AudioRtpRecordHandler / AudioRtpSession : simplify

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();
@@ -383,10 +375,8 @@ 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));
@@ -407,22 +397,16 @@ void AudioRtpRecordHandler::processDataDecode (unsigned char *spkrData, unsigned
 
 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;
 }
 
 }

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;
 }
@@ -324,7 +313,6 @@ void AudioRtpSession::stopRtpThread ()
 bool AudioRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&)
 {
     receiveSpeakerData();
-
     return true;
 }