fixed buffer size issue when using input chan > 1, removed redundant member vars

src/ofApp.cpp

@@ -64,17 +64,15 @@ void ofApp::setup() {
 		ofLogWarning(PACKAGE) << "audio input device does not have enough input channels";
 		inputChannel = 0;
 	}
-	numInputChannels = inputChannel+1;
-	recordedSamplesPerBuffer = bufferSize * numInputChannels;
 	ofLogNotice(PACKAGE) << "audio input channel: " << inputChannel;
 	ofLogNotice(PACKAGE) << "audio input samplerate: " << sampleRate;
-	ofLogNotice(PACKAGE) << "audio input buffer size: " << recordedSamplesPerBuffer;
+	ofLogNotice(PACKAGE) << "audio input buffer size: " << bufferSize;
 	settings.setInDevice(device);
 	settings.setInListener(this);
 	settings.sampleRate = sampleRate;
 	settings.numOutputChannels = 0;
-	settings.numInputChannels = numInputChannels;
-	settings.bufferSize = recordedSamplesPerBuffer;
+	settings.numInputChannels = inputChannel + 1;
+	settings.bufferSize = bufferSize * (inputChannel + 1);
 	if(!soundStream.setup(settings)) {
 		ofLogError(PACKAGE) << "audio input device " << inputDevice << " setup failed";
 		ofLogError(PACKAGE) << "perhaps try a different device or samplerate?";
@@ -84,6 +82,9 @@ void ofApp::setup() {
 	if(!listening) {
 		soundStream.stop();
 	}
+	if(soundStream.getSampleRate() == 44100) {
+		ofLogWarning(PACKAGE) << "treating sample rate of 44100 as 48000, may or may not affect detection";
+	}
 
 	// display
 	volHistory.assign(400, 0.0);
@@ -255,11 +256,13 @@ void ofApp::exit() {
 
 //--------------------------------------------------------------
 void ofApp::audioIn(ofSoundBuffer & input) {
+	// beh, ofSoundBuffer::getNumFrames() actually returns the buffer size?
+	std::size_t numFrames = input.getNumFrames() / input.getNumChannels();
 
 	// copy desired channel out of interleaved stream into mono buffer,
 	// assume input stream has enough channels...
-	for(std::size_t i = 0; i < input.getNumFrames(); i++) {
-		monoBuffer[i] = input[i*numInputChannels+inputChannel];
+	for(std::size_t i = 0; i < numFrames; i++) {
+		monoBuffer[i] = input[(i*input.getNumChannels())+inputChannel];
 	}
 
 	// calculate the root mean square which is a rough way to calculate volume

src/ofApp.h

@@ -67,8 +67,8 @@ class ofApp : public ofBaseApp {
 
 		// audio 
 		ofSoundStream soundStream;
-		int inputDevice = -1;			// -1 means search for default device
-		int inputChannel = 0;			// 0 means mono, 1 means stereo
+		int inputDevice = -1; // -1 means search for default device
+		int inputChannel = 0; // 0 - chan 1 (left), 1 - chan 2 (right), 2 - chan 3, etc
 		bool listening = true;
 
 		// neural network input parameters
@@ -76,20 +76,17 @@ class ofApp : public ofBaseApp {
 		// we want to keep the buffersize a multiple of the downsampling factor
 		// downsamplingFactor = sampleRate / modelSampleRate
 		// downsampling is required for microphones that do not have 16kHz sampling
-		std::size_t bufferSize = 1024;
+		std::size_t bufferSize = 1024; //< in this case, number of sample frames
 		std::size_t sampleRate = 48000;
 		std::size_t downsamplingFactor = 3;
-		std::size_t recordedSamplesPerBuffer;
-		std::size_t numInputChannels;
 
-		
-		// since volume detection has some latency we keep a history of buffers
+		// since volume detection has some latency,d we keep a history of buffers
 		AudioBufferFifo previousBuffers;
 		std::size_t numPreviousBuffers = 10; // how many buffers to save before trigger happens
 		// sampleBuffers acts as a buffer for recording (could be fused)
 		AudioBufferFifo sampleBuffers;
 		std::size_t numBuffers;
-		SimpleAudioBuffer monoBuffer;	//< mono inputChannel stream buffer
+		SimpleAudioBuffer monoBuffer; //< mono inputChannel stream buffer
 		
 		// volume
 		float curVol = 0.0;