Merge branch 'master' of github.com:VCVRack/Rack

Makefile

@@ -102,6 +102,7 @@ ifeq ($(ARCH), mac)
 	cp dep/lib/libzip.5.dylib $(BUNDLE)/Contents/MacOS/
 	cp dep/lib/libportaudio.2.dylib $(BUNDLE)/Contents/MacOS/
 	cp dep/lib/librtmidi.4.dylib $(BUNDLE)/Contents/MacOS/
+	cp dep/lib/librtaudio.6.dylib $(BUNDLE)/Contents/MacOS/
 
 	install_name_tool -change /usr/local/lib/libGLEW.2.1.0.dylib @executable_path/libGLEW.2.1.0.dylib $(BUNDLE)/Contents/MacOS/Rack
 	install_name_tool -change lib/libglfw.3.dylib @executable_path/libglfw.3.dylib $(BUNDLE)/Contents/MacOS/Rack
@@ -111,6 +112,7 @@ ifeq ($(ARCH), mac)
 	install_name_tool -change $(PWD)/dep/lib/libzip.5.dylib @executable_path/libzip.5.dylib $(BUNDLE)/Contents/MacOS/Rack
 	install_name_tool -change $(PWD)/dep/lib/libportaudio.2.dylib @executable_path/libportaudio.2.dylib $(BUNDLE)/Contents/MacOS/Rack
 	install_name_tool -change $(PWD)/dep/lib/librtmidi.4.dylib @executable_path/librtmidi.4.dylib $(BUNDLE)/Contents/MacOS/Rack
+	install_name_tool -change $(PWD)/dep/lib/librtaudio.6.dylib @executable_path/librtaudio.6.dylib $(BUNDLE)/Contents/MacOS/Rack
 
 	otool -L $(BUNDLE)/Contents/MacOS/Rack
 

dep/Makefile

@@ -58,13 +58,15 @@ endif
 # Library configuration
 ifdef RTAUDIO_ALL_APIS
 ifeq ($(ARCH),mac)
-	RTAUDIO_FLAGS = --with-core --with-jack
+	RTAUDIO_FLAGS = --with-core
+	#--with-jack
 endif
 ifeq ($(ARCH),win)
 	RTAUDIO_FLAGS = --with-asio --with-wasapi --with-ds
 endif
 ifeq ($(ARCH),lin)
-	RTAUDIO_FLAGS = --with-alsa --with-pulse --with-jack
+	RTAUDIO_FLAGS = --with-alsa --with-pulse
+	#--with-jack
 endif
 endif
 

src/core/AudioInterface.cpp

@@ -36,9 +36,10 @@ struct AudioInterface : Module {
 	int blockSize = 256;
 	int numOutputs = 0;
 	int numInputs = 0;
+	bool streamRunning = false;
 
 	// Used because the GUI thread and Rack thread can both interact with this class
-	std::mutex bufferMutex;
+	std::timed_mutex bufferMutex;
 
 	SampleRateConverter<8> inputSrc;
 	SampleRateConverter<8> outputSrc;
@@ -117,17 +118,18 @@ struct AudioInterface : Module {
 void AudioInterface::step() {
 	// Read/write stream if we have enough input, OR the output buffer is empty if we have no input
 	if (numOutputs > 0) {
-		while (inputSrcBuffer.size() >= blockSize && stream.isStreamRunning()) {
+		while (inputSrcBuffer.size() >= blockSize && streamRunning) {
 			std::this_thread::sleep_for(std::chrono::duration<float>(100e-6));
 		}
 	}
 	else if (numInputs > 0) {
-		while (outputBuffer.empty() && stream.isStreamRunning()) {
+		while (outputBuffer.empty() && streamRunning) {
 			std::this_thread::sleep_for(std::chrono::duration<float>(100e-6));
 		}
 	}
 
-	std::lock_guard<std::mutex> lock(bufferMutex);
+	if (!bufferMutex.try_lock_for(std::chrono::duration<float>(10e-3)))
+		return;
 
 	// Get input and pass it through the sample rate converter
 	if (numOutputs > 0) {
@@ -158,6 +160,8 @@ void AudioInterface::step() {
 			outputs[AUDIO1_OUTPUT + i].value = 10.0 * f.samples[i];
 		}
 	}
+
+	bufferMutex.unlock();
 }
 
 void AudioInterface::stepStream(const float *input, float *output, int numFrames) {
@@ -168,14 +172,14 @@ void AudioInterface::stepStream(const float *input, float *output, int numFrames
 
 	if (numOutputs > 0) {
 		// Wait for enough input before proceeding
-		while (inputSrcBuffer.size() < numFrames) {
-			if (!stream.isStreamRunning())
-				return;
+		while (inputSrcBuffer.size() < numFrames && streamRunning) {
 			std::this_thread::sleep_for(std::chrono::duration<float>(100e-6));
 		}
 	}
 
-	std::lock_guard<std::mutex> lock(bufferMutex);
+	if (!bufferMutex.try_lock_for(std::chrono::duration<float>(10e-3)))
+		return;
+
 	// input stream -> output buffer
 	if (numInputs > 0) {
 		Frame<8> inputFrames[numFrames];
@@ -204,6 +208,7 @@ void AudioInterface::stepStream(const float *input, float *output, int numFrames
 			}
 		}
 	}
+	bufferMutex.unlock();
 }
 
 int AudioInterface::getDeviceCount() {
@@ -233,7 +238,7 @@ static int rtCallback(void *outputBuffer, void *inputBuffer, unsigned int nFrame
 
 void AudioInterface::openDevice(int deviceId, float sampleRate, int blockSize) {
 	closeDevice();
-	std::lock_guard<std::mutex> lock(bufferMutex);
+	std::lock_guard<std::timed_mutex> lock(bufferMutex);
 
 	this->sampleRate = sampleRate;
 	this->blockSize = blockSize;
@@ -265,6 +270,7 @@ void AudioInterface::openDevice(int deviceId, float sampleRate, int blockSize) {
 
 		try {
 			// Don't use stream parameters if 0 input or output channels
+			debug("Opening audio stream %d", deviceId);
 			stream.openStream(
 				numOutputs == 0 ? NULL : &outParameters,
 				numInputs == 0 ? NULL : &inParameters,
@@ -275,7 +281,10 @@ void AudioInterface::openDevice(int deviceId, float sampleRate, int blockSize) {
 			return;
 		}
 
+		streamRunning = true;
+
 		try {
+			debug("Starting audio stream %d", deviceId);
 			stream.startStream();
 		}
 		catch (RtAudioError &e) {
@@ -289,11 +298,14 @@ void AudioInterface::openDevice(int deviceId, float sampleRate, int blockSize) {
 }
 
 void AudioInterface::closeDevice() {
-	std::lock_guard<std::mutex> lock(bufferMutex);
+	std::lock_guard<std::timed_mutex> lock(bufferMutex);
 
 	if (stream.isStreamOpen()) {
+		streamRunning = false;
 		try {
+			debug("Aborting audio stream %d", deviceId);
 			stream.abortStream();
+			debug("Closing audio stream %d", deviceId);
 			stream.closeStream();
 		}
 		catch (RtAudioError &e) {

src/main.cpp

@@ -35,13 +35,7 @@ int main(int argc, char* argv[]) {
 	engineInit();
 	guiInit();
 	sceneInit();
-	if (argc >= 2) {
-		// TODO Set gRackWidget->lastPath
-		gRackWidget->loadPatch(argv[1]);
-	}
-	else {
-		gRackWidget->loadPatch(assetLocal("autosave.vcv"));
-	}
+	gRackWidget->loadPatch(assetLocal("autosave.vcv"));
 	settingsLoad(assetLocal("settings.json"));
 
 	engineStart();