diff --git a/sflphone-gtk/src/actions.c b/sflphone-gtk/src/actions.c
index 07f892264b1535111c388f9a2242c0ccb8c28cdf..bb4d6dbc644e0a408278aa7f9906da15b5408e84 100644
--- a/sflphone-gtk/src/actions.c
+++ b/sflphone-gtk/src/actions.c
@@ -208,10 +208,10 @@ sflphone_pick_up()
{
case CALL_STATE_DIALING:
sflphone_place_call (selectedCall);
- printf("accountID=%s\n",selectedCall->accountID);
+ //printf("accountID=%s\n",selectedCall->accountID);
break;
case CALL_STATE_INCOMING:
- printf("CALL ID = %s\n", selectedCall->callID);
+ //printf("CALL ID = %s\n", selectedCall->callID);
dbus_accept (selectedCall);
break;
case CALL_STATE_HOLD:
@@ -435,7 +435,6 @@ sflphone_keypad( guint keyval, gchar * key)
switch(c->state)
{
case CALL_STATE_DIALING: // Currently dialing => edit number
- //dbus_play_dtmf(key);
process_dialing(c, keyval, key);
break;
case CALL_STATE_CURRENT:
@@ -454,7 +453,7 @@ sflphone_keypad( guint keyval, gchar * key)
g_free(before);
g_free(temp);
//screen_set_call(c);
- update_call_tree(c);
+ //update_call_tree(c);
}
break;
}
@@ -507,7 +506,8 @@ sflphone_keypad( guint keyval, gchar * key)
switch (keyval)
{
case 65307: /* ESCAPE */
- dbus_hang_up(c);
+ //dbus_hang_up(c);
+ sflphone_hang_up(c);
break;
}
break;
diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index 38337dbc14f3af78dc1f7d4bd1b11855dd02a5aa..06583b3b01ce52bec8ed1bfb6c9f71a39d7927e5 100644
--- a/src/audio/audiolayer.cpp
+++ b/src/audio/audiolayer.cpp
@@ -43,7 +43,6 @@
, _CaptureHandle( NULL )
, deviceClosed( true )
, _urgentBuffer( SIZEBUF )
- , _fstream("/tmp/audio.dat")
{
_inChannel = 1; // don't put in stereo
@@ -59,8 +58,6 @@ AudioLayer::~AudioLayer (void)
closeCaptureStream();
closePlaybackStream();
deviceClosed = true;
- _fstream.flush();
- _fstream.close();
}
@@ -264,7 +261,6 @@ AudioLayer::playTones( void )
int spkrVol = _manager -> getSpkrVolume();
if( tone != 0 ){
tone -> getNext( out , frames , spkrVol );
- //_fstream.write( (char*)out, maxBytes );
write( out , maxBytes );
}
else if( ( tone=_manager->getTelephoneFile() ) != 0 ){
@@ -413,7 +409,6 @@ AudioLayer::write(void* buffer, int length)
//handle_xrun_playback();
//_debugAlsa("avail = %d - toWrite = %d\n" , snd_pcm_avail_update( _PlaybackHandle ) , length / 2);
-
snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames( _PlaybackHandle, length);
int err = snd_pcm_mmap_writei( _PlaybackHandle , buffer , frames );
switch(err) {
@@ -444,7 +439,6 @@ AudioLayer::write(void* buffer, int length)
int
AudioLayer::read( void* buffer, int toCopy)
{
-
if(deviceClosed || _CaptureHandle == NULL)
return 0;
int err;
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index faaa07e510db33966763aae8c04bee8503571cb0..af8038886f2f2ac922dce49441a0dadc0f8adb60 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -180,7 +180,6 @@ IAXVoIPLink::terminateIAXCall()
iter++;
}
_callMap.clear();
- delete _audiocodec;
}
void
@@ -204,6 +203,7 @@ IAXVoIPLink::getEvent()
if (call) {
// We know that call, deal with it
iaxHandleCallEvent(event, call);
+ //_audiocodec = Manager::instance().getCodecDescriptorMap().getCodec( call -> getAudioCodec() );
}
else if (event->session && event->session == _regSession) {
// This is a registration session, deal with it
@@ -235,12 +235,16 @@ IAXVoIPLink::getEvent()
void
IAXVoIPLink::sendAudioFromMic(void)
{
- IAXCall* currentCall = getIAXCall(Manager::instance().getCurrentCallId());
+ 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",
@@ -252,8 +256,9 @@ IAXVoIPLink::sendAudioFromMic(void)
return;
}
- _audiocodec = Manager::instance().getCodecDescriptorMap().getCodec( currentCall -> getAudioCodec() );
- if (!_audiocodec) {
+ AudioCodec* ac = currentCall -> getCodecMap().getCodec( currentCall -> getAudioCodec() );
+ _debug("Audiocodec name = %s\n" , ac->getCodecName().c_str());
+ if (!ac) {
// Audio codec still not determined.
if (audiolayer) {
// To keep latency low..
@@ -285,10 +290,9 @@ IAXVoIPLink::sendAudioFromMic(void)
int nbSample = audiolayer->getMic(_dataAudioLayer, bytesAvail) / sizeof(SFLDataFormat);
// Audio ici est PARFAIT
-
int16* toIAX = NULL;
//if (audiolayer->getSampleRate() != audiocodec->getClockRate() && nbSample) {
- if (audiolayer->getSampleRate() != _audiocodec->getClockRate() && nbSample) {
+ if (audiolayer->getSampleRate() != ac ->getClockRate() && nbSample) {
SRC_DATA src_data;
#ifdef DATAFORMAT_IS_FLOAT
src_data.data_in = _dataAudioLayer;
@@ -298,8 +302,7 @@ IAXVoIPLink::sendAudioFromMic(void)
#endif
// Audio parfait à ce point.
-
- double factord = (double) _audiocodec->getClockRate() / audiolayer->getSampleRate();
+ double factord = (double) ac->getClockRate() / audiolayer->getSampleRate();
src_data.src_ratio = factord;
src_data.input_frames = nbSample;
@@ -312,12 +315,10 @@ IAXVoIPLink::sendAudioFromMic(void)
nbSample = src_data.output_frames_gen;
// Bon, l'audio en float 8000 est laid mais yé consistant.
-
src_float_to_short_array (_floatBuffer8000, _intBuffer8000, nbSample);
toIAX = _intBuffer8000;
// Audio bon ici aussi..
-
} else {
#ifdef DATAFORMAT_IS_FLOAT
// convert _receiveDataDecoded to float inside _receiveData
@@ -329,34 +330,8 @@ IAXVoIPLink::sendAudioFromMic(void)
#endif
}
- // NOTE: L'audio ici est bon.
-
- //
- // LE PROBLÈME est dans cette snippet de fonction:
- // C'est une fonction destructrice ! On n'en veut pas!
- //if ( nbSample < (IAX__20S_8KHZ_MAX - 10) ) { // if only 10 is missing, it's ok
- // fill end with 0...
- //_debug("begin: %p, nbSample: %d\n", toIAX, nbSample);
- //_debug("has to fill: %d chars at %p\n", (IAX__20S_8KHZ_MAX-nbSample)*sizeof(int16), toIAX + nbSample);
- //memset(toIAX + nbSample, 0, (IAX__20S_8KHZ_MAX-nbSample)*sizeof(int16));
- //nbSample = IAX__20S_8KHZ_MAX;
- //}
-
- //_debug("AR: Nb sample: %d int, [0]=%d [1]=%d [2]=%d\n", nbSample, toIAX[0], toIAX[1], toIAX[2]);
- // NOTE: Le son dans toIAX (nbSamle*sizeof(int16)) est mauvais,
- // s'il passe par le snippet précédent.
-
-
- // DEBUG
- //_fstream.write((char *) toIAX, nbSample*sizeof(int16));
- //_fstream.flush();
-
-
// for the mono: range = 0 to IAX_FRAME2SEND * sizeof(int16)
- int compSize = _audiocodec->codecEncode(_sendDataEncoded, toIAX, nbSample*sizeof(int16));
-
-
-
+ int compSize = ac->codecEncode(_sendDataEncoded, toIAX, nbSample*sizeof(int16));
// Send it out!
_mutexIAX.enterMutex();
@@ -368,7 +343,6 @@ IAXVoIPLink::sendAudioFromMic(void)
}
_mutexIAX.leaveMutex();
}
- //unloadCodec(audiocodec);
}
@@ -488,6 +462,7 @@ bool
IAXVoIPLink::answer(const CallID& id)
{
IAXCall* call = getIAXCall(id);
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
CHK_VALID_CALL;
@@ -497,7 +472,6 @@ IAXVoIPLink::answer(const CallID& id)
call->setState(Call::Active);
call->setConnectionState(Call::Connected);
-
// Start audio
audiolayer->startStream();
//audiolayer->flushMic();
@@ -769,16 +743,18 @@ IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
}
if (audiolayer) {
- _audiocodec = Manager::instance().getCodecDescriptorMap().getCodec( call -> getAudioCodec() );
+ // _debug("codec = %i\n" , call->getFormat());
+ // _debug("codec = %i\n" , _audiocodec->getPayload());
+ //_debug("codec = %s\n" , _audiocodec->getCodecName().c_str());
// On-the-fly codec changing (normally, when we receive a full packet)
// as per http://tools.ietf.org/id/draft-guy-iax-03.txt
// - subclass holds the voiceformat property.
if (event->subclass && event->subclass != call->getFormat()) {
call->setFormat(event->subclass);
}
- //audiocodec = loadCodec(call->getAudioCodec());
//_debug("Receive: len=%d, format=%d, _receiveDataDecoded=%p\n", event->datalen, call->getFormat(), _receiveDataDecoded);
-
+ AudioCodec* ac = call->getCodecMap().getCodec( call -> getAudioCodec() );
+
unsigned char* data = (unsigned char*)event->data;
unsigned int size = event->datalen;
@@ -787,7 +763,7 @@ IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
size = IAX__20S_8KHZ_MAX;
}
- int expandedSize = _audiocodec->codecDecode(_receiveDataDecoded, data, size);
+ int expandedSize = ac->codecDecode(_receiveDataDecoded, data, size);
int nbInt16 = expandedSize/sizeof(int16);
if (nbInt16 > IAX__20S_8KHZ_MAX) {
@@ -799,9 +775,9 @@ IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
int nbSample = nbInt16;
int nbSampleMaxRate = nbInt16 * 6;
- if ( audiolayer->getSampleRate() != _audiocodec->getClockRate() && nbSample ) {
+ if ( audiolayer->getSampleRate() != ac->getClockRate() && nbSample ) {
// Do sample rate conversion
- double factord = (double) audiolayer->getSampleRate() / _audiocodec->getClockRate();
+ double factord = (double) audiolayer->getSampleRate() / ac->getClockRate();
// SRC_DATA from samplerate.h
SRC_DATA src_data;
src_data.data_in = _floatBuffer8000;
@@ -837,7 +813,6 @@ IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
} else {
_debug("IAX: incoming audio, but no sound card open");
}
- //unloadCodec(audiocodec);
}
@@ -901,6 +876,7 @@ IAXVoIPLink::iaxHandlePrecallEvent(iax_event* event)
id = Manager::instance().getNewCallID();
+
call = new IAXCall(id, Call::Incoming);
if (!call) {
diff --git a/src/iaxvoiplink.h b/src/iaxvoiplink.h
index 9043c2e4dbda99abf72e7ead305538a77bea1cf6..05f1a08735d3fe9350bd0a426f63f98db690ede2 100644
--- a/src/iaxvoiplink.h
+++ b/src/iaxvoiplink.h
@@ -211,11 +211,6 @@ private:
/** libsamplerate error */
int _src_err;
- /** Debugging output file
- * @todo Remove this */
- //std::ofstream _fstream;
-
- AudioCodec* _audiocodec;
};
#endif
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 17d38782c30d8d20b52aa241fdd0c77f657ff5a6..cd0b8da0eefbc22a0344fba0fb613d48faf2c79a 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -719,8 +719,8 @@ ManagerImpl::callBusy(const CallID& id) {
void
ManagerImpl::callFailure(const CallID& id)
{
- _debug("Call failed\n");
if (_dbus) _dbus->getCallManager()->callStateChanged(id, "FAILURE");
+ _debug("CALL ID = %s\n" , id.c_str());
if (isCurrentCall(id) ) {
playATone(Tone::TONE_BUSY);
switchCall("");
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index e61224cb931cfb2bccaa7309245b1e15ac5a679b..6b56befcb370fe8e427ed7850c6ecf7d5d2a6892 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -860,7 +860,7 @@ SIPVoIPLink::transfer(const CallID& id, const std::string& to)
eXosip_unlock();
// shall we delete the call?
- removeCall(id);
+ //removeCall(id);
return true;
}