diff --git a/sflphone-common/src/audio/alsa/alsalayer.cpp b/sflphone-common/src/audio/alsa/alsalayer.cpp
index c13d56450e42fe44aebe23dc3eea51688d58e0ec..4261eb4896e793f6e50b215e233b7eb5b846af78 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.cpp
+++ b/sflphone-common/src/audio/alsa/alsalayer.cpp
@@ -1038,7 +1038,7 @@ void AlsaLayer::audioCallback (void)
_audiofilter->processAudio (rsmpl_out, nbSample*sizeof (SFLDataFormat));
- getMainBuffer()->putData (rsmpl_out, nbSample * sizeof (SFLDataFormat), 100);
+ getMainBuffer()->putData (rsmpl_out, nbSample * sizeof (SFLDataFormat));
free (rsmpl_out);
} else {
@@ -1047,7 +1047,7 @@ void AlsaLayer::audioCallback (void)
if (filter_out) {
_audiofilter->processAudio (in, filter_out, toPut);
// captureFile->write ( (const char *) filter_out, toPut);
- getMainBuffer()->putData (filter_out, toPut, 100);
+ getMainBuffer()->putData (filter_out, toPut);
free (filter_out);
}
}
diff --git a/sflphone-common/src/audio/audiolayer.cpp b/sflphone-common/src/audio/audiolayer.cpp
index f7eb216a8ff1485d78871658afde44db444b8524..1615d97c71317f10ba23ef9eac06cedcbbe9baf0 100644
--- a/sflphone-common/src/audio/audiolayer.cpp
+++ b/sflphone-common/src/audio/audiolayer.cpp
@@ -56,9 +56,9 @@ int AudioLayer::putUrgent (void* buffer, int toCopy)
a = _urgentRingBuffer.AvailForPut();
if (a >= toCopy) {
- return _urgentRingBuffer.Put (buffer, toCopy, _defaultVolume);
+ return _urgentRingBuffer.Put (buffer, toCopy);
} else {
- return _urgentRingBuffer.Put (buffer, a, _defaultVolume);
+ return _urgentRingBuffer.Put (buffer, a);
}
return 0;
@@ -72,10 +72,10 @@ int AudioLayer::putMain (void *buffer, int toCopy, std::string call_id)
a = getMainBuffer()->availForPut (call_id);
if (a >= toCopy) {
- return getMainBuffer()->putData (buffer, toCopy, _defaultVolume, call_id);
+ return getMainBuffer()->putData (buffer, toCopy, call_id);
} else {
_debug ("Chopping sound, Ouch! RingBuffer full ?");
- return getMainBuffer()->putData (buffer, a, _defaultVolume, call_id);
+ return getMainBuffer()->putData (buffer, a, call_id);
}
return 0;
diff --git a/sflphone-common/src/audio/audiolayer.h b/sflphone-common/src/audio/audiolayer.h
index 77e72610565cd7eac72b36cd69eb730c70aaf702..89ea65bb9be66a8e6d4c37d9278e0b66afc068b3 100644
--- a/sflphone-common/src/audio/audiolayer.h
+++ b/sflphone-common/src/audio/audiolayer.h
@@ -70,8 +70,7 @@ class AudioLayer
* @param manager An instance of managerimpl
*/
AudioLayer (ManagerImpl* manager , int type)
- : _defaultVolume (100)
- , _layerType (type)
+ : _layerType (type)
, _isStarted(false)
, _manager (manager)
, _urgentRingBuffer (SIZEBUF, default_id)
@@ -245,12 +244,6 @@ class AudioLayer
_mainBuffer = mainbuffer;
}
- /**
- * Default volume for incoming RTP and Urgent sounds.
- */
- unsigned short _defaultVolume; // 100
-
-
/**
* Set the audio recorder
*/
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
index 822b10492036b4ec3e344a6d36cf5638f9e83d2d..6cc6862971f6d76f30d811ffcafa9304fc16c578 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
@@ -260,7 +260,6 @@ int AudioRtpRecordHandler::processDataEncode (void)
{
SFLDataFormat *micData = _audioRtpRecord._micData;
unsigned char *micDataEncoded = _audioRtpRecord._micDataEncoded;
- //SFLDataFormat *micDataEchoCancelled = _audioRtpRecord._micDataEchoCancelled;
SFLDataFormat *micDataConverted = _audioRtpRecord._micDataConverted;
int codecFrameSize = getCodecFrameSize();
@@ -303,11 +302,9 @@ int AudioRtpRecordHandler::processDataEncode (void)
_audioRtpRecord.audioProcessMutex.enter();
if (Manager::instance().audioPreference.getNoiseReduce()) {
- // _audioRtpRecord._audioProcess->processAudio (micDataEchoCancelled, nbSample * sizeof (SFLDataFormat));
_audioRtpRecord._audioProcess->processAudio (micDataConverted, nbSample * sizeof (SFLDataFormat));
}
- // echoCanceller.process(micDataConverted, micDataEchoCancelled, nbSample * sizeof(SFLDataFormat));
if(Manager::instance().getEchoCancelState() == "enabled") {
echoCanceller.getData(micData);
}
@@ -317,7 +314,6 @@ int AudioRtpRecordHandler::processDataEncode (void)
_audioRtpRecord.audioCodecMutex.enter();
compSize = _audioRtpRecord._audioCodec->encode (micDataEncoded, micData, nbSample * sizeof (SFLDataFormat));
- // compSize = _audioRtpRecord._audioCodec->encode (micDataEncoded, micDataConverted, nbSample * sizeof (SFLDataFormat));
_audioRtpRecord.audioCodecMutex.leave();
@@ -327,11 +323,8 @@ int AudioRtpRecordHandler::processDataEncode (void)
if (Manager::instance().audioPreference.getNoiseReduce()) {
_audioRtpRecord._audioProcess->processAudio (micData, nbSample * sizeof (SFLDataFormat));
- // _audioRtpRecord._audioProcess->processAudio (micDataEchoCancelled, nbSample * sizeof (SFLDataFormat));
}
- // echoCanceller.process(micData, micDataEchoCancelled, nbSample * sizeof(SFLDataFormat));
- // echoCanceller.process(micData, micDataEchoCancelled, nbSample * sizeof(SFLDataFormat));
if(Manager::instance().getEchoCancelState() == "enabled") {
echoCanceller.getData(micData);
}
@@ -346,7 +339,6 @@ int AudioRtpRecordHandler::processDataEncode (void)
_audioRtpRecord.audioCodecMutex.enter();
// no resampling required
- // compSize = _audioRtpRecord._audioCodec->encode (micDataEncoded, micData, nbSample * sizeof (SFLDataFormat));
compSize = _audioRtpRecord._audioCodec->encode (micDataEncoded, micData, nbSample * sizeof (SFLDataFormat));
_audioRtpRecord.audioCodecMutex.leave();
@@ -383,17 +375,15 @@ void AudioRtpRecordHandler::processDataDecode (unsigned char *spkrData, unsigned
// test if resampling is required
if (codecSampleRate != mainBufferSampleRate) {
-
// Do sample rate conversion
- int nbSampleDown = nbSample;
- nbSample = _audioRtpRecord._converter->upsampleData (spkrDataDecoded, spkrDataConverted, codecSampleRate, mainBufferSampleRate, nbSampleDown);
+ nbSample = _audioRtpRecord._converter->upsampleData (spkrDataDecoded, spkrDataConverted, codecSampleRate, mainBufferSampleRate, nbSample);
if(Manager::instance().getEchoCancelState() == "enabled") {
echoCanceller.putData(spkrDataConverted, nbSample * sizeof(SFLDataFormat));
}
// put data in audio layer, size in byte
- Manager::instance().getMainBuffer()->putData (spkrDataConverted, nbSample * sizeof (SFLDataFormat), 100, id_);
+ Manager::instance().getMainBuffer()->putData (spkrDataConverted, nbSample * sizeof (SFLDataFormat), id_);
} else {
@@ -401,28 +391,22 @@ void AudioRtpRecordHandler::processDataDecode (unsigned char *spkrData, unsigned
echoCanceller.putData(spkrDataDecoded, expandedSize);
}
// put data in audio layer, size in byte
- Manager::instance().getMainBuffer()->putData (spkrDataDecoded, expandedSize, 100, id_);
+ Manager::instance().getMainBuffer()->putData (spkrDataDecoded, expandedSize, id_);
}
}
bool AudioRtpRecordHandler::fadeIn (SFLDataFormat *audio, int size, SFLDataFormat *factor)
{
-
// if factor reach 0, this function should no be called anymore
if (*factor <= 0)
return true;
- // apply amplitude factor;
- while (size) {
- size--;
- audio[size] /= *factor;
- }
+ while (size)
+ audio[--size] /= *factor;
- // decrease factor
*factor /= FADEIN_STEP_SIZE;
- return false;
-
+ return *factor <= 0;
}
}
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
index 7a19a6df813c1abb9771b187a63a7b3f2d981a17..9bd49b20b1ceb3515eef8ab77075030e0a5b5b3d 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
@@ -181,7 +181,6 @@ void AudioRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
dtmf->payload.duration++;
-
// next packet is going to be the last one
if ( (dtmf->length - increment) < increment)
dtmf->payload.ebit = true;
@@ -195,26 +194,16 @@ void AudioRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
void AudioRtpSession::receiveSpeakerData ()
{
- const ost::AppDataUnit* adu = NULL;
-
- int packetTimestamp = _queue->getFirstTimestamp();
-
- adu = _queue->getData (packetTimestamp);
-
- if (!adu) {
+ const ost::AppDataUnit* adu = _queue->getData (_queue->getFirstTimestamp());
+ if (!adu)
return;
- }
-
- unsigned char* spkrDataIn = NULL;
- unsigned int size = 0;
- spkrDataIn = (unsigned char*) adu->getData(); // data in char
- size = adu->getSize(); // size in char
+ unsigned char* spkrDataIn = (unsigned char*) adu->getData(); // data in char
+ unsigned int size = adu->getSize(); // size in char
// DTMF over RTP, size must be over 4 in order to process it as voice data
- if (size > 4) {
+ if (size > 4)
processDataDecode (spkrDataIn, size);
- }
delete adu;
}
@@ -327,7 +316,6 @@ void AudioRtpSession::stopRtpThread ()
bool AudioRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&)
{
receiveSpeakerData();
-
return true;
}
diff --git a/sflphone-common/src/audio/mainbuffer.cpp b/sflphone-common/src/audio/mainbuffer.cpp
index 0232fbad4c0d6dfb6cb3b9cf3489e9cebda93e1b..5514c28ffa3a49553880f36cfafe89ee83b49861 100644
--- a/sflphone-common/src/audio/mainbuffer.cpp
+++ b/sflphone-common/src/audio/mainbuffer.cpp
@@ -35,81 +35,49 @@
MainBuffer::MainBuffer() : _internalSamplingRate (8000)
{
- mixBuffer = new SFLDataFormat[STATIC_BUFSIZE];
}
MainBuffer::~MainBuffer()
{
-
- delete [] mixBuffer;
- mixBuffer = NULL;
}
void MainBuffer::setInternalSamplingRate (int sr)
{
- _debug("MainBuffer: Set internal sampling rate");
-
if (sr > _internalSamplingRate) {
-
- _debug ("MainBuffer: Internal sampling rate changed %d", sr);
-
- // This call takes the mutex
flushAllBuffers();
-
- // ost::MutexLock guard (_mutex);
-
_internalSamplingRate = sr;
-
- // Manager::instance().audioSamplingRateChanged(sr);
-
}
}
CallIDSet* MainBuffer::getCallIDSet (std::string call_id)
{
-
CallIDMap::iterator iter = _callIDMap.find (call_id);
-
- if (iter != _callIDMap.end())
- return iter->second;
- else
- return NULL;
-
+ return (iter != _callIDMap.end()) ? iter->second : NULL;
}
-bool MainBuffer::createCallIDSet (std::string set_id)
+void MainBuffer::createCallIDSet (std::string set_id)
{
-
- CallIDSet* newCallIDSet = new CallIDSet;
-
- _callIDMap.insert (std::pair<std::string, CallIDSet*> (set_id, newCallIDSet));
-
- return true;
-
+ _callIDMap.insert (std::pair<std::string, CallIDSet*> (set_id, new CallIDSet));
}
bool MainBuffer::removeCallIDSet (std::string set_id)
{
-
-
CallIDSet* callid_set = getCallIDSet (set_id);
- if (callid_set != NULL) {
- if (_callIDMap.erase (set_id) != 0) {
- delete callid_set;
- callid_set = NULL;
- return true;
- } else {
- _debug ("removeCallIDSet error while removing callid set %s!", set_id.c_str());
- return false;
- }
- } else {
+ if (!callid_set) {
_debug ("removeCallIDSet error callid set %s does not exist!", set_id.c_str());
return false;
}
+ if (_callIDMap.erase (set_id) == 0) {
+ _debug ("removeCallIDSet error while removing callid set %s!", set_id.c_str());
+ return false;
+ }
+ delete callid_set;
+ callid_set = NULL;
+ return true;
}
void MainBuffer::addCallIDtoSet (std::string set_id, std::string call_id)
@@ -148,9 +116,7 @@ RingBuffer* MainBuffer::getRingBuffer (std::string call_id)
RingBuffer* MainBuffer::createRingBuffer (std::string call_id)
{
RingBuffer* newRingBuffer = new RingBuffer (SIZEBUF, call_id);
-
_ringBufferMap.insert (std::pair<std::string, RingBuffer*> (call_id, newRingBuffer));
-
return newRingBuffer;
}
@@ -332,7 +298,7 @@ void MainBuffer::unBindAllHalfDuplexOut (std::string process_id)
}
-int MainBuffer::putData (void *buffer, int toCopy, unsigned short volume, std::string call_id)
+int MainBuffer::putData (void *buffer, int toCopy, std::string call_id)
{
ost::MutexLock guard (_mutex);
@@ -348,11 +314,11 @@ int MainBuffer::putData (void *buffer, int toCopy, unsigned short volume, std::s
if (a >= toCopy) {
- return ring_buffer->Put (buffer, toCopy, volume);
+ return ring_buffer->Put (buffer, toCopy);
} else {
- return ring_buffer->Put (buffer, a, volume);
+ return ring_buffer->Put (buffer, a);
}
}
@@ -378,12 +344,7 @@ int MainBuffer::getData (void *buffer, int toCopy, unsigned short volume, std::s
CallIDSet* callid_set = getCallIDSet (call_id);
- int nbSmplToCopy = toCopy / sizeof (SFLDataFormat);
-
- if (callid_set == NULL)
- return 0;
-
- if (callid_set->empty()) {
+ if (!callid_set || callid_set->empty()) {
return 0;
}
@@ -396,18 +357,17 @@ int MainBuffer::getData (void *buffer, int toCopy, unsigned short volume, std::s
} else
return 0;
} else {
-
- memset (buffer, 0, nbSmplToCopy*sizeof (SFLDataFormat));
+ memset (buffer, 0, toCopy);
int size = 0;
CallIDSet::iterator iter_id = callid_set->begin();
while (iter_id != callid_set->end()) {
-
+ int nbSmplToCopy = toCopy / sizeof (SFLDataFormat);
+ SFLDataFormat mixBuffer[nbSmplToCopy];
memset (mixBuffer, 0, toCopy);
-
- size = getDataByID (mixBuffer, toCopy, volume, (std::string) (*iter_id), call_id);
+ size = getDataByID (mixBuffer, toCopy, volume, *iter_id, call_id);
if (size > 0) {
for (int k = 0; k < nbSmplToCopy; k++) {
diff --git a/sflphone-common/src/audio/mainbuffer.h b/sflphone-common/src/audio/mainbuffer.h
index 973039fc2b51e1398191b858c02351cfefeaca6c..8d74dc3deb532cbf1ec122d46b727f1b958bcc86 100644
--- a/sflphone-common/src/audio/mainbuffer.h
+++ b/sflphone-common/src/audio/mainbuffer.h
@@ -89,7 +89,7 @@ class MainBuffer
void unBindAllHalfDuplexOut (std::string process_id);
- int putData (void *buffer, int toCopy, unsigned short volume = 100, std::string call_id = default_id);
+ int putData (void *buffer, int toCopy, std::string call_id = default_id);
int getData (void *buffer, int toCopy, unsigned short volume = 100, std::string call_id = default_id);
@@ -113,7 +113,7 @@ class MainBuffer
CallIDSet* getCallIDSet (std::string call_id);
- bool createCallIDSet (std::string set_id);
+ void createCallIDSet (std::string set_id);
bool removeCallIDSet (std::string set_id);
@@ -145,8 +145,6 @@ class MainBuffer
CallIDMap _callIDMap;
- SFLDataFormat* mixBuffer;
-
ost::Mutex _mutex;
int _internalSamplingRate;
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index 08c2e6db9803242da2530bb7a8e27071ee485bfc..03e3049d401b7ee886a3d653504a263908446f81 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -889,7 +889,7 @@ void PulseLayer::readFromMic (void)
// remove dc offset
_audiofilter->processAudio (rsmpl_out, nbSample*sizeof (SFLDataFormat));
- getMainBuffer()->putData (rsmpl_out, nbSample*sizeof (SFLDataFormat), 100);
+ getMainBuffer()->putData (rsmpl_out, nbSample*sizeof (SFLDataFormat));
pa_xfree (rsmpl_out);
@@ -902,7 +902,7 @@ void PulseLayer::readFromMic (void)
// remove dc offset
_audiofilter->processAudio ( (SFLDataFormat *) data, filter_out, r);
- getMainBuffer()->putData (filter_out, r, 100);
+ getMainBuffer()->putData (filter_out, r);
pa_xfree (filter_out);
}
diff --git a/sflphone-common/src/audio/ringbuffer.cpp b/sflphone-common/src/audio/ringbuffer.cpp
index 2a1d5fdea84be7577ab91c81af5418c2e22c5e0a..2b15ef80b36f55a5b189e1f215f3ea17cd414348 100644
--- a/sflphone-common/src/audio/ringbuffer.cpp
+++ b/sflphone-common/src/audio/ringbuffer.cpp
@@ -227,9 +227,8 @@ RingBuffer::AvailForPut()
}
// This one puts some data inside the ring buffer.
-// Change the volume if it's not 100
int
-RingBuffer::Put (void* buffer, int toCopy, unsigned short volume)
+RingBuffer::Put (void* buffer, int toCopy)
{
samplePtr src;
int block;
@@ -258,18 +257,6 @@ RingBuffer::Put (void* buffer, int toCopy, unsigned short volume)
block = mBufferSize - pos;
}
- // Gain adjustment (when Mic vol. is changed)
- if (volume != 100) {
- SFLDataFormat* start = (SFLDataFormat*) src;
- int nbSample = block / sizeof (SFLDataFormat);
-
- for (int i=0; i<nbSample; i++) {
- start[i] = start[i] * volume / 100;
- }
- }
-
- // bcopy(src, dest, len)
- //fprintf(stderr, "has %d put %d\t", len, block);
bcopy (src, mBuffer + pos, block);
src += block;
diff --git a/sflphone-common/src/audio/ringbuffer.h b/sflphone-common/src/audio/ringbuffer.h
index d931470eaa634418f18b0f3f43fb201632b4f1c6..8b84e94418212005a45c590095956f05c9927bac 100644
--- a/sflphone-common/src/audio/ringbuffer.h
+++ b/sflphone-common/src/audio/ringbuffer.h
@@ -107,10 +107,9 @@ class RingBuffer
* Write data in the ring buffer
* @param buffer Data to copied
* @param toCopy Number of bytes to copy
- * @param volume The volume
* @return int Number of bytes copied
*/
- int Put (void* buffer, int toCopy, unsigned short volume = 100);
+ int Put (void* buffer, int toCopy);
/**
* To get how much space is available in the buffer to read in
diff --git a/sflphone-common/src/global.h b/sflphone-common/src/global.h
index d244ece14fff906ff1b7111b5f1de6b4a459c871..504713b89ca223616f3b0189973c11a268fdd91d 100644
--- a/sflphone-common/src/global.h
+++ b/sflphone-common/src/global.h
@@ -83,7 +83,6 @@ static const SOUND_FORMAT INT32 = 0x8;
#define CODECDIR "codecs" /** Codecs directory */
#define SIZEBUF 400000 /** About 12 sec of buffering at 8000 Hz*/
-#define STATIC_BUFSIZE 5000
#define ALSA_DFT_CARD_ID 0 /** Index of the default soundcard */
diff --git a/sflphone-common/src/iax/iaxvoiplink.cpp b/sflphone-common/src/iax/iaxvoiplink.cpp
index 4e38d0ce35f0d279a623477b7380a91a7cfa1c6b..5a23dfdd35df029dc3f9e370e2973f299e9147d9 100644
--- a/sflphone-common/src/iax/iaxvoiplink.cpp
+++ b/sflphone-common/src/iax/iaxvoiplink.cpp
@@ -95,46 +95,18 @@ IAXVoIPLink::IAXVoIPLink (const std::string& accountID) : VoIPLink ()
IAXVoIPLink::~IAXVoIPLink()
{
- if (_evThread) {
- delete _evThread;
- _evThread = NULL;
- }
+ delete _evThread;
+ _evThread = NULL;
_regSession = NULL; // shall not delete it
terminate();
- audiolayer = NULL;
-
- if (converter) {
- delete converter;
- converter = NULL;
- }
-
- if (micData) {
- delete [] micData;
- micData = NULL;
- }
-
- if (micDataConverted) {
- delete [] micDataConverted;
- micDataConverted = NULL;
- }
-
- if (micDataEncoded) {
- delete [] micDataEncoded;
- micDataEncoded = NULL;
- }
-
- if (spkrDataDecoded) {
- delete [] spkrDataDecoded;
- spkrDataDecoded = NULL;
- }
-
- if (spkrDataConverted) {
- delete [] spkrDataConverted;
- spkrDataConverted = NULL;
- }
-
+ delete converter;
+ delete [] micData;
+ delete [] micDataConverted;
+ delete [] micDataEncoded;
+ delete [] spkrDataDecoded;
+ delete [] spkrDataConverted;
}
bool
@@ -211,21 +183,17 @@ IAXVoIPLink::terminate()
void
IAXVoIPLink::terminateIAXCall()
{
- std::string reason = "Dumped Call";
ost::MutexLock m (_callMapMutex);
CallMap::iterator iter = _callMap.begin();
- IAXCall *call;
while (iter != _callMap.end()) {
- call = dynamic_cast<IAXCall*> (iter->second);
-
+ IAXCall *call = dynamic_cast<IAXCall*> (iter->second);
if (call) {
_mutexIAX.enterMutex();
- iax_hangup (call->getSession(), (char*) reason.c_str());
+ iax_hangup (call->getSession(), (char*) "Dumped Call");
_mutexIAX.leaveMutex();
call->setSession (NULL);
delete call;
- call = NULL;
}
iter++;
@@ -307,11 +275,6 @@ IAXVoIPLink::getEvent()
void
IAXVoIPLink::sendAudioFromMic (void)
{
-
- int maxBytesToGet, availBytesFromMic, bytesAvail, compSize;
- sfl::AudioCodec *audioCodec = NULL;
- IAXCall *currentCall;
-
// We have to update the audio layer type in case we switched
// TODO Find out a better way to do it
updateAudiolayer();
@@ -322,89 +285,53 @@ IAXVoIPLink::sendAudioFromMic (void)
CallMap::iterator iter_call = _callMap.begin();
while (iter_call != _callMap.end()) {
-
- currentCall = (IAXCall*) iter_call->second;
-
+ IAXCall *currentCall = (IAXCall*) iter_call->second;
iter_call++;
- if (currentCall) {
-
- // bool sessionIsConnected = (currentCall->getConnectionState() == Call::Connected);
- bool callIsActive = (currentCall->getState() == Call::Active);
-
- // if (sessionIsConnected || callIsActive) {
-
- if (callIsActive) {
+ if (!currentCall || currentCall->getState() != Call::Active)
+ continue;
AudioCodecType codecType = currentCall->getAudioCodec();
- audioCodec = static_cast<sfl::AudioCodec *>(currentCall->getAudioCodecFactory().getCodec (codecType));
-
- // Send sound here
-
- if (audioCodec && audiolayer) {
-
- // _debug("Send sound");
- // audiolayer->getMainBuffer()->flush(currentCall->getCallId());
-
- Manager::instance().getMainBuffer()->setInternalSamplingRate (audioCodec->getClockRate());
-
- int _mainBufferSampleRate = audiolayer->getMainBuffer()->getInternalSamplingRate();
-
- // we have to get 20ms of data from the mic *20/1000 = /50
- // rate/50 shall be lower than IAX__20S_48KHZ_MAX
- maxBytesToGet = _mainBufferSampleRate * audiolayer->getFrameSize() / 1000 * sizeof (SFLDataFormat);
-
- // available bytes inside ringbuffer
- availBytesFromMic = audiolayer->getMainBuffer()->availForGet (currentCall->getCallId());
- // We need packets full!
-
- if (availBytesFromMic > maxBytesToGet) {
+ sfl::AudioCodec *audioCodec = static_cast<sfl::AudioCodec *>(currentCall->getAudioCodecFactory().getCodec (codecType));
- // take the lowest
- bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
+ // Send sound here
- // Get bytes from micRingBuffer to data_from_mic
- nbSample_ = audiolayer->getMainBuffer()->getData (micData, bytesAvail, 100, currentCall->getCallId()) / sizeof (SFLDataFormat);
+ if (!audioCodec || !audiolayer)
+ continue;
+ Manager::instance().getMainBuffer()->setInternalSamplingRate (audioCodec->getClockRate());
- // Store the number of samples for recording
- nbSampleForRec_ = nbSample_;
+ int _mainBufferSampleRate = audiolayer->getMainBuffer()->getInternalSamplingRate();
+ // we have to get 20ms of data from the mic *20/1000 = /50
+ // rate/50 shall be lower than IAX__20S_48KHZ_MAX
+ int needed = _mainBufferSampleRate * audiolayer->getFrameSize() / 1000 * sizeof (SFLDataFormat);
+ if (audiolayer->getMainBuffer()->availForGet (currentCall->getCallId()) < needed)
+ continue;
- if (audioCodec->getClockRate() && ((int) audioCodec->getClockRate() != _mainBufferSampleRate)) {
+ // Get bytes from micRingBuffer to data_from_mic
+ int bytes = audiolayer->getMainBuffer()->getData (micData, needed, 100, currentCall->getCallId()) / sizeof (SFLDataFormat);
- // resample
- nbSample_ = converter->downsampleData (micData , micDataConverted , (int) audioCodec->getClockRate(), _mainBufferSampleRate, nbSample_);
+ int compSize;
+ if (audioCodec->getClockRate() && ((int) audioCodec->getClockRate() != _mainBufferSampleRate)) {
+ // resample
+ bytes = converter->downsampleData (micData , micDataConverted , (int) audioCodec->getClockRate(), _mainBufferSampleRate, bytes);
+ compSize = audioCodec->encode (micDataEncoded, micDataConverted , bytes);
+ } else {
+ compSize = audioCodec->encode (micDataEncoded, micData, bytes);
+ }
- // for the mono: range = 0 to IAX_FRAME2SEND * sizeof(int16)
- compSize = audioCodec->encode (micDataEncoded, micDataConverted , nbSample_*sizeof (int16));
+ // Send it out!
+ _mutexIAX.enterMutex();
- } else {
-
- // for the mono: range = 0 to IAX_FRAME2SEND * sizeof(int16)
- compSize = audioCodec->encode (micDataEncoded, micData, nbSample_*sizeof (int16));
-
- }
-
- // Send it out!
- _mutexIAX.enterMutex();
-
- // Make sure the session and the call still exists.
- if (currentCall->getSession() && (micDataEncoded != NULL) && (nbSample_ > 0)) {
- if (iax_send_voice (currentCall->getSession(), currentCall->getFormat(), micDataEncoded, compSize, nbSample_) == -1) {
- _debug ("IAX: Error sending voice data.");
- }
- }
-
- _mutexIAX.leaveMutex();
-
- }
- }
-
- }
-
- }
+ // Make sure the session and the call still exists.
+ if (currentCall->getSession() && (micDataEncoded != NULL) && (bytes > 0)) {
+ if (iax_send_voice (currentCall->getSession(), currentCall->getFormat(), micDataEncoded, compSize, bytes) == -1) {
+ _debug ("IAX: Error sending voice data.");
+ }
+ }
+ _mutexIAX.leaveMutex();
}
}
@@ -1049,12 +976,12 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
nbInt16 = converter->upsampleData (spkrDataDecoded, spkrDataConverted, audioCodec->getClockRate(), _mainBufferSampleRate, nbSample_);
/* Write the data to the mic ring buffer */
- audiolayer->getMainBuffer()->putData (spkrDataConverted, nbInt16 * sizeof (SFLDataFormat), 100, call->getCallId());
+ audiolayer->getMainBuffer()->putData (spkrDataConverted, nbInt16 * sizeof (SFLDataFormat), call->getCallId());
} else {
/* Write the data to the mic ring buffer */
- audiolayer->getMainBuffer()->putData (spkrDataDecoded, nbInt16 * sizeof (SFLDataFormat), 100, call->getCallId());
+ audiolayer->getMainBuffer()->putData (spkrDataDecoded, nbInt16 * sizeof (SFLDataFormat), call->getCallId());
}
diff --git a/sflphone-common/src/iax/iaxvoiplink.h b/sflphone-common/src/iax/iaxvoiplink.h
index 7ac602372e70520e7cc671d3b80b20fe26b218b4..88da9088d337fe3a74b9d03d42f2e63d0c0639ef 100644
--- a/sflphone-common/src/iax/iaxvoiplink.h
+++ b/sflphone-common/src/iax/iaxvoiplink.h
@@ -320,12 +320,6 @@ class IAXVoIPLink : public VoIPLink
int converterSamplingRate;
- /** number of sample */
- int nbSample_;
-
- /** number of sample before conversion (recording) */
- int nbSampleForRec_;
-
/* URL hook */
UrlHook *urlhook;
diff --git a/sflphone-common/src/sip/sipcall.h b/sflphone-common/src/sip/sipcall.h
index d0bb4f76a241972cfa49b35b579b80eebe0fcd8e..1d9366f95bcfd49287dbee06e283983e2d8eaa23 100644
--- a/sflphone-common/src/sip/sipcall.h
+++ b/sflphone-common/src/sip/sipcall.h
@@ -99,13 +99,6 @@ class SIPCall : public Call
return _local_sdp;
}
- /**
- * Set the local SDP session
- */
- void setLocalSDP (Sdp *local_sdp) {
- _local_sdp = local_sdp;
- }
-
/**
* Returns a pointer to the AudioRtp object
*/
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index 5e5d05ba509d113c93cec02afb2018a8176ab048..8001baf75dcf64d7c11aeca025fdbd00e71b3182 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -3674,13 +3674,9 @@ void registration_cb (struct pjsip_regc_cbparam *param)
const pj_str_t * description = pjsip_get_status_text (param->code);
if (param->code && description) {
-
- //std::string descriptionprint(description->ptr, description->slen);
- //_debug("Received client registration callback wiht code: %i, %s\n", param->code, descriptionprint.c_str());
- DBusManager::instance().getCallManager()->registrationStateChanged (account->getAccountID(), std::string (description->ptr, description->slen), param->code);
- std::pair<int, std::string> details (param->code, std::string (description->ptr, description->slen));
-
-
+ std::string state(description->ptr, description->slen);
+ DBusManager::instance().getCallManager()->registrationStateChanged (account->getAccountID(), state, param->code);
+ std::pair<int, std::string> details (param->code, state);
// TODO: there id a race condition for this ressource when closing the application
account->setRegistrationStateDetailed (details);
}
diff --git a/sflphone-common/test/mainbuffertest.cpp b/sflphone-common/test/mainbuffertest.cpp
index a55f843fb4dae4d6405066ab9ba3ebb330defe16..68c4d087de9d9e40e63c77f89dd9f81cd6bcf2fe 100644
--- a/sflphone-common/test/mainbuffertest.cpp
+++ b/sflphone-common/test/mainbuffertest.cpp
@@ -177,7 +177,7 @@ void MainBufferTest::testCallIDSet()
CPPUNIT_ASSERT (iter_map ==_mainbuffer._callIDMap.end());
// test callidset creation
- CPPUNIT_ASSERT (_mainbuffer.createCallIDSet (test_id) == true);
+ _mainbuffer.createCallIDSet (test_id);
CPPUNIT_ASSERT (_mainbuffer._callIDMap.size() == 1);
iter_map = _mainbuffer._callIDMap.find (test_id);
CPPUNIT_ASSERT (iter_map->first == test_id);
@@ -929,14 +929,14 @@ void MainBufferTest::testGetPutDataByID()
// pu by test_id get by test_id
CPPUNIT_ASSERT (_mainbuffer.availForPut (test_id) == avail_for_put_defaultid);
- CPPUNIT_ASSERT (_mainbuffer.putData (&test_input2, sizeof (int), 100, test_id) == sizeof (int));
+ CPPUNIT_ASSERT (_mainbuffer.putData (&test_input2, sizeof (int), test_id) == sizeof (int));
CPPUNIT_ASSERT (_mainbuffer.availForGetByID (test_id, default_id) == sizeof (int));
CPPUNIT_ASSERT (_mainbuffer.getDataByID (&test_output, sizeof (int), 100, test_id, default_id) == sizeof (int));
CPPUNIT_ASSERT (_mainbuffer.availForGetByID (test_id, default_id) == 0);
CPPUNIT_ASSERT (test_input2 == test_output);
// put/get by false id
- CPPUNIT_ASSERT (_mainbuffer.putData (&test_input2, sizeof (int), 100, false_id) == 0);
+ CPPUNIT_ASSERT (_mainbuffer.putData (&test_input2, sizeof (int), false_id) == 0);
CPPUNIT_ASSERT (_mainbuffer.getDataByID (&test_input2, sizeof (int), 100, false_id, false_id) == 0);
CPPUNIT_ASSERT (_mainbuffer.getDataByID (&test_input2, sizeof (int), 100, default_id, false_id) == 0);
CPPUNIT_ASSERT (_mainbuffer.getDataByID (&test_input2, sizeof (int), 100, false_id, default_id) == 0);
@@ -969,7 +969,7 @@ void MainBufferTest::testGetPutData()
// put by default_id, get by test_id
CPPUNIT_ASSERT (_mainbuffer.availForPut() == avail_for_put_defaultid);
- CPPUNIT_ASSERT (_mainbuffer.putData (&test_input1, sizeof (int), 100) == sizeof (int));
+ CPPUNIT_ASSERT (_mainbuffer.putData (&test_input1, sizeof (int)) == sizeof (int));
CPPUNIT_ASSERT (_mainbuffer.availForPut() == (avail_for_put_defaultid - (int) sizeof (int)));
CPPUNIT_ASSERT (_mainbuffer.availForGet (test_id) == sizeof (int));
CPPUNIT_ASSERT (_mainbuffer.getData (&test_output, sizeof (int), 100, test_id) == sizeof (int));
@@ -983,7 +983,7 @@ void MainBufferTest::testGetPutData()
// put by test_id, get by default_id
CPPUNIT_ASSERT (_mainbuffer.availForPut (test_id) == avail_for_put_testid);
- CPPUNIT_ASSERT (_mainbuffer.putData (&test_input2, sizeof (int), 100, test_id) == sizeof (int));
+ CPPUNIT_ASSERT (_mainbuffer.putData (&test_input2, sizeof (int), test_id) == sizeof (int));
CPPUNIT_ASSERT (_mainbuffer.availForPut (test_id) == (avail_for_put_testid - (int) sizeof (int)));
CPPUNIT_ASSERT (_mainbuffer.availForGet() == sizeof (int));
CPPUNIT_ASSERT (_mainbuffer.getData (&test_output, sizeof (int), 100) == sizeof (int));
@@ -1007,7 +1007,7 @@ void MainBufferTest::testDiscardFlush()
// int test_output_size;
// int init_size;
- CPPUNIT_ASSERT (_mainbuffer.putData (&test_input1, sizeof (int), 100, test_id) == sizeof (int));
+ CPPUNIT_ASSERT (_mainbuffer.putData (&test_input1, sizeof (int), test_id) == sizeof (int));
CPPUNIT_ASSERT (_mainbuffer.availForGet() == sizeof (int));
_mainbuffer.discard (sizeof (int));
CPPUNIT_ASSERT (_mainbuffer.availForGet() == 0);
@@ -1341,7 +1341,7 @@ void MainBufferTest::testConference()
CPPUNIT_ASSERT (_mainbuffer.availForGet (test_id1) == 0);
CPPUNIT_ASSERT (_mainbuffer.availForGet (test_id2) == 0);
// put data test ring buffers
- CPPUNIT_ASSERT (_mainbuffer.putData (&testint, sizeof (int), 100) == sizeof (int));
+ CPPUNIT_ASSERT (_mainbuffer.putData (&testint, sizeof (int)) == sizeof (int));
test_ring_buffer = _mainbuffer.getRingBuffer (default_id);
CPPUNIT_ASSERT (test_ring_buffer->putLen() == sizeof (int));
CPPUNIT_ASSERT (test_ring_buffer->AvailForPut() == (int) (init_put_defaultid - sizeof (int)));