From 9b32b8112770fac4b78514a35340cf1e2db8e429 Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandre.savard@savoirfairelinux.net>
Date: Thu, 19 Feb 2009 13:01:11 -0500
Subject: [PATCH] Display codec used
---
sflphone-gtk/src/actions.c | 31 +++++++++++++++++++++
sflphone-gtk/src/actions.h | 4 +++
sflphone-gtk/src/callmanager-glue.h | 38 ++++++++++++++++++++++++++
sflphone-gtk/src/dbus.c | 42 ++++++++++++++++++++++++++++-
sflphone-gtk/src/dbus.h | 6 +++++
src/audio/audiortp.cpp | 2 ++
src/audio/audiortp.h | 8 ++----
src/dbus/callmanager-glue.h | 35 ++++++++++++++++++++++++
src/dbus/callmanager-introspec.xml | 11 +++++++-
src/dbus/callmanager.cpp | 10 ++++++-
src/dbus/callmanager.h | 1 +
src/iaxvoiplink.cpp | 31 ++++++++++++++++-----
src/iaxvoiplink.h | 7 +++++
src/managerimpl.cpp | 19 ++++++++++++-
src/managerimpl.h | 7 +++++
src/sipvoiplink.cpp | 14 ++++++++++
src/sipvoiplink.h | 7 +++++
src/voiplink.h | 6 +++++
18 files changed, 262 insertions(+), 17 deletions(-)
diff --git a/sflphone-gtk/src/actions.c b/sflphone-gtk/src/actions.c
index 9de09864e0..96a9db1111 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 */
@@ -733,6 +735,31 @@ 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()
{
@@ -754,6 +781,9 @@ 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 */
@@ -816,3 +846,4 @@ sflphone_fill_codec_list()
exit(0);
}
}
+
diff --git a/sflphone-gtk/src/actions.h b/sflphone-gtk/src/actions.h
index ebe0157fe8..7c22eeb1b9 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 c4123cd1e3..1921fbb0e0 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/dbus.c b/sflphone-gtk/src/dbus.c
index 305a20bd91..0e3ddd9303 100644
--- a/sflphone-gtk/src/dbus.c
+++ b/sflphone-gtk/src/dbus.c
@@ -61,6 +61,15 @@ 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,
@@ -265,6 +274,14 @@ 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);
@@ -782,6 +799,30 @@ 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**
@@ -1298,7 +1339,6 @@ dbus_set_record(const call_t * c)
{
g_error_free(error);
}
- g_print("called dbus_set_record on CallManager\n");
}
void
diff --git a/sflphone-gtk/src/dbus.h b/sflphone-gtk/src/dbus.h
index b6c1ebddfd..f347a4a47e 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/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index e50cb2060b..d294b8b7d4 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -410,6 +410,8 @@ 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
diff --git a/src/audio/audiortp.h b/src/audio/audiortp.h
index 8b54fc1700..a0a3cf6846 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;
@@ -170,10 +170,6 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
/** The audio codec used during the session */
AudioCodec* _audiocodec;
- /**
- * Audio recording object
- */
- // AudioRecord recAudio;
};
///////////////////////////////////////////////////////////////////////////////
@@ -206,7 +202,7 @@ class AudioRtp {
* Start recording
*/
void setRecording ();
-
+
private:
// copy constructor
diff --git a/src/dbus/callmanager-glue.h b/src/dbus/callmanager-glue.h
index 4abf94d05c..71dd6b5170 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 6e415f9bfb..9c503cac8d 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 d516dba31b..126b7ef3b6 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 a8c4c0c2ed..7f2e7e689f 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 688fd7f37b..99ab771c6e 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -213,6 +213,12 @@ IAXVoIPLink::getEvent()
}
_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();
@@ -220,12 +226,8 @@ IAXVoIPLink::getEvent()
if (_nextRefreshStamp && _nextRefreshStamp - 2 < time(NULL)) {
sendRegister("");
}
-
- if(call){
- // _debug("Are we recording");
- 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;
@@ -239,6 +241,8 @@ IAXVoIPLink::getEvent()
void
IAXVoIPLink::sendAudioFromMic(void)
{
+ // _debug("IAXVoIPLink::sendAudioFromMic");
+
int maxBytesToGet, availBytesFromMic, bytesAvail, compSize;
AudioCodec *ac;
@@ -299,7 +303,9 @@ 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_ );
@@ -600,6 +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)
@@ -730,6 +746,7 @@ 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;
diff --git a/src/iaxvoiplink.h b/src/iaxvoiplink.h
index 7cdaeaa543..2cd4d16357 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/managerimpl.cpp b/src/managerimpl.cpp
index 3352eeb310..67653bb71a 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -239,6 +239,10 @@ ManagerImpl::answerCall(const CallID& id)
removeCallAccount(id);
return false;
}
+
+ 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());
// if it was waiting, it's waiting no more
if (_dbus) _dbus->getCallManager()->callStateChanged(id, "CURRENT");
@@ -251,7 +255,7 @@ 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;
@@ -635,6 +639,10 @@ 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
@@ -1186,6 +1194,15 @@ 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 51fefa0496..0765bac476 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/sipvoiplink.cpp b/src/sipvoiplink.cpp
index febcdf4ef1..e7223bfe4e 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -799,6 +799,20 @@ SIPVoIPLink::isRecording(const CallID& id)
return call->isRecording();
}
+
+std::string
+SIPVoIPLink::getCurrentCodecName()
+{
+
+ // _debug("SIPVoIPLink::getCurrentCodecName : \n");
+
+ SIPCall *call = getSIPCall(Manager::instance().getCurrentCallId());
+
+ AudioCodec *ac = call->getCodecMap().getCodec(call->getAudioCodec());
+
+ return ac->getCodecName();
+}
+
bool
SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code)
{
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index ac889c9493..fef56310f9 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 92b7795eb4..6de21106b3 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; }
--
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