Merge branch 'wfserver' of https://gitlab.com/eliggett/wfview into wfserver

audioconverter.cpp

@@ -82,10 +82,9 @@ audioConverter::~audioConverter()
 bool audioConverter::convert(audioPacket audio)
 {
 
-	if (audio.data.size() == 0)
+    // If inFormat and outFormat are identical, just emit the data back.
+    if (audio.data.size() != 0 && inFormat != outFormat)
 	{
-		return false;
-	}
 
         if (inFormat.codec() == "audio/opus")
         {
@@ -300,6 +299,8 @@ bool audioConverter::convert(audioPacket audio)
 
             }
         }
+    }
+
 	emit converted(audio);
 	return true;
 }

audioconverter.h

@@ -85,6 +85,7 @@ static inline QAudioFormat toQAudioFormat(quint8 codec, quint32 sampleRate)
 	format.setSampleType(QAudioFormat::UnSignedInt);
 	format.setByteOrder(QAudioFormat::LittleEndian);
 	format.setCodec("audio/pcm");
+    format.setSampleRate(sampleRate);
 
 	if (codec == 0x01 || codec == 0x20) {
 		/* Set sample to be what is expected by the encoder and the output of the decoder */
@@ -108,7 +109,6 @@ static inline QAudioFormat toQAudioFormat(quint8 codec, quint32 sampleRate)
 		format.setCodec("audio/opus");
 	}
 
-	format.setSampleRate(sampleRate);
 	return format;
 }
 

audiohandler.cpp

@@ -209,8 +209,8 @@ void audioHandler::setVolume(unsigned char volume)
 
 void audioHandler::incomingAudio(audioPacket packet)
 {
-	//QTime startProcessing = QTime::currentTime();
-	if (audioDevice != Q_NULLPTR) {
+
+    if (audioDevice != Q_NULLPTR && packet.data.size() > 0) {
 		packet.volume = volume;
 
 		emit sendToConverter(packet);
@@ -221,6 +221,8 @@ void audioHandler::incomingAudio(audioPacket packet)
 
 void audioHandler::convertedOutput(audioPacket packet) {
 	
+    if (packet.data.size() > 0 ) {
+
         currentLatency = packet.time.msecsTo(QTime::currentTime()) + (outFormat.durationForBytes(audioOutput->bufferSize() - audioOutput->bytesFree()) / 1000);
         if (audioDevice != Q_NULLPTR) {
             audioDevice->write(packet.data);
@@ -239,14 +241,15 @@ void audioHandler::convertedOutput(audioPacket packet) {
         emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun);
 
         amplitude = packet.amplitude;
-
+    }
 }
 
 void audioHandler::getNextAudioChunk()
 {
+    if (audioDevice) {
         tempBuf.data.append(audioDevice->readAll());
-
-	while (tempBuf.data.length() >= outFormat.bytesForDuration(setup.blockSize * 1000)) {
+    }
+    if (tempBuf.data.length() >= outFormat.bytesForDuration(setup.blockSize * 1000)) {
 		audioPacket packet;
 		packet.time = QTime::currentTime();
 		packet.sent = 0;
@@ -259,13 +262,19 @@ void audioHandler::getNextAudioChunk()
 
 		emit sendToConverter(packet);
 	}
-	return;
 
+    /* If there is still enough data in the buffer, call myself again in 20ms */
+    if (tempBuf.data.length() >= outFormat.bytesForDuration(setup.blockSize * 1000)) {
+        QTimer::singleShot(setup.blockSize, this, &audioHandler::getNextAudioChunk);
+    }
+
+	return;
 }
 
 
 void audioHandler::convertedInput(audioPacket audio) 
 {
+    if (audio.data.size() > 0) {
         emit haveAudioData(audio);
         if (lastReceived.msecsTo(QTime::currentTime()) > 100) {
             qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Time since last audio packet" << lastReceived.msecsTo(QTime::currentTime()) << "Expected around" << setup.blockSize ;
@@ -274,6 +283,7 @@ void audioHandler::convertedInput(audioPacket audio)
         amplitude = audio.amplitude;
         emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun);
     }
+}
 
 void audioHandler::changeLatency(const quint16 newSize)
 {