diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
index 30971aaa53ae5b2cfca4e9e73e95895f41c9abfe..b9616466520df4056a6ca410759fb1328b40b0db 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -81,6 +81,7 @@ typedef struct DtmfEvent {
typedef list<DtmfEvent *> EventQueue;
+
template <typename D>
class AudioRtpSession : public ost::Thread, public ost::TimerPort
{
@@ -123,18 +124,51 @@ class AudioRtpSession : public ost::Thread, public ost::TimerPort
private:
+ /**
+ * Allocate memory for RTP buffers and fill them with zeros
+ */
void initBuffers (void);
+ /**
+ * Set RTP Sockets send/receive timeouts
+ */
void setSessionTimeouts (void);
+
+ /**
+ * Set the audio codec for this RTP session
+ */
void setSessionMedia (AudioCodec*);
+
+ /**
+ * Retreive destination address for this session. Stored in CALL
+ */
void setDestinationIpAddress (void);
+ /**
+ * Encode audio data from mainbuffer
+ */
int processDataEncode (void);
+
+ /**
+ * Decode audio data received from peer
+ */
void processDataDecode (unsigned char * spkrData, unsigned int size);
+ /**
+ * Send encoded data to peer
+ */
void sendMicData();
+
+ /**
+ * Receive data from peer
+ */
void receiveSpeakerData ();
+ /**
+ * Ramp In audio data to avoid audio click from peer
+ */
+ bool fadeIn (SFLDataFormat *audio, int size, SFLDataFormat *factor);
+
ost::Time * _time;
// This semaphore is not used
@@ -254,9 +288,31 @@ class AudioRtpSession : public ost::Thread, public ost::TimerPort
*/
int _currentTime;
+ /**
+ * Preprocess internal data
+ */
SpeexPreprocessState *_noiseState;
- // ofstream *captureFile;
+ /**
+ * State of mic fade in
+ */
+ bool _micFadeInComplete;
+
+ /**
+ * State of spkr fade in
+ */
+ bool _spkrFadeInComplete;
+
+ /**
+ * Ampliturde factor to fade in mic data
+ */
+ SFLDataFormat _micAmplFactor;
+
+ /**
+ * Amplitude factor to fade in spkr data
+ */
+ SFLDataFormat _spkrAmplFactor;
+
protected:
@@ -288,6 +344,10 @@ AudioRtpSession<D>::AudioRtpSession (ManagerImpl * manager, SIPCall * sipcall) :
_countNotificationTime (0),
_jbuffer (NULL),
_noiseState (NULL),
+ _micFadeInComplete (false),
+ _spkrFadeInComplete (false),
+ _micAmplFactor (32000),
+ _spkrAmplFactor (32000),
_ca (sipcall)
{
setCancel (cancelDefault);
@@ -316,7 +376,6 @@ AudioRtpSession<D>::AudioRtpSession (ManagerImpl * manager, SIPCall * sipcall) :
_packetLength = 20;
_currentTime = 0;
- // captureFile = new ofstream ("probeCaptureFile", ofstream::binary);
}
template <typename D>
@@ -374,11 +433,6 @@ AudioRtpSession<D>::~AudioRtpSession()
speex_preprocess_state_destroy (_noiseState);
}
-
- // captureFile->close();
-
- // delete captureFile;
-
}
template <typename D>
@@ -624,6 +678,12 @@ int AudioRtpSession<D>::processDataEncode (void)
// Get bytes from micRingBuffer to data_from_mic
int nbSample = _manager->getAudioDriver()->getMainBuffer()->getData (_micData , bytesAvail, 100, _ca->getCallId()) / sizeof (SFLDataFormat);
+ if (!_micFadeInComplete)
+ _micFadeInComplete = fadeIn (_micData, nbSample, &_micAmplFactor);
+
+ if (nbSample == 0)
+ return nbSample;
+
// nb bytes to be sent over RTP
int compSize = 0;
@@ -635,14 +695,15 @@ int AudioRtpSession<D>::processDataEncode (void)
nbSample = _converter->downsampleData (_micData , _micDataConverted , _audiocodec->getClockRate(), _mainBufferSampleRate, nb_sample_up);
- compSize = _audiocodec->codecEncode (_micDataEncoded, _micDataConverted, nbSample*sizeof (int16));
+ compSize = _audiocodec->codecEncode (_micDataEncoded, _micDataConverted, nbSample*sizeof (SFLDataFormat));
} else {
_nSamplesMic = nbSample;
// no resampling required
- compSize = _audiocodec->codecEncode (_micDataEncoded, _micData, nbSample*sizeof (int16));
+ compSize = _audiocodec->codecEncode (_micDataEncoded, _micData, nbSample*sizeof (SFLDataFormat));
+
}
return compSize;
@@ -651,7 +712,6 @@ int AudioRtpSession<D>::processDataEncode (void)
template <typename D>
void AudioRtpSession<D>::processDataDecode (unsigned char * spkrData, unsigned int size)
{
-
if (_audiocodec != NULL) {
@@ -660,11 +720,12 @@ void AudioRtpSession<D>::processDataDecode (unsigned char * spkrData, unsigned i
// Return the size of data in bytes
int expandedSize = _audiocodec->codecDecode (_spkrDataDecoded , spkrData , size);
- // captureFile->write ((const char *)_spkrDataDecoded, expandedSize);
-
// buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
int nbSample = expandedSize / sizeof (SFLDataFormat);
+ if (!_spkrFadeInComplete)
+ _spkrFadeInComplete = fadeIn (_spkrDataDecoded, nbSample, &_spkrAmplFactor);
+
// test if resampling is required
if (_audiocodec->getClockRate() != _mainBufferSampleRate) {
@@ -764,36 +825,42 @@ void AudioRtpSession<D>::receiveSpeakerData ()
unsigned char* spkrDataIn = NULL;
unsigned int size = 0;
- int result;
-
- jb_frame frame;
-
- _jbuffer->info.conf.resync_threshold = 0;
if (adu) {
spkrDataIn = (unsigned char*) adu->getData(); // data in char
size = adu->getSize(); // size in char
- result = jb_put (_jbuffer, spkrDataIn, JB_TYPE_VOICE, _packetLength, _ts+=20, _currentTime);
-
} else {
_debug ("No RTP packet available !!!!!!!!!!!!!!!!!!!!!!!\n");
}
- result = jb_get (_jbuffer, &frame, _currentTime+=20, _packetLength);
-
// DTMF over RTP, size must be over 4 in order to process it as voice data
if (size > 4) {
processDataDecode (spkrDataIn, size);
- //if(result == JB_OK) {
- // processDataDecode((unsigned char *)(frame.data), 160);
- //}
}
delete adu;
}
+template <typename D>
+bool AudioRtpSession<D>::fadeIn (SFLDataFormat *audio, int size, SFLDataFormat *factor)
+{
+
+ // apply amplitude factor;
+ while (size) {
+ size--;
+ audio[size] /= *factor;
+ }
+
+ *factor /= 2;
+
+ if (*factor == 0)
+ return true;
+
+ return false;
+}
+
template <typename D>
int AudioRtpSession<D>::startRtpThread (AudioCodec* audiocodec)
{
@@ -835,6 +902,8 @@ void AudioRtpSession<D>::run ()
_debug ("RTP: Entering mainloop for call %s",_ca->getCallId().c_str());
+ _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
+
while (!testCancel()) {
// Reset timestamp to make sure the timing information are up to date