diff --git a/sflphone-gtk/src/actions.c b/sflphone-gtk/src/actions.c
index 8e0839237eedfea8ed94d1804744bb51cf539dcb..9bd34bd59445b3e889e486c8b7343719f8ad0fa3 100644
--- a/sflphone-gtk/src/actions.c
+++ b/sflphone-gtk/src/actions.c
@@ -259,6 +259,7 @@ sflphone_hang_up()
case CALL_STATE_HOLD:
case CALL_STATE_BUSY:
case CALL_STATE_FAILURE:
+ case CALL_STATE_RECORDING:
dbus_hang_up (selectedCall);
selectedCall->state = CALL_STATE_DIALING;
(void) time(&selectedCall->_stop);
@@ -330,6 +331,10 @@ sflphone_on_hold ()
case CALL_STATE_CURRENT:
dbus_hold (selectedCall);
break;
+ case CALL_STATE_RECORD:
+ dbus_hold (selectedCall);
+ break;
+
default:
g_warning("Should not happen in sflphone_on_hold!");
break;
@@ -714,17 +719,16 @@ sflphone_rec_call()
{
case CALL_STATE_CURRENT:
selectedCall->state = CALL_STATE_RECORD;
- update_call_tree(current_calls,selectedCall);
- update_menus();
break;
case CALL_STATE_RECORD:
selectedCall->state = CALL_STATE_CURRENT;
- update_call_tree(current_calls,selectedCall);
- update_menus();
break;
default:
+ g_warning("Should not happen in sflphone_off_hold ()!");
break;
}
+ update_call_tree(current_calls,selectedCall);
+ update_menus();
}
/* Internal to action - set the __CURRENT_ACCOUNT variable */
@@ -761,7 +765,7 @@ sflphone_fill_codec_list()
c->_bandwidth = atof(details[3]);
codec_list_add(c);
}
-
+
for(pl=codecs; *codecs; codecs++)
{
details = (gchar **)dbus_codec_details(atoi(*codecs));
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index b4c1bd0d095329b65d1b3e80cb74ca59ef299c6f..785078c0c8c18eafe02286f71da6689dbb2c73c7 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -101,7 +101,8 @@ void
AudioRtp::setRecording() {
_debug("AudioRtp::setRecording");
- _RTXThread->recAudio.setRecording();
+ // _RTXThread->recAudio.setRecording();
+ _RTXThread->_ca->setRecording();
}
@@ -203,7 +204,11 @@ AudioRtpRTX::initAudioRtpSession (void)
FILE_TYPE ft = FILE_WAV;
SOUND_FORMAT sf = INT16;
recAudio.setSndSamplingRate(44100);
- recAudio.openFile(_ca->getFileName(),ft,sf,_ca->getCallId());
+ recAudio.openFile(_ca->getFileName(),ft,sf);
+
+ //_debug("Opening the wave file in call nstance\n");
+ //_ca->recAudio.setSndSamplingRate(44100);
+ //_ca->recAudio.openFile("testRecFromCall",ft,sf,_ca->getCallId());
if (!_sym) {
@@ -459,13 +464,17 @@ AudioRtpRTX::run () {
// Let's wait for the next transmit cycle
recAudio.recData(spkrDataConverted,micData,_nSamplesSpkr,_nSamplesMic);
-
+ _ca->recAudio.recData(spkrDataConverted,micData,_nSamplesSpkr,_nSamplesMic);
+
Thread::sleep(TimerPort::getTimer());
TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
}
_debug("Close wave file\n");
recAudio.closeFile();
+
+ //_debug("Close wave file in call instance\n");
+ //_ca->recAudio.closeFile();
// _debug("stop stream for audiortp loop\n");
audiolayer->stopStream();
@@ -475,6 +484,9 @@ AudioRtpRTX::run () {
_debug("! Close wave file\n");
recAudio.closeFile();
+
+ //_debug("Close wave file in call instance\n");
+ //_ca->recAudio.closeFile();
throw;
} catch(...) {
@@ -482,6 +494,9 @@ AudioRtpRTX::run () {
_debugException("* ARTP Action: Stop");
_debug("* Close wave file\n");
recAudio.closeFile();
+
+ //_debug("Close wave file in call instance\n");
+ //_ca->recAudio.closeFile();
throw;
}
diff --git a/src/audio/audiortp.h b/src/audio/audiortp.h
index 30e2333901aea0af455291dd978240b860bcab93..89b0f257905823cb8954e20c8f7c92c862fa9f60 100644
--- a/src/audio/audiortp.h
+++ b/src/audio/audiortp.h
@@ -29,7 +29,7 @@
#include <cc++/numbers.h>
#include "../global.h"
-#include "plug-in/audiorecorder/audiorecord.h"
+// #include "plug-in/audiorecorder/audiorecord.h"
#include "../samplerateconverter.h"
#define UP_SAMPLING 0
@@ -71,6 +71,9 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
*/
AudioRecord recAudio;
+ /** A SIP call */
+ SIPCall* _ca;
+
private:
// copy constructor
@@ -79,9 +82,6 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
// assignment operator
AudioRtpRTX& operator=(const AudioRtpRTX& rh);
- /** A SIP call */
- SIPCall* _ca;
-
/** RTP session to send data */
ost::RTPSession *_sessionSend;
diff --git a/src/call.cpp b/src/call.cpp
index dc74fcc3f4e022c7332159b27d6a3dfd59b3a58c..3e0074532594260036b28a3748f5ee7dde0b2f65 100644
--- a/src/call.cpp
+++ b/src/call.cpp
@@ -55,11 +55,21 @@ Call::Call(const CallID& id, Call::CallType type)
_filename = out.str();
printf("Call::constructor filename for tis call %s \n",_filename.c_str());
+
+
+
+ FILE_TYPE ft = FILE_WAV;
+ SOUND_FORMAT sf = INT16;
+ _debug("CALL::Opening the wave file in call nstance\n");
+ recAudio.setSndSamplingRate(44100);
+ recAudio.openFile("testRecFromCall",ft,sf);
}
Call::~Call()
{
+ _debug("CALL::Close wave file in call instance\n");
+ recAudio.closeFile();
}
void
@@ -146,3 +156,8 @@ Call::isAudioStarted()
return _audioStarted;
}
+void
+Call::setRecording()
+{
+ recAudio.setRecording();
+}
diff --git a/src/call.h b/src/call.h
index 2de79c722f7db181ef938548d8163146270ae5b3..e14e01712e664fc2e388ffdb1d24ba914f25c655 100644
--- a/src/call.h
+++ b/src/call.h
@@ -218,7 +218,13 @@ class Call{
/**
* A recorder fro this call
*/
- AudioRecord audioRec;
+ AudioRecord recAudio;
+
+ /**
+ * SetRecording
+ */
+ void setRecording();
+
protected:
/** Protect every attribute that can be changed by two threads */
diff --git a/src/plug-in/audiorecorder/audiorecord.cpp b/src/plug-in/audiorecorder/audiorecord.cpp
index 6921bf60dd878d9046786a412de960e57d8325ff..e05f3af1a1784c42994c338da2813a54f9c82b99 100644
--- a/src/plug-in/audiorecorder/audiorecord.cpp
+++ b/src/plug-in/audiorecorder/audiorecord.cpp
@@ -36,7 +36,7 @@ void AudioRecord::setSndSamplingRate(int smplRate){
}
-void AudioRecord::openFile(std::string name, FILE_TYPE type, SOUND_FORMAT format, CallID& id){
+void AudioRecord::openFile(std::string name, FILE_TYPE type, SOUND_FORMAT format){
_debug("AudioRecord::openFile()\n");
diff --git a/src/plug-in/audiorecorder/audiorecord.h b/src/plug-in/audiorecorder/audiorecord.h
index e1f6c79c867537686ca3d9d8c6610490a3a74da7..a15e8a1c78093893ed7e812e86e28ac416db36bf 100644
--- a/src/plug-in/audiorecorder/audiorecord.h
+++ b/src/plug-in/audiorecorder/audiorecord.h
@@ -61,7 +61,7 @@ public:
* @param type The sound file format (FILE_RAW, FILE_WAVE)
* @param format Internal sound format (INT16 / INT32)
*/
- void openFile(std::string fileName, FILE_TYPE type, SOUND_FORMAT format, CallID& id);
+ void openFile(std::string fileName, FILE_TYPE type, SOUND_FORMAT format);
/**
* Close the opend recording file. If wave: cout the number of byte