = audiolayer->getFrameSize(); // en ms
+ _layerSampleRate = audiolayer->getSampleRate();
+ initBuffers();
+ int step;
+
+ int sessionWaiting;
+
+ //try {
+
// Init the session
initAudioRtpSession();
step = (int) (_layerFrameSize * _codecSampleRate / 1000);
@@ -433,27 +453,58 @@ AudioRtpRTX::run () {
audiolayer->startStream();
_start.post();
- _debug("- ARTP Action: Start\n");
+ _debug("- ARTP Action: Start call %s\n",_ca->getCallId().c_str());
while (!testCancel()) {
- ////////////////////////////
- // Send session
- ////////////////////////////
- sendSessionFromMic(timestamp);
- timestamp += step;
- ////////////////////////////
- // Recv session
- ////////////////////////////
- receiveSessionForSpkr(countTime);
- // Let's wait for the next transmit cycle
+
+ // printf("AudioRtpRTX::run() _session->getFirstTimestamp() %i \n",_session->getFirstTimestamp());
+
+ // printf("AudioRtpRTX::run() _session->isWaiting() %i \n",_session->isWaiting());
+ /////////////////////
+ ////////////////////////////
+ // Send session
+ ////////////////////////////
+
+ sessionWaiting = _session->isWaiting();
+
+ sendSessionFromMic(timestamp);
+ timestamp += step;
+
+ ////////////////////////////
+ // Recv session
+ ////////////////////////////
+ receiveSessionForSpkr(countTime);
+
+ // Let's wait for the next transmit cycle
+
+
+ if(sessionWaiting == 1){
+ // _debug("Record TWO buffer \n");
_ca->recAudio.recData(spkrDataConverted,micData,_nSamplesSpkr,_nSamplesMic);
-
- Thread::sleep(TimerPort::getTimer());
- TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
+ }
+ else {
+ // _debug("Record ONE buffer \n");
+ _ca->recAudio.recData(micData,_nSamplesMic);
+ }
+
+ Thread::sleep(TimerPort::getTimer());
+ TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
+
}
-
+
+ // _debug("stop stream for audiortp loop\n");
audiolayer->stopStream();
- _debug("- ARTP Action: Stop\n");
+ _debug("- ARTP Action: Stop call %s\n",_ca->getCallId().c_str());
+ //} catch(std::exception &e) {
+ //_start.post();
+ //_debug("! ARTP: Stop %s\n", e.what());
+ //throw;
+ //} catch(...) {
+ //_start.post();
+ //_debugException("* ARTP Action: Stop");
+ //throw;
+ //}
+
}
diff --git a/src/audio/audiortp.h b/src/audio/audiortp.h
index fe75441c0ee6ad36a84f77a103707e5daf2c8064..a0a3cf6846ec858ed6ec342b80b5a624d503e842 100644
--- a/src/audio/audiortp.h
+++ b/src/audio/audiortp.h
@@ -69,7 +69,7 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
/**
* Audio recording object
*/
- AudioRecord recAudio;
+ // AudioRecord recAudio;
/** A SIP call */
SIPCall* _ca;
@@ -128,7 +128,10 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
*/
int _nSamplesMic;
-
+ /**
+ * Maximum number of sample for audio buffers (mic and spkr)
+ */
+ int nbSamplesMax;
/**
* Init the RTP session. Create either symmetric or double sessions to manage data transport
@@ -167,10 +170,6 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
/** The audio codec used during the session */
AudioCodec* _audiocodec;
- /**
- * Audio recording object
- */
- // AudioRecord recAudio;
};
///////////////////////////////////////////////////////////////////////////////
@@ -203,7 +202,7 @@ class AudioRtp {
* Start recording
*/
void setRecording ();
-
+
private:
// copy constructor
diff --git a/src/call.cpp b/src/call.cpp
index e3a4f43cce078937e76271b8316205e8de4d4f20..32cc4d7bcda73c5da7f5fb5f38d4eac563d92f33 100644
--- a/src/call.cpp
+++ b/src/call.cpp
@@ -42,16 +42,16 @@ Call::Call(const CallID& id, Call::CallType type)
SOUND_FORMAT soundFormat = INT16;
recAudio.setRecordingOption(fileType,soundFormat,44100, Manager::instance().getConfigString (AUDIO, RECORD_PATH),id);
- _debug("CALL::Constructor for this clss is called \n");
+ // _debug("CALL::Constructor for this clss is called \n");
}
Call::~Call()
{
- _debug("CALL::~Call(): Destructor for this clss is called \n");
+ // _debug("CALL::~Call(): Destructor for this clss is called \n");
if(recAudio.isOpenFile()) {
- _debug("CALL::~Call(): A recording file is open, close it \n");
+ // _debug("CALL::~Call(): A recording file is open, close it \n");
recAudio.closeFile();
}
}
@@ -122,7 +122,6 @@ Call::getRemoteIp()
AudioCodecType
Call::getAudioCodec()
{
- ost::MutexLock m(_callMutex);
return _audioCodec;
}
@@ -146,6 +145,11 @@ Call::setRecording()
recAudio.setRecording();
}
+void
+Call::initRecFileName() {
+ recAudio.initFileName(_peerNumber);
+}
+
void
Call::stopRecording()
{
diff --git a/src/call.h b/src/call.h
index f0ad6b02eff1439155d710f9a99b709a5c997348..6b247e036017d684056f9db64a59d8a1460d35ae 100644
--- a/src/call.h
+++ b/src/call.h
@@ -237,6 +237,11 @@ class Call{
*/
bool isRecording();
+ /**
+ *
+ */
+ void initRecFileName();
+
protected:
/** Protect every attribute that can be changed by two threads */
ost::Mutex _callMutex;
diff --git a/src/dbus/callmanager-glue.h b/src/dbus/callmanager-glue.h
index 4abf94d05cbb97ae8bb735547f9d401f251307aa..71dd6b5170eca847ee33cbc5d457c7593c2c9f44 100644
--- a/src/dbus/callmanager-glue.h
+++ b/src/dbus/callmanager-glue.h
@@ -34,6 +34,7 @@ public:
register_method(CallManager_adaptor, setRecording, _setRecording_stub);
register_method(CallManager_adaptor, getCallDetails, _getCallDetails_stub);
register_method(CallManager_adaptor, getCurrentCallID, _getCurrentCallID_stub);
+ register_method(CallManager_adaptor, getCurrentCodecName, _getCurrentCodecName_stub);
}
::DBus::IntrospectedInterface *const introspect() const
@@ -115,6 +116,18 @@ public:
{ "callID", "s", false },
{ 0, 0, 0 }
};
+ static ::DBus::IntrospectedArgument getCurrentCodecName_args[] =
+ {
+ { "callID", "s", true },
+ { "codecName", "s", false },
+ { 0, 0, 0 }
+ };
+ static ::DBus::IntrospectedArgument currentSelectedCodec_args[] =
+ {
+ { "callID", "s", false },
+ { "codecName", "s", false },
+ { 0, 0, 0 }
+ };
static ::DBus::IntrospectedArgument incomingCall_args[] =
{
{ "accountID", "s", false },
@@ -167,10 +180,12 @@ public:
{ "setRecording", setRecording_args },
{ "getCallDetails", getCallDetails_args },
{ "getCurrentCallID", getCurrentCallID_args },
+ { "getCurrentCodecName", getCurrentCodecName_args },
{ 0, 0 }
};
static ::DBus::IntrospectedMethod CallManager_adaptor_signals[] =
{
+ { "currentSelectedCodec", currentSelectedCodec_args },
{ "incomingCall", incomingCall_args },
{ "incomingMessage", incomingMessage_args },
{ "callStateChanged", callStateChanged_args },
@@ -218,11 +233,20 @@ public:
virtual void setRecording(const std::string& callID) = 0;
virtual std::map< std::string, std::string > getCallDetails(const std::string& callID) = 0;
virtual std::string getCurrentCallID() = 0;
+ virtual std::string getCurrentCodecName(const std::string& callID) = 0;
public:
/* signal emitters for this interface
*/
+ void currentSelectedCodec(const std::string& arg1, const std::string& arg2)
+ {
+ ::DBus::SignalMessage sig("currentSelectedCodec");
+ ::DBus::MessageIter wi = sig.writer();
+ wi << arg1;
+ wi << arg2;
+ emit_signal(sig);
+ }
void incomingCall(const std::string& arg1, const std::string& arg2, const std::string& arg3)
{
::DBus::SignalMessage sig("incomingCall");
@@ -412,6 +436,17 @@ private:
wi << argout1;
return reply;
}
+ ::DBus::Message _getCurrentCodecName_stub(const ::DBus::CallMessage &call)
+ {
+ ::DBus::MessageIter ri = call.reader();
+
+ std::string argin1; ri >> argin1;
+ std::string argout1 = getCurrentCodecName(argin1);
+ ::DBus::ReturnMessage reply(call);
+ ::DBus::MessageIter wi = reply.writer();
+ wi << argout1;
+ return reply;
+ }
};
} } }
diff --git a/src/dbus/callmanager-introspec.xml b/src/dbus/callmanager-introspec.xml
index 6e415f9bfb71c26a48d3495733a9ced0545b1f32..9c503cac8dcd7c6a84d784b82d6854c2fcad029b 100644
--- a/src/dbus/callmanager-introspec.xml
+++ b/src/dbus/callmanager-introspec.xml
@@ -64,6 +64,16 @@
<method name="getCurrentCallID">
<arg type="s" name="callID" direction="out"/>
</method>
+
+ <method name="getCurrentCodecName">
+ <arg type="s" name="callID" direction="in"/>
+ <arg type="s" name="codecName" direction="out"/>
+ </method>
+
+ <signal name="currentSelectedCodec">
+ <arg type="s" name="callID" direction="out" />
+ <arg type="s" name="codecName" direction="out"/>
+ </signal>
<signal name="incomingCall">
<arg type="s" name="accountID" />
@@ -91,7 +101,6 @@
<arg type="s" name="device" direction="out"/>
<arg type="d" name="value" direction="out"/>
</signal>
-
<signal name="error">
diff --git a/src/dbus/callmanager.cpp b/src/dbus/callmanager.cpp
index d516dba31b305fe9c8e30e30cbb7ed5b8d628e46..126b7ef3b69ff5ed5a4afa8df9af05c545d5914f 100644
--- a/src/dbus/callmanager.cpp
+++ b/src/dbus/callmanager.cpp
@@ -101,7 +101,7 @@ CallManager::setVolume( const std::string& device, const double& value )
double
CallManager::getVolume( const std::string& device )
{
- _debug("CallManager::getVolume received\n");
+ _debug("CallManager::getVolume received \n");
if(device == "speaker")
{
_debug("Current speaker = %d\n", Manager::instance().getSpkrVolume());
@@ -123,6 +123,14 @@ CallManager::setRecording(const std::string& callID)
}
+std::string
+CallManager::getCurrentCodecName(const std::string& callID)
+{
+ _debug("CallManager::getCurrentCodecName received %s \n",Manager::instance().getCurrentCodecName(callID).c_str());
+ return Manager::instance().getCurrentCodecName(callID).c_str();
+}
+
+
std::map< std::string, std::string >
CallManager::getCallDetails( const std::string& callID UNUSED )
{
diff --git a/src/dbus/callmanager.h b/src/dbus/callmanager.h
index a8c4c0c2ed8dba4bc841d258ab7fa6a9c70bc2bc..7f2e7e689f5c6e2dc0f65cbe21d29c1492885e24 100644
--- a/src/dbus/callmanager.h
+++ b/src/dbus/callmanager.h
@@ -49,6 +49,7 @@ public:
void setVolume( const std::string& device, const double& value );
double getVolume( const std::string& device );
void setRecording( const std::string& callID );
+ std::string getCurrentCodecName(const std::string& callID);
std::map< std::string, std::string > getCallDetails( const std::string& callID );
std::string getCurrentCallID( );
void playDTMF( const std::string& key );
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index f5c65eea29615baf338a9346a120ab5c87d06fd1..231e024d4619433cbc8ad291411a1155a1183150 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -43,6 +43,7 @@
IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
: VoIPLink(accountID)
{
+ // _debug("IAXVoIPLink::IAXVoIPLink : creating eventhread \n ");
_evThread = new EventThread(this);
_regSession = NULL;
_nextRefreshStamp = 0;
@@ -89,7 +90,7 @@ IAXVoIPLink::init()
return false;
bool returnValue = false;
- //_localAddress = "127.0.0.1";
+ // _localAddress = "127.0.0.1";
// port 0 is default
// iax_enable_debug(); have to enable debug when compiling iax...
int port = IAX_DEFAULT_PORTNO;
@@ -207,10 +208,18 @@ IAXVoIPLink::getEvent()
iaxHandlePrecallEvent(event);
}
+ // _debug("IAXVoIPLink::getEvent() : timestamp %i \n",event->ts);
+
iax_event_free(event);
}
_mutexIAX.leaveMutex();
+
+ if(call){
+ // _debug("Are we recording");
+ call->recAudio.recData(spkrDataConverted,micData,nbSampleForRec_,nbSampleForRec_);
+ }
+
// Do the doodle-moodle to send audio from the microphone to the IAX channel.
sendAudioFromMic();
@@ -218,9 +227,13 @@ IAXVoIPLink::getEvent()
if (_nextRefreshStamp && _nextRefreshStamp - 2 < time(NULL)) {
sendRegister("");
}
- if(call){
- call->recAudio.recData(spkrDataConverted,micData,nbSampleForRec_,nbSampleForRec_);
- }
+
+ // _debug("IAXVoIPLink::getEvent() \n");
+ // reinitialize speaker buffer for recording (when recording a voice mail)
+ for (int i = 0; i < nbSampleForRec_; i++)
+ spkrDataConverted[i] = 0;
+
+
// thread wait 3 millisecond
_evThread->sleep(3);
free(event);
@@ -229,6 +242,8 @@ IAXVoIPLink::getEvent()
void
IAXVoIPLink::sendAudioFromMic(void)
{
+ // _debug("IAXVoIPLink::sendAudioFromMic");
+
int maxBytesToGet, availBytesFromMic, bytesAvail, compSize;
AudioCodec *ac;
@@ -289,7 +304,9 @@ IAXVoIPLink::sendAudioFromMic(void)
nbSample_ = audiolayer->getMic( micData, bytesAvail ) / sizeof(SFLDataFormat);
// Store the number of samples for recording
- nbSampleForRec_ = nbSample_;
+ nbSampleForRec_ = nbSample_;
+
+ // _debug("IAXVoIPLink::sendAudioFromMic : %i \n",nbSampleForRec_);
// resample
nbSample_ = converter->downsampleData( micData , micDataConverted , (int)ac ->getClockRate() , (int)audiolayer->getSampleRate() , nbSample_ );
@@ -400,7 +417,8 @@ IAXVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
if (call) {
call->setPeerNumber(toUrl);
-
+ call->initRecFileName();
+
if ( iaxOutgoingInvite(call) ) {
call->setConnectionState(Call::Progressing);
call->setState(Call::Active);
@@ -588,6 +606,16 @@ IAXVoIPLink::carryingDTMFdigits(const CallID& id, char code)
}
+std::string
+IAXVoIPLink::getCurrentCodecName()
+{
+ IAXCall *call = getIAXCall(Manager::instance().getCurrentCallId());
+
+ AudioCodec *ac = call->getCodecMap().getCodec(call->getAudioCodec());
+
+ return ac->getCodecName();
+}
+
bool
IAXVoIPLink::iaxOutgoingInvite(IAXCall* call)
@@ -717,6 +745,7 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
case IAX_EVENT_VOICE:
//if (!audiolayer->isCaptureActive ())
// audiolayer->startStream ();
+ // _debug("IAX_EVENT_VOICE: \n");
iaxHandleVoiceEvent(event, call);
break;
@@ -924,8 +953,11 @@ IAXVoIPLink::iaxHandlePrecallEvent(iax_event* event)
if (event->ies.calling_number)
call->setPeerNumber(std::string(event->ies.calling_number));
- if (event->ies.calling_name)
+ if (event->ies.calling_name)
call->setPeerName(std::string(event->ies.calling_name));
+
+ // if peerNumber exist append it to the name string
+ call->initRecFileName();
if (Manager::instance().incomingCall(call, getAccountID())) {
/** @todo Faudra considérer éventuellement le champ CODEC PREFS pour
diff --git a/src/iaxvoiplink.h b/src/iaxvoiplink.h
index 7cdaeaa5439782e9b2855b357eccdb549c8f45fd..2cd4d16357dab62162080f9d7003bd8a759aac6c 100644
--- a/src/iaxvoiplink.h
+++ b/src/iaxvoiplink.h
@@ -190,6 +190,13 @@ class IAXVoIPLink : public VoIPLink
bool isContactPresenceSupported() { return false; }
+ /**
+ * Return the codec protocol used for this call
+ * @param id The call identifier
+ */
+ std::string getCurrentCodecName();
+
+
public: // iaxvoiplink only
void updateAudiolayer( void );
diff --git a/src/main.cpp b/src/main.cpp
index 7ae724194b4ae53525d0587e490195b6e7772dc9..85f9170e5782cd44b28a9b4297447f4a2408c3ad 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -110,6 +110,7 @@ main (int argc, char **argv) {
}
bool initOK = false;
try {
+ _debug("Initialize manager in main \n");
Manager::instance().initConfigFile();
Manager::instance().init();
initOK = true;
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 7a98835bace3e7b923e73aae58c80ebd22052105..7afaf2aab64f8c498cc7921342cb66b753e5c039 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -188,6 +188,8 @@ ManagerImpl::switchCall(const CallID& id ) {
bool
ManagerImpl::outgoingCall(const std::string& accountid, const CallID& id, const std::string& to)
{
+ _debug("ManagerImpl::outgoingCall() method \n");
+
if (!accountExists(accountid)) {
_debug("! Manager Error: Outgoing Call: account doesn't exist\n");
return false;
@@ -219,6 +221,7 @@ ManagerImpl::outgoingCall(const std::string& accountid, const CallID& id, const
bool
ManagerImpl::answerCall(const CallID& id)
{
+
stopTone(false);
_debug("Try to answer call: %s\n", id.data());
AccountID accountid = getAccountFromCall( id );
@@ -237,11 +240,16 @@ ManagerImpl::answerCall(const CallID& id)
removeCallAccount(id);
return false;
}
-
+
// if it was waiting, it's waiting no more
if (_dbus) _dbus->getCallManager()->callStateChanged(id, "CURRENT");
removeWaitingCall(id);
switchCall(id);
+
+ std::string codecName = getCurrentCodecName(id);
+ _debug("ManagerImpl::hangupCall(): broadcast codec name %s \n",codecName.c_str());
+ if (_dbus) _dbus->getCallManager()->currentSelectedCodec(id,codecName.c_str());
+
return true;
}
@@ -249,7 +257,7 @@ ManagerImpl::answerCall(const CallID& id)
bool
ManagerImpl::hangupCall(const CallID& id)
{
- _debug("ManagerImpl::hangupCall(): This function is called when user hangup \n");
+ _debug("ManagerImpl::hangupCall(): This function is called when user hangup \n");
PulseLayer *pulselayer;
AccountID accountid;
bool returnValue;
@@ -361,6 +369,10 @@ ManagerImpl::offHoldCall(const CallID& id)
}
switchCall(id);
+ std::string codecName = getCurrentCodecName(id);
+ _debug("ManagerImpl::hangupCall(): broadcast codec name %s \n",codecName.c_str());
+ if (_dbus) _dbus->getCallManager()->currentSelectedCodec(id,codecName.c_str());
+
return returnValue;
}
@@ -574,7 +586,7 @@ ManagerImpl::isWaitingCall(const CallID& id) {
// Management of event peer IP-phone
////////////////////////////////////////////////////////////////////////////////
// SipEvent Thread
- bool
+bool
ManagerImpl::incomingCall(Call* call, const AccountID& accountId)
{
PulseLayer *pulselayer;
@@ -633,6 +645,10 @@ ManagerImpl::peerAnsweredCall(const CallID& id)
stopTone(false);
}
if (_dbus) _dbus->getCallManager()->callStateChanged(id, "CURRENT");
+
+ std::string codecName = getCurrentCodecName(id);
+ _debug("ManagerImpl::hangupCall(): broadcast codec name %s \n",codecName.c_str());
+ if (_dbus) _dbus->getCallManager()->currentSelectedCodec(id,codecName.c_str());
}
//THREAD=VoIP Call=Outgoing
@@ -738,7 +754,7 @@ bool ManagerImpl::playATone(Tone::TONEID toneId)
return false;
audiolayer = getAudioDriver();
-
+
if (_telephoneTone != 0) {
_toneMutex.enterMutex();
_telephoneTone->setCurrentTone(toneId);
@@ -1184,6 +1200,15 @@ ManagerImpl::getCodecDetails( const int32_t& payload )
return v;
}
+std::string
+ManagerImpl::getCurrentCodecName(const CallID& id)
+{
+ // _debug("ManagerImpl::getCurrentCodecName method called \n");
+ AccountID accountid = getAccountFromCall(id);
+ // _debug("ManagerImpl::getCurrentCodecName : %s \n",getAccountLink(accountid)->getCurrentCodecName().c_str());
+ return getAccountLink(accountid)->getCurrentCodecName();
+}
+
/**
* Get list of supported input audio plugin
*/
diff --git a/src/managerimpl.h b/src/managerimpl.h
index 51fefa04968eb802444b42e477edc4fcc1ffe884..0765bac47619e05229a9f0fbca829f97a17611f1 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -326,6 +326,13 @@ class ManagerImpl {
*/
std::vector< ::std::string > getCodecDetails( const int32_t& payload);
+ /**
+ * Get current codec name
+ * @param call id
+ * @return std::string The codec name
+ */
+ std::string getCurrentCodecName(const CallID& id);
+
/**
* Get a list of supported input audio plugin
* @return std::vector<std::string> List of names
diff --git a/src/plug-in/audiorecorder/audiorecord.cpp b/src/plug-in/audiorecorder/audiorecord.cpp
index 5e73d442db62a69cc449c199e7983fc8905977cb..4b0077a92cb1107801a41fccae5b639efb1c1f84 100644
--- a/src/plug-in/audiorecorder/audiorecord.cpp
+++ b/src/plug-in/audiorecorder/audiorecord.cpp
@@ -56,7 +56,7 @@ void AudioRecord::setSndSamplingRate(int smplRate){
void AudioRecord::setRecordingOption(FILE_TYPE type, SOUND_FORMAT format, int sndSmplRate, std::string path, std::string id){
- std::string fName;
+ // std::string fName;
fileType_ = type;
sndFormat_ = format;
@@ -64,10 +64,10 @@ void AudioRecord::setRecordingOption(FILE_TYPE type, SOUND_FORMAT format, int sn
sndSmplRate_ = sndSmplRate;
call_id_ = id;
- fName = fileName_;
- fName.append("-"+call_id_);
-
+ // fName = fileName_;
+ // fName.append("-"+call_id_);
+ /*
if (fileType_ == FILE_RAW){
if ( strstr(fileName_, ".raw") == NULL){
printf("AudioRecord::openFile::concatenate .raw file extension: name : %s \n", fileName_);
@@ -80,8 +80,34 @@ void AudioRecord::setRecordingOption(FILE_TYPE type, SOUND_FORMAT format, int sn
fName.append(".wav");
}
}
+ */
savePath_ = path + "/";
+ // savePath_.append(fName);
+}
+
+
+
+void AudioRecord::initFileName( std::string peerNumber){
+
+ std::string fName;
+
+ fName = fileName_;
+ fName.append("-"+peerNumber);
+
+ if (fileType_ == FILE_RAW){
+ if ( strstr(fileName_, ".raw") == NULL){
+ printf("AudioRecord::openFile::concatenate .raw file extension: name : %s \n", fileName_);
+ fName.append(".raw");
+ }
+ }
+ else if (fileType_ == FILE_WAV){
+ if ( strstr(fileName_, ".wav") == NULL){
+ printf("AudioRecord::openFile::concatenate .wav file extension: name : %s \n", fileName_);
+ fName.append(".wav");
+ }
+ }
+
savePath_.append(fName);
}
@@ -434,8 +460,8 @@ void AudioRecord::recData(SFLDataFormat* buffer_1, SFLDataFormat* buffer_2, int
if ( sndFormat_ == INT16 ) { // TODO change INT16 to SINT16
for (int k=0; k<nSamples_1; k++){
- mixBuffer_[k] = (buffer_1[k]+buffer_2[k])/2;
-
+ // mixBuffer_[k] = ((buffer_1[k]+buffer_2[k])/2)/32767;
+ mixBuffer_[k] = (buffer_1[k]+buffer_2[k]);
// dsp.getRMS(mixBuffer_[k]);
if ( fwrite(&buffer_1[k], 2, 1, fp) != 1)
diff --git a/src/plug-in/audiorecorder/audiorecord.h b/src/plug-in/audiorecorder/audiorecord.h
index d0d75465e234b0773db68339ecccad0cf1050fd2..532c96e4bd784dc96dfe31a05bf099f9d1380685 100644
--- a/src/plug-in/audiorecorder/audiorecord.h
+++ b/src/plug-in/audiorecorder/audiorecord.h
@@ -45,6 +45,8 @@ public:
void setRecordingOption(FILE_TYPE type, SOUND_FORMAT format, int sndSmplRate, std::string path, std::string id);
+ void initFileName( std::string peerNumber );
+
/**
* Check if no otehr file is opened, then create a new one
* @param fileName A string containing teh file (with/without extension)
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index 112d2d49692030ae9f31411e348acf5b38a94caa..bf7ac01b377f0a7c4a7ce515bb8021ba240eee64 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -420,6 +420,9 @@ SIPVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
}
//call->setPeerNumber(toUrl);
call->setPeerNumber(getSipTo(toUrl, account->getHostname()));
+
+ call->initRecFileName();
+
_debug("Try to make a call to: %s with call ID: %s\n", toUrl.data(), id.data());
// we have to add the codec before using it in SIPOutgoingInvite...
call->setCodecMap(Manager::instance().getCodecDescriptorMap());
@@ -443,7 +446,7 @@ SIPVoIPLink::answer(const CallID& id)
pj_status_t status;
pjsip_tx_data *tdata;
- _debug("- SIP Action: start answering\n");
+ _debug("SIPVoIPLink::answer: start answering \n");
call = getSIPCall(id);
@@ -455,22 +458,23 @@ SIPVoIPLink::answer(const CallID& id)
// User answered the incoming call, tell peer this news
if (call->startNegociation(_pool)) {
// Create and send a 200(OK) response
- _debug("UserAgent: Negociation success!\n");
+ _debug("SIPVoIPLink::answer:UserAgent: Negociation success! : call %s \n", call->getCallId().c_str());
status = pjsip_inv_answer(call->getInvSession(), PJSIP_SC_OK, NULL, NULL, &tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
status = pjsip_inv_send_msg(call->getInvSession(), tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
- _debug("* SIP Info: Starting AudioRTP when answering\n");
+ _debug("SIPVoIPLink::answer: Starting AudioRTP when answering : call %s \n", call->getCallId().c_str());
if (_audiortp->createNewSession(call) >= 0) {
call->setAudioStart(true);
call->setConnectionState(Call::Connected);
call->setState(Call::Active);
return true;
} else {
- _debug("! SIP Failure: Unable to start sound when answering %s/%d\n", __FILE__, __LINE__);
+ _debug("SIPVoIPLink::answer: Unable to start sound when answering %s/%d\n", __FILE__, __LINE__);
}
}
+ _debug("SIPVoIPLink::answer: fail terminate call %s \n",call->getCallId().c_str());
terminateOneCall(call->getCallId());
removeCall(call->getCallId());
return false;
@@ -796,6 +800,26 @@ SIPVoIPLink::isRecording(const CallID& id)
return call->isRecording();
}
+
+std::string
+SIPVoIPLink::getCurrentCodecName()
+{
+
+ _debug("SIPVoIPLink::getCurrentCodecName Ok 1 : \n");
+
+ printf("AAAAAAAAAAAAAHHHHHHH!!!!!!:: %s \n",Manager::instance().getCurrentCallId().c_str());
+
+ SIPCall *call = getSIPCall(Manager::instance().getCurrentCallId());
+
+ _debug("SIPVoIPLink::getCurrentCodecName Ok 2 : \n");
+
+ printf("OOOOOOOUUUUUUPPPPPPPPPPSSSSSSSSSSS!!!!!!:: %s \n",call->getCallId().c_str());
+
+ AudioCodec *ac = call->getCodecMap().getCodec(call->getAudioCodec());
+
+ return ac->getCodecName();
+}
+
bool
SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code)
{
@@ -1077,8 +1101,8 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
}
- SIPCall*
- SIPVoIPLink::getSIPCall(const CallID& id)
+SIPCall*
+SIPVoIPLink::getSIPCall(const CallID& id)
{
Call* call = getCall(id);
if (call) {
@@ -1087,9 +1111,11 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
return NULL;
}
- void SIPVoIPLink::setStunServer( const std::string &server )
- {
- if(server != "") {
+
+void SIPVoIPLink::setStunServer( const std::string &server )
+{
+ if(server != "") {
+
useStun(true);
_stunServer = server;
} else {
@@ -1838,6 +1864,8 @@ std::string SIPVoIPLink::getSipTo(const std::string& to_url, std::string hostnam
call->setIp(link->getLocalIPAddress());
call->setPeerNumber(peerNumber);
+ call->initRecFileName();
+
/* Call the SIPCallInvite function to generate the local sdp,
* remote sdp and negociator.
* This function is also used to set the parameters of audio RTP, including:
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index ac889c9493c6f69bc20e6a64cdc93dbdc91f6dbb..fef56310f9c5f8fe66c24d0d9d1c72c8d30ea320 100644
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -298,6 +298,13 @@ class SIPVoIPLink : public VoIPLink
*/
bool isRecording(const CallID& id);
+ /**
+ * Return the codec protocol used for this call
+ * @param id The call identifier
+ */
+ std::string getCurrentCodecName();
+
+
private:
/**
* Constructor
diff --git a/src/voiplink.h b/src/voiplink.h
index 92b7795eb479a987674e71cc9d75a88fcd181565..6de21106b354a5cab3e1694e3728742b0fb87b6e 100644
--- a/src/voiplink.h
+++ b/src/voiplink.h
@@ -173,6 +173,12 @@ class VoIPLink {
*/
virtual bool isRecording(const CallID& id) = 0;
+ /**
+ * Return the codec protocol used for this call
+ * @param id The call identifier
+ */
+ virtual std::string getCurrentCodecName() = 0;
+
bool initDone (void) { return _initDone; }
void initDone (bool state) { _initDone = state; }