* #6991 : fix IAX problems

.gitignore

@@ -42,7 +42,8 @@ doc/doxygen/gtk-gui-doc
 doc/*.html
 
 # Ignore built stuff
-/src/.libs
+.libs
+.deps
 
 # PJSIP files
 *.depend

daemon/src/audio/alsa/alsalayer.cpp

@@ -523,7 +523,7 @@ namespace
 
 void AlsaLayer::capture(void)
 {
-    unsigned int mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
+    unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
     bool resample = audioSampleRate_ != mainBufferSampleRate;
 
     int toGetSamples = snd_pcm_avail_update (captureHandle_);
@@ -550,11 +550,11 @@ void AlsaLayer::capture(void)
         SFLDataFormat* rsmpl_out = (SFLDataFormat*) malloc (outBytes);
         converter_->resample ( (SFLDataFormat*) in, rsmpl_out, mainBufferSampleRate, audioSampleRate_, toGetSamples);
         dcblocker_.process(rsmpl_out, rsmpl_out, outSamples);
-        getMainBuffer()->putData (rsmpl_out, outBytes);
+        Manager::instance().getMainBuffer()->putData (rsmpl_out, outBytes);
         free (rsmpl_out);
     } else {
         dcblocker_.process(in, in, toGetSamples);
-		getMainBuffer()->putData (in, toGetBytes);
+        Manager::instance().getMainBuffer()->putData (in, toGetBytes);
     }
 
 end:
@@ -565,10 +565,10 @@ void AlsaLayer::playback(int maxSamples)
 {
     unsigned short spkrVolume = Manager::instance().getSpkrVolume();
 
-    unsigned int mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
+    unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
     bool resample = audioSampleRate_ != mainBufferSampleRate;
 
-    int toGet = getMainBuffer()->availForGet();
+    int toGet = Manager::instance().getMainBuffer()->availForGet();
     int toPut = maxSamples * sizeof(SFLDataFormat);
 
     if (toGet <= 0) {    	// no audio available, play tone or silence
@@ -602,7 +602,7 @@ void AlsaLayer::playback(int maxSamples)
 		toGet = maxNbBytesToGet;
 
 	SFLDataFormat *out = (SFLDataFormat*) malloc (toGet);
-	getMainBuffer()->getData (out, toGet);
+	Manager::instance().getMainBuffer()->getData (out, toGet);
 	adjustVolume(out, toGet / sizeof(SFLDataFormat), spkrVolume);
 
 	if (resample) {
@@ -644,7 +644,7 @@ void AlsaLayer::audioCallback (void)
 		write (out, toGet, playbackHandle_);
 		free (out);
         // Consume the regular one as well (same amount of bytes)
-        getMainBuffer()->discard (toGet);
+		Manager::instance().getMainBuffer()->discard (toGet);
     } else {
     	// regular audio data
     	playback(playbackAvailSmpl);

daemon/src/audio/audiolayer.cpp

@@ -56,7 +56,7 @@ void AudioLayer::flushMain (void)
 {
     ost::MutexLock guard (mutex_);
     // should pass call id
-    getMainBuffer()->flushAllBuffers();
+    Manager::instance().getMainBuffer()->flushAllBuffers();
 }
 
 void AudioLayer::flushUrgent (void)

daemon/src/audio/audiolayer.h

@@ -111,18 +111,6 @@ class AudioLayer
             return audioSampleRate_;
         }
 
-        /**
-             * Get a pointer to the application MainBuffer class.
-         *
-         * In order to send signal to other parts of the application, one must pass through the mainbuffer.
-         * Audio instances must be registered into the MainBuffer and bound together via the ManagerImpl.
-         *
-             * @return MainBuffer* a pointer to the MainBuffer instance
-             */
-        MainBuffer* getMainBuffer (void) const {
-            return Manager::instance().getMainBuffer();
-        }
-
         /**
          * Get the mutex lock for the entire audio layer
          */

daemon/src/audio/pulseaudio/pulselayer.cpp

@@ -434,7 +434,7 @@ void PulseLayer::writeToSpeaker (void)
         urgentRingBuffer_.Get (data, urgentBytes);
         pa_stream_write (s, data, urgentBytes, NULL, 0, PA_SEEK_RELATIVE);
         // Consume the regular one as well (same amount of bytes)
-        getMainBuffer()->discard (urgentBytes);
+        Manager::instance().getMainBuffer()->discard (urgentBytes);
         return;
     }
 
@@ -450,7 +450,7 @@ void PulseLayer::writeToSpeaker (void)
 
 	flushUrgent(); // flush remaining samples in _urgentRingBuffer
 
-	size_t availSamples = getMainBuffer()->availForGet() / sizeof(SFLDataFormat);
+	size_t availSamples = Manager::instance().getMainBuffer()->availForGet() / sizeof(SFLDataFormat);
 	if (availSamples == 0) {
 		pa_stream_begin_write(s, &data, &bytes);
 		memset(data, 0, bytes);
@@ -458,7 +458,7 @@ void PulseLayer::writeToSpeaker (void)
 		return;
 	}
 
-	unsigned int mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
+	unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
 	bool resample = audioSampleRate_ != mainBufferSampleRate;
 
 	// how much samples we can write in the output
@@ -479,7 +479,7 @@ void PulseLayer::writeToSpeaker (void)
 
 	size_t inBytes = inSamples * sizeof (SFLDataFormat);
 	pa_stream_begin_write(s, &data, &inBytes);
-	getMainBuffer()->getData (data, inBytes);
+	Manager::instance().getMainBuffer()->getData (data, inBytes);
 
 	if (resample) {
 		SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (outBytes);
@@ -502,7 +502,7 @@ void PulseLayer::readFromMic (void)
         return;
     }
 
-	unsigned int mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
+	unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
 	bool resample = audioSampleRate_ != mainBufferSampleRate;
 	if (resample) {
 		double resampleFactor = (double) audioSampleRate_ / mainBufferSampleRate;
@@ -520,7 +520,7 @@ void PulseLayer::readFromMic (void)
 		converter_->resample((SFLDataFormat*)data, mic_buffer_, mainBufferSampleRate, audioSampleRate_, samples);
 
 	dcblocker_.process(mic_buffer_, resample ? mic_buffer_ : (SFLDataFormat*)data, samples);
-	getMainBuffer()->putData(mic_buffer_, bytes);
+	Manager::instance().getMainBuffer()->putData(mic_buffer_, bytes);
 
     if (pa_stream_drop (record_->pulseStream()) < 0)
         _error ("Audio: Error: capture stream drop failed: %s" , pa_strerror (pa_context_errno (context_)));

daemon/src/iax/iaxcall.cpp

@@ -72,17 +72,10 @@ namespace {
     }
 }
 
-IAXCall::IAXCall (const std::string& id, Call::CallType type) : Call (id, type), _session (NULL)
+IAXCall::IAXCall (const std::string& id, Call::CallType type) : Call (id, type), session (NULL)
 {
 }
 
-void
-IAXCall::setFormat (int format)
-{
-    _format = format;
-    setAudioCodec(ASTFormatToCodec(format));
-}
-
 int
 IAXCall::getSupportedFormat (const std::string &accountID) const
 {
@@ -118,7 +111,7 @@ int IAXCall::getFirstMatchingFormat (int needles, const std::string &accountID)
     return 0;
 }
 
-int IAXCall::getAudioCodec()
+int IAXCall::getAudioCodec(void)
 {
-    return _audioCodec;
+    return ASTFormatToCodec(format);
 }

daemon/src/iax/iaxcall.h

@@ -50,38 +50,6 @@ class IAXCall : public Call
          */
         IAXCall (const std::string& id, Call::CallType type);
 
-        /**
-         * @return iax_session* The session pointer or NULL
-         */
-        iax_session* getSession() const {
-            return _session;
-        }
-
-        /**
-         * Set the session pointer
-         * @param session the session pointer to assign
-         */
-        void setSession (iax_session* session) {
-            _session = session;
-        }
-
-        /**
-         * Set format (one single bit)
-         * This function sets the _audioCodec variable with the correct
-         * codec.
-         * @param format  The format representing the codec
-         */
-        void setFormat (int format);
-
-        /**
-         * Get format for the voice codec used
-         * @return int  Bitmask for codecs defined in iax/frame.h
-         */
-        int getFormat() const {
-            return _format;
-        }
-
-
         /**
          * @return int  The bitwise list of supported formats
          */
@@ -102,32 +70,10 @@ class IAXCall : public Call
          */
         int getFirstMatchingFormat (int needles, const std::string &accountID) const;
 
-        /**
-         * Return audio codec [mutex protected]
-         * @return int The payload of the codec
-         */
-        int getAudioCodec();
-
-    private:
-        /** Each call is associated with an iax_session */
-        iax_session* _session;
+        int getAudioCodec(void);
 
-        /**
-         * Set the audio codec used.  [not protected]
-         * @param audioCodec  The payload of the codec
-         */
-        void setAudioCodec (int audioCodec) {
-            _audioCodec = audioCodec;
-        }
-
-        /** Codec pointer */
-        int _audioCodec;
-
-        /**
-         * Format currently in use in the conversation,
-         * sent in each outgoing voice packet.
-         */
-        int _format;
+        int format;
+        iax_session* session;
 };
 
 #endif

daemon/src/iax/iaxvoiplink.h

@@ -36,12 +36,12 @@
 #include <iax-client.h>
 #include "audio/codecs/audiocodec.h" // for DEC_BUFFER_SIZE
 #include "global.h"
+#include "audio/samplerateconverter.h"
 
 namespace sfl {
     class InstantMessaging;
 }
 
-class SamplerateConverter;
 class EventThread;
 class IAXCall;
 
@@ -164,10 +164,8 @@ class IAXVoIPLink : public VoIPLink
         /**
          * Refuse a call
          * @param id The ID of the call
-         * @return bool true on success
-         *		  false otherwise
          */
-        virtual bool refuse (const std::string& id);
+        virtual void refuse (const std::string& id);
 
         /**
          * Send DTMF
@@ -185,11 +183,6 @@ class IAXVoIPLink : public VoIPLink
          */
         virtual std::string getCurrentCodecName(Call *c) const;
 
-
-    public: // iaxvoiplink only
-
-        void updateAudiolayer (void);
-
     private:
         /*
          * Decode the message count IAX send.
@@ -208,11 +201,6 @@ class IAXVoIPLink : public VoIPLink
          */
         IAXCall* getIAXCall (const std::string& id);
 
-        /**
-         * Delete every call
-         */
-        void terminateIAXCall();
-
         /**
          * Find a iaxcall by iax session number
          * @param session an iax_session valid pointer
@@ -255,7 +243,7 @@ class IAXVoIPLink : public VoIPLink
          * Send an outgoing call invite to iax
          * @param call An IAXCall pointer
          */
-        bool iaxOutgoingInvite (IAXCall* call);
+        void iaxOutgoingInvite (IAXCall* call);
 
         /** Threading object */
         EventThread* evThread_;
@@ -272,17 +260,13 @@ class IAXVoIPLink : public VoIPLink
          * iax_stuff inside this class. */
         ost::Mutex mutexIAX_;
 
-        /** Connection to audio card/device */
-        AudioLayer* audiolayer_;
-
         /** encoder/decoder/resampler buffers */
         SFLDataFormat decData[DEC_BUFFER_SIZE];
         SFLDataFormat resampledData[DEC_BUFFER_SIZE];
         unsigned char encodedData[DEC_BUFFER_SIZE];
 
-        int converterSamplingRate_;
         /** Sample rate converter object */
-        SamplerateConverter* converter_;
+        SamplerateConverter converter_;
 
         /** Whether init() was called already or not
          * This should be used in init() and terminate(), to

daemon/src/managerimpl.cpp

@@ -585,10 +585,8 @@ bool ManagerImpl::attendedTransfer(const std::string& transferID, const std::str
 }
 
 //THREAD=Main : Call:Incoming
-bool ManagerImpl::refuseCall (const std::string& id)
+void ManagerImpl::refuseCall (const std::string& id)
 {
-    bool returnValue;
-
     stopTone();
 
     if (getCallList().size() <= 1) {
@@ -600,31 +598,25 @@ bool ManagerImpl::refuseCall (const std::string& id)
     /* Direct IP to IP call */
 
     if (getConfigFromCall (id) == Call::IPtoIP)
-        returnValue = SIPVoIPLink::instance ()-> refuse (id);
+        SIPVoIPLink::instance()->refuse (id);
     else {
         /* Classic call, attached to an account */
         std::string accountid = getAccountFromCall (id);
         if (accountid.empty())
-            return false;
+            return;
 
-        returnValue = getAccountLink (accountid)->refuse (id);
+        getAccountLink(accountid)->refuse(id);
 
         removeCallAccount (id);
     }
 
-    // if the call was outgoing or established, we didn't refuse it
-    // so the method did nothing
-    if (returnValue) {
-        removeWaitingCall (id);
-        _dbus.getCallManager()->callStateChanged (id, "HUNGUP");
-    }
+	removeWaitingCall(id);
+	_dbus.getCallManager()->callStateChanged(id, "HUNGUP");
 
     // Disconnect streams
     removeStream(id);
 
     getMainBuffer()->stateInfo();
-
-    return returnValue;
 }
 
 Conference*
@@ -1361,8 +1353,6 @@ bool ManagerImpl::incomingCallWaiting ()
 
 void ManagerImpl::addWaitingCall (const std::string& id)
 {
-    _info ("Manager: Add waiting call %s (%d calls)", id.c_str(), _nbIncomingWaitingCall);
-
     ost::MutexLock m(_waitingCallMutex);
     _waitingCall.insert (id);
     _nbIncomingWaitingCall++;
@@ -1370,12 +1360,8 @@ void ManagerImpl::addWaitingCall (const std::string& id)
 
 void ManagerImpl::removeWaitingCall (const std::string& id)
 {
-    _info ("Manager: Remove waiting call %s (%d calls)", id.c_str(), _nbIncomingWaitingCall);
-
     ost::MutexLock m(_waitingCallMutex);
-    // should return more than 1 if it erase a call
-
-    if (_waitingCall.erase (id))
+    if (_waitingCall.erase(id))
         _nbIncomingWaitingCall--;
 }
 
@@ -1388,17 +1374,14 @@ bool ManagerImpl::isWaitingCall (const std::string& id)
 // Management of event peer IP-phone
 ////////////////////////////////////////////////////////////////////////////////
 // SipEvent Thread
-bool ManagerImpl::incomingCall (Call* call, const std::string& accountId)
+void ManagerImpl::incomingCall (Call* call, const std::string& accountId)
 {
     assert(call);
     stopTone();
 
-    _debug ("Manager: Incoming call %s for account %s", call->getCallId().data(), accountId.c_str());
-
     associateCallToAccount (call->getCallId(), accountId);
 
-    // If account is null it is an ip to ip call
-    if (accountId.empty())
+    if (accountId == "")
         associateConfigToCall (call->getCallId(), Call::IPtoIP);
     else {
         // strip sip: which is not required and bring confusion with ip to ip calls
@@ -1407,45 +1390,26 @@ bool ManagerImpl::incomingCall (Call* call, const std::string& accountId)
 
         size_t startIndex = peerNumber.find ("sip:");
 
-        if (startIndex != std::string::npos) {
-            std::string strippedPeerNumber = peerNumber.substr (startIndex + 4);
-            call->setPeerNumber (strippedPeerNumber);
-        }
+        if (startIndex != std::string::npos)
+            call->setPeerNumber (peerNumber.substr (startIndex + 4));
     }
 
     if (!hasCurrentCall()) {
-        _debug ("Manager: Has no current call start ringing");
         call->setConnectionState (Call::Ringing);
         ringtone (accountId);
-
-    } else
-        _debug ("Manager: has current call, beep in current audio stream");
+    }
 
     addWaitingCall (call->getCallId());
 
     std::string from(call->getPeerName());
     std::string number(call->getPeerNumber());
-    std::string display_name(call->getDisplayName());
 
-    if (not from.empty() and not number.empty()) {
-        from.append (" <");
-        from.append (number);
-        from.append (">");
-    } else if (from.empty()) {
-        from.append ("<");
-        from.append (number);
-        from.append (">");
-    }
-
-    /* Broadcast a signal over DBus */
-    _debug ("Manager: From: %s, Number: %s, Display Name: %s", from.c_str(), number.c_str(), display_name.c_str());
+    if (not from.empty() and not number.empty())
+        from += " ";
 
-    std::string display(display_name);
-    display.append(" ");
-    display.append(from);
+    from += "<" + number + ">";
 
-    _dbus.getCallManager()->incomingCall(accountId, call->getCallId(), display);
-    return true;
+    _dbus.getCallManager()->incomingCall(accountId, call->getCallId(), call->getDisplayName() + " " + from);
 }
 
 

daemon/src/managerimpl.h

@@ -231,7 +231,7 @@ class ManagerImpl
          * Refuse the call
          * @param id  The call identifier
          */
-        bool refuseCall (const std::string& id);
+        void refuseCall (const std::string& id);
 
         /**
          * Create a new conference given two participant
@@ -365,9 +365,8 @@ class ManagerImpl
          * and notify user
          * @param call A call pointer
          * @param accountId an account id
-         * @return bool True if the call was added correctly
          */
-        bool incomingCall (Call* call, const std::string& accountId);
+        void incomingCall (Call* call, const std::string& accountId);
 
         /**
          * Notify the user that the recipient of the call has answered and the put the

daemon/src/sip/sipvoiplink.cpp

@@ -667,28 +667,26 @@ bool SIPVoIPLink::attendedTransfer(const std::string& id, const std::string& to)
     return transferCommon(getSIPCall(id), &dst);
 }
 
-bool
+void
 SIPVoIPLink::refuse (const std::string& id)
 {
     SIPCall *call = getSIPCall (id);
     if (!call->isIncoming() or call->getConnectionState() == Call::Connected)
-        return false;
+        return;
 
     call->getAudioRtp()->stop();
 
     pjsip_tx_data *tdata;
     if (pjsip_inv_end_session (call->inv, PJSIP_SC_DECLINE, NULL, &tdata) != PJ_SUCCESS)
-        return false;
+        return;
 
     if (pjsip_inv_send_msg (call->inv, tdata) != PJ_SUCCESS)
-        return false;
+        return;
 
     // Make sure the pointer is NULL in callbacks
     call->inv->mod_data[_mod_ua.id] = NULL;
 
     removeCall (id);
-
-    return true;
 }
 
 std::string
@@ -1730,8 +1728,12 @@ static pj_bool_t transaction_request_cb (pjsip_rx_data *rdata)
 	call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
 
     pjsip_dialog* dialog;
-    if (pjsip_dlg_create_uas (pjsip_ua_instance(), rdata, NULL, &dialog) != PJ_SUCCESS)
-        goto fail;
+    if (pjsip_dlg_create_uas (pjsip_ua_instance(), rdata, NULL, &dialog) != PJ_SUCCESS) {
+		delete call;
+		pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL, NULL);
+		return false;
+
+    }
 
     pjsip_inv_create_uas (dialog, rdata, call->getLocalSDP()->getLocalSdpSession(), 0, &call->inv);
 
@@ -1767,18 +1769,11 @@ static pj_bool_t transaction_request_cb (pjsip_rx_data *rdata)
 
     	call->setConnectionState (Call::Ringing);
 
-    	if (!Manager::instance().incomingCall (call, account_id))
-    		goto fail;
-
+    	Manager::instance().incomingCall (call, account_id);
 		Manager::instance().getAccountLink (account_id)->addCall (call);
     }
 
     return true;
-
-fail:
-	delete call;
-	pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL, NULL);
-	return false;
 }
 
 static pj_bool_t transaction_response_cb (pjsip_rx_data *rdata)

daemon/src/sip/sipvoiplink.h

@@ -164,9 +164,8 @@ class SIPVoIPLink : public VoIPLink
         /**
          * Refuse the call
          * @param id The call identifier
-         * @return bool True on success
          */
-        virtual bool refuse (const std::string& id);
+        virtual void refuse (const std::string& id);
 
         /**
          * Send DTMF refering to account configuration

daemon/src/voiplink.h

@@ -148,9 +148,8 @@ class VoIPLink
         /**
          * Refuse incoming call
          * @param id The call identifier
-         * @return bool True on success
          */
-        virtual bool refuse (const std::string& id) = 0;
+        virtual void refuse (const std::string& id) = 0;
 
         /**
          * Send DTMF