Split AudioInterface into AudioInterfacePort, avoiding multiple inheritance.

src/core/AudioInterface.cpp

@@ -13,7 +13,166 @@ namespace core {
 
 
 template <int NUM_AUDIO_INPUTS, int NUM_AUDIO_OUTPUTS>
-struct AudioInterface : Module, audio::Port {
+struct AudioInterfacePort : audio::Port {
+	Module* module;
+
+	dsp::DoubleRingBuffer<dsp::Frame<NUM_AUDIO_INPUTS>, 32768> engineInputBuffer;
+	dsp::DoubleRingBuffer<dsp::Frame<NUM_AUDIO_OUTPUTS>, 32768> engineOutputBuffer;
+
+	dsp::SampleRateConverter<NUM_AUDIO_INPUTS> inputSrc;
+	dsp::SampleRateConverter<NUM_AUDIO_OUTPUTS> outputSrc;
+
+	// Port variable caches
+	int deviceNumInputs = 0;
+	int deviceNumOutputs = 0;
+	float deviceSampleRate = 0.f;
+	int requestedEngineFrames = 0;
+
+	AudioInterfacePort(Module* module) {
+		this->module = module;
+		maxOutputs = NUM_AUDIO_INPUTS;
+		maxInputs = NUM_AUDIO_OUTPUTS;
+		inputSrc.setQuality(6);
+		outputSrc.setQuality(6);
+	}
+
+	void setPrimary() {
+		APP->engine->setPrimaryModule(module);
+	}
+
+	bool isPrimary() {
+		return APP->engine->getPrimaryModule() == module;
+	}
+
+	void processInput(const float* input, int inputStride, int frames) override {
+		// DEBUG("%p: new device block ____________________________", this);
+		// Claim primary module if there is none
+		if (!APP->engine->getPrimaryModule()) {
+			setPrimary();
+		}
+		bool isPrimaryCached = isPrimary();
+
+		// Set sample rate of engine if engine sample rate is "auto".
+		if (isPrimaryCached) {
+			APP->engine->setSuggestedSampleRate(deviceSampleRate);
+		}
+
+		float engineSampleRate = APP->engine->getSampleRate();
+		float sampleRateRatio = engineSampleRate / deviceSampleRate;
+
+		// DEBUG("%p: %d block, engineOutputBuffer still has %d", this, frames, (int) engineOutputBuffer.size());
+
+		// Consider engine buffers "too full" if they contain a bit more than the audio device's number of frames, converted to engine sample rate.
+		int maxEngineFrames = (int) std::ceil(frames * sampleRateRatio * 2.0) - 1;
+		// If the engine output buffer is too full, clear it to keep latency low. No need to clear if primary because it's always cleared below.
+		if (!isPrimaryCached && (int) engineOutputBuffer.size() > maxEngineFrames) {
+			engineOutputBuffer.clear();
+			// DEBUG("%p: clearing engine output", this);
+		}
+
+		if (deviceNumInputs > 0) {
+			// Always clear engine output if primary
+			if (isPrimaryCached) {
+				engineOutputBuffer.clear();
+			}
+			// Set up sample rate converter
+			outputSrc.setRates(deviceSampleRate, engineSampleRate);
+			outputSrc.setChannels(deviceNumInputs);
+			// Convert audio input -> engine output
+			dsp::Frame<NUM_AUDIO_OUTPUTS> audioInputBuffer[frames];
+			std::memset(audioInputBuffer, 0, sizeof(audioInputBuffer));
+			for (int i = 0; i < frames; i++) {
+				for (int j = 0; j < deviceNumInputs; j++) {
+					float v = input[i * inputStride + j];
+					audioInputBuffer[i].samples[j] = v;
+				}
+			}
+			int audioInputFrames = frames;
+			int outputFrames = engineOutputBuffer.capacity();
+			outputSrc.process(audioInputBuffer, &audioInputFrames, engineOutputBuffer.endData(), &outputFrames);
+			engineOutputBuffer.endIncr(outputFrames);
+			// Request exactly as many frames as we have in the engine output buffer.
+			requestedEngineFrames = engineOutputBuffer.size();
+		}
+		else {
+			// Upper bound on number of frames so that `audioOutputFrames >= frames` when processOutput() is called.
+			requestedEngineFrames = std::max((int) std::ceil(frames * sampleRateRatio) - (int) engineInputBuffer.size(), 0);
+		}
+	}
+
+	void processBuffer(const float* input, int inputStride, float* output, int outputStride, int frames) override {
+		// Step engine
+		if (isPrimary() && requestedEngineFrames > 0) {
+			// DEBUG("%p: %d block, stepping %d", this, frames, requestedEngineFrames);
+			APP->engine->stepBlock(requestedEngineFrames);
+		}
+	}
+
+	void processOutput(float* output, int outputStride, int frames) override {
+		// bool isPrimaryCached = isPrimary();
+		float engineSampleRate = APP->engine->getSampleRate();
+		float sampleRateRatio = engineSampleRate / deviceSampleRate;
+
+		if (deviceNumOutputs > 0) {
+			// Set up sample rate converter
+			inputSrc.setRates(engineSampleRate, deviceSampleRate);
+			inputSrc.setChannels(deviceNumOutputs);
+			// Convert engine input -> audio output
+			dsp::Frame<NUM_AUDIO_OUTPUTS> audioOutputBuffer[frames];
+			int inputFrames = engineInputBuffer.size();
+			int audioOutputFrames = frames;
+			inputSrc.process(engineInputBuffer.startData(), &inputFrames, audioOutputBuffer, &audioOutputFrames);
+			engineInputBuffer.startIncr(inputFrames);
+			// Copy the audio output buffer
+			for (int i = 0; i < audioOutputFrames; i++) {
+				for (int j = 0; j < deviceNumOutputs; j++) {
+					float v = audioOutputBuffer[i].samples[j];
+					v = clamp(v, -1.f, 1.f);
+					output[i * outputStride + j] = v;
+				}
+			}
+			// Fill the rest of the audio output buffer with zeros
+			for (int i = audioOutputFrames; i < frames; i++) {
+				for (int j = 0; j < deviceNumOutputs; j++) {
+					output[i * outputStride + j] = 0.f;
+				}
+			}
+		}
+
+		// DEBUG("%p: %d block, engineInputBuffer left %d", this, frames, (int) engineInputBuffer.size());
+
+		// If the engine input buffer is too full, clear it to keep latency low.
+		int maxEngineFrames = (int) std::ceil(frames * sampleRateRatio * 2.0) - 1;
+		if ((int) engineInputBuffer.size() > maxEngineFrames) {
+			engineInputBuffer.clear();
+			// DEBUG("%p: clearing engine input", this);
+		}
+
+		// DEBUG("%p %s:\tframes %d requestedEngineFrames %d\toutputBuffer %d engineInputBuffer %d\t", this, isPrimaryCached ? "primary" : "secondary", frames, requestedEngineFrames, engineOutputBuffer.size(), engineInputBuffer.size());
+	}
+
+	void onStartStream() override {
+		deviceNumInputs = std::min(getNumInputs(), NUM_AUDIO_OUTPUTS);
+		deviceNumOutputs = std::min(getNumOutputs(), NUM_AUDIO_INPUTS);
+		deviceSampleRate = getSampleRate();
+		engineInputBuffer.clear();
+		engineOutputBuffer.clear();
+		// DEBUG("onStartStream %d %d %f", deviceNumInputs, deviceNumOutputs, deviceSampleRate);
+	}
+
+	void onStopStream() override {
+		deviceNumInputs = 0;
+		deviceNumOutputs = 0;
+		deviceSampleRate = 0.f;
+		engineInputBuffer.clear();
+		engineOutputBuffer.clear();
+		// DEBUG("onStopStream");
+	}
+};
+
+
+template <int NUM_AUDIO_INPUTS, int NUM_AUDIO_OUTPUTS>
+struct AudioInterface : Module {
 	static constexpr int NUM_INPUT_LIGHTS = (NUM_AUDIO_INPUTS > 2) ? (NUM_AUDIO_INPUTS / 2) : 0;
 	static constexpr int NUM_OUTPUT_LIGHTS = (NUM_AUDIO_OUTPUTS > 2) ? (NUM_AUDIO_OUTPUTS / 2) : 0;
 
@@ -36,28 +195,18 @@ struct AudioInterface : Module, audio::Port {
 		NUM_LIGHTS
 	};
 
+	AudioInterfacePort<NUM_AUDIO_INPUTS, NUM_AUDIO_OUTPUTS> port;
+
 	dsp::RCFilter dcFilters[NUM_AUDIO_INPUTS];
 	bool dcFilterEnabled = false;
 
-	dsp::DoubleRingBuffer<dsp::Frame<NUM_AUDIO_INPUTS>, 32768> engineInputBuffer;
-	dsp::DoubleRingBuffer<dsp::Frame<NUM_AUDIO_OUTPUTS>, 32768> engineOutputBuffer;
-
-	dsp::SampleRateConverter<NUM_AUDIO_INPUTS> inputSrc;
-	dsp::SampleRateConverter<NUM_AUDIO_OUTPUTS> outputSrc;
-
 	dsp::ClockDivider lightDivider;
 	// For each pair of inputs/outputs
 	float inputClipTimers[(NUM_AUDIO_INPUTS > 0) ? NUM_INPUT_LIGHTS : 0] = {};
 	float outputClipTimers[(NUM_AUDIO_INPUTS > 0) ? NUM_OUTPUT_LIGHTS : 0] = {};
 	dsp::VuMeter2 vuMeter[(NUM_AUDIO_INPUTS == 2) ? 2 : 0];
 
-	// Port variable caches
-	int deviceNumInputs = 0;
-	int deviceNumOutputs = 0;
-	float deviceSampleRate = 0.f;
-	int requestedEngineFrames = 0;
-
-	AudioInterface() {
+	AudioInterface() : port(this) {
 		config(NUM_PARAMS, NUM_INPUTS, NUM_OUTPUTS, NUM_LIGHTS);
 		if (NUM_AUDIO_INPUTS == 2)
 			configParam(GAIN_PARAM, 0.f, 2.f, 1.f, "Level", " dB", -10, 40);
@@ -71,26 +220,22 @@ struct AudioInterface : Module, audio::Port {
 			configLight(OUTPUT_LIGHTS + 2 * i, string::f("Device input %d/%d status", 2 * i + 1, 2 * i + 2));
 
 		lightDivider.setDivision(512);
-		maxOutputs = NUM_AUDIO_INPUTS;
-		maxInputs = NUM_AUDIO_OUTPUTS;
-		inputSrc.setQuality(6);
-		outputSrc.setQuality(6);
 
 		float sampleTime = APP->engine->getSampleTime();
 		for (int i = 0; i < NUM_AUDIO_INPUTS; i++) {
 			dcFilters[i].setCutoffFreq(10.f * sampleTime);
 		}
 
-		reset();
+		onReset();
 	}
 
 	~AudioInterface() {
 		// Close stream here before destructing AudioInterfacePort, so processBuffer() etc are not called on another thread while destructing.
-		setDriverId(-1);
+		port.setDriverId(-1);
 	}
 
 	void onReset() override {
-		setDriverId(-1);
+		port.setDriverId(-1);
 
 		if (NUM_AUDIO_INPUTS == 2)
 			dcFilterEnabled = true;
@@ -99,8 +244,8 @@ struct AudioInterface : Module, audio::Port {
 	}
 
 	void onSampleRateChange(const SampleRateChangeEvent& e) override {
-		engineInputBuffer.clear();
-		engineOutputBuffer.clear();
+		port.engineInputBuffer.clear();
+		port.engineOutputBuffer.clear();
 
 		for (int i = 0; i < NUM_AUDIO_INPUTS; i++) {
 			dcFilters[i].setCutoffFreq(10.f * e.sampleTime);
@@ -111,9 +256,9 @@ struct AudioInterface : Module, audio::Port {
 		const float clipTime = 0.25f;
 
 		// Push inputs to buffer
-		if (deviceNumOutputs > 0) {
+		if (port.deviceNumOutputs > 0) {
 			dsp::Frame<NUM_AUDIO_INPUTS> inputFrame = {};
-			for (int i = 0; i < deviceNumOutputs; i++) {
+			for (int i = 0; i < port.deviceNumOutputs; i++) {
 				// Get input
 				float v = 0.f;
 				if (inputs[AUDIO_INPUTS + i].isConnected())
@@ -144,8 +289,8 @@ struct AudioInterface : Module, audio::Port {
 				}
 			}
 
-			if (!engineInputBuffer.full()) {
-				engineInputBuffer.push(inputFrame);
+			if (!port.engineInputBuffer.full()) {
+				port.engineInputBuffer.push(inputFrame);
 			}
 
 			// Audio-2: VU meter process
@@ -165,8 +310,8 @@ struct AudioInterface : Module, audio::Port {
 		}
 
 		// Pull outputs from buffer
-		if (!engineOutputBuffer.empty()) {
-			dsp::Frame<NUM_AUDIO_OUTPUTS> outputFrame = engineOutputBuffer.shift();
+		if (!port.engineOutputBuffer.empty()) {
+			dsp::Frame<NUM_AUDIO_OUTPUTS> outputFrame = port.engineOutputBuffer.shift();
 			for (int i = 0; i < NUM_AUDIO_OUTPUTS; i++) {
 				float v = outputFrame.samples[i];
 				outputs[AUDIO_OUTPUTS + i].setVoltage(10.f * v);
@@ -203,7 +348,7 @@ struct AudioInterface : Module, audio::Port {
 			else {
 				// Turn on light if at least one port is enabled in the nearby pair.
 				for (int i = 0; i < NUM_AUDIO_INPUTS / 2; i++) {
-					bool active = deviceNumOutputs >= 2 * i + 1;
+					bool active = port.deviceNumOutputs >= 2 * i + 1;
 					bool clip = inputClipTimers[i] > 0.f;
 					if (clip)
 						inputClipTimers[i] -= lightTime;
@@ -211,7 +356,7 @@ struct AudioInterface : Module, audio::Port {
 					lights[INPUT_LIGHTS + i * 2 + 1].setBrightness(active && clip);
 				}
 				for (int i = 0; i < NUM_AUDIO_OUTPUTS / 2; i++) {
-					bool active = deviceNumInputs >= 2 * i + 1;
+					bool active = port.deviceNumInputs >= 2 * i + 1;
 					bool clip = outputClipTimers[i] > 0.f;
 					if (clip)
 						outputClipTimers[i] -= lightTime;
@@ -224,7 +369,7 @@ struct AudioInterface : Module, audio::Port {
 
 	json_t* dataToJson() override {
 		json_t* rootJ = json_object();
-		json_object_set_new(rootJ, "audio", audio::Port::toJson());
+		json_object_set_new(rootJ, "audio", port.toJson());
 
 		if (isPrimary())
 			json_object_set_new(rootJ, "primary", json_boolean(true));
@@ -237,7 +382,7 @@ struct AudioInterface : Module, audio::Port {
 	void dataFromJson(json_t* rootJ) override {
 		json_t* audioJ = json_object_get(rootJ, "audio");
 		if (audioJ)
-			audio::Port::fromJson(audioJ);
+			port.fromJson(audioJ);
 
 		// For not, don't deserialize primary module state.
 		// json_t* primaryJ = json_object_get(rootJ, "primary");
@@ -258,133 +403,6 @@ struct AudioInterface : Module, audio::Port {
 	bool isPrimary() {
 		return APP->engine->getPrimaryModule() == this;
 	}
-
-	// audio::Port
-
-	void processInput(const float* input, int inputStride, int frames) override {
-		// DEBUG("%p: new device block ____________________________", this);
-		// Claim primary module if there is none
-		if (!APP->engine->getPrimaryModule()) {
-			setPrimary();
-		}
-		bool isPrimaryCached = isPrimary();
-
-		// Set sample rate of engine if engine sample rate is "auto".
-		if (isPrimaryCached) {
-			APP->engine->setSuggestedSampleRate(deviceSampleRate);
-		}
-
-		float engineSampleRate = APP->engine->getSampleRate();
-		float sampleRateRatio = engineSampleRate / deviceSampleRate;
-
-		// DEBUG("%p: %d block, engineOutputBuffer still has %d", this, frames, (int) engineOutputBuffer.size());
-
-		// Consider engine buffers "too full" if they contain a bit more than the audio device's number of frames, converted to engine sample rate.
-		int maxEngineFrames = (int) std::ceil(frames * sampleRateRatio * 2.0) - 1;
-		// If the engine output buffer is too full, clear it to keep latency low. No need to clear if primary because it's always cleared below.
-		if (!isPrimaryCached && (int) engineOutputBuffer.size() > maxEngineFrames) {
-			engineOutputBuffer.clear();
-			// DEBUG("%p: clearing engine output", this);
-		}
-
-		if (deviceNumInputs > 0) {
-			// Always clear engine output if primary
-			if (isPrimaryCached) {
-				engineOutputBuffer.clear();
-			}
-			// Set up sample rate converter
-			outputSrc.setRates(deviceSampleRate, engineSampleRate);
-			outputSrc.setChannels(deviceNumInputs);
-			// Convert audio input -> engine output
-			dsp::Frame<NUM_AUDIO_OUTPUTS> audioInputBuffer[frames];
-			std::memset(audioInputBuffer, 0, sizeof(audioInputBuffer));
-			for (int i = 0; i < frames; i++) {
-				for (int j = 0; j < deviceNumInputs; j++) {
-					float v = input[i * inputStride + j];
-					audioInputBuffer[i].samples[j] = v;
-				}
-			}
-			int audioInputFrames = frames;
-			int outputFrames = engineOutputBuffer.capacity();
-			outputSrc.process(audioInputBuffer, &audioInputFrames, engineOutputBuffer.endData(), &outputFrames);
-			engineOutputBuffer.endIncr(outputFrames);
-			// Request exactly as many frames as we have in the engine output buffer.
-			requestedEngineFrames = engineOutputBuffer.size();
-		}
-		else {
-			// Upper bound on number of frames so that `audioOutputFrames >= frames` when processOutput() is called.
-			requestedEngineFrames = std::max((int) std::ceil(frames * sampleRateRatio) - (int) engineInputBuffer.size(), 0);
-		}
-	}
-
-	void processBuffer(const float* input, int inputStride, float* output, int outputStride, int frames) override {
-		// Step engine
-		if (isPrimary() && requestedEngineFrames > 0) {
-			// DEBUG("%p: %d block, stepping %d", this, frames, requestedEngineFrames);
-			APP->engine->stepBlock(requestedEngineFrames);
-		}
-	}
-
-	void processOutput(float* output, int outputStride, int frames) override {
-		// bool isPrimaryCached = isPrimary();
-		float engineSampleRate = APP->engine->getSampleRate();
-		float sampleRateRatio = engineSampleRate / deviceSampleRate;
-
-		if (deviceNumOutputs > 0) {
-			// Set up sample rate converter
-			inputSrc.setRates(engineSampleRate, deviceSampleRate);
-			inputSrc.setChannels(deviceNumOutputs);
-			// Convert engine input -> audio output
-			dsp::Frame<NUM_AUDIO_OUTPUTS> audioOutputBuffer[frames];
-			int inputFrames = engineInputBuffer.size();
-			int audioOutputFrames = frames;
-			inputSrc.process(engineInputBuffer.startData(), &inputFrames, audioOutputBuffer, &audioOutputFrames);
-			engineInputBuffer.startIncr(inputFrames);
-			// Copy the audio output buffer
-			for (int i = 0; i < audioOutputFrames; i++) {
-				for (int j = 0; j < deviceNumOutputs; j++) {
-					float v = audioOutputBuffer[i].samples[j];
-					v = clamp(v, -1.f, 1.f);
-					output[i * outputStride + j] = v;
-				}
-			}
-			// Fill the rest of the audio output buffer with zeros
-			for (int i = audioOutputFrames; i < frames; i++) {
-				for (int j = 0; j < deviceNumOutputs; j++) {
-					output[i * outputStride + j] = 0.f;
-				}
-			}
-		}
-
-		// DEBUG("%p: %d block, engineInputBuffer left %d", this, frames, (int) engineInputBuffer.size());
-
-		// If the engine input buffer is too full, clear it to keep latency low.
-		int maxEngineFrames = (int) std::ceil(frames * sampleRateRatio * 2.0) - 1;
-		if ((int) engineInputBuffer.size() > maxEngineFrames) {
-			engineInputBuffer.clear();
-			// DEBUG("%p: clearing engine input", this);
-		}
-
-		// DEBUG("%p %s:\tframes %d requestedEngineFrames %d\toutputBuffer %d engineInputBuffer %d\t", this, isPrimaryCached ? "primary" : "secondary", frames, requestedEngineFrames, engineOutputBuffer.size(), engineInputBuffer.size());
-	}
-
-	void onStartStream() override {
-		deviceNumInputs = std::min(getNumInputs(), NUM_AUDIO_OUTPUTS);
-		deviceNumOutputs = std::min(getNumOutputs(), NUM_AUDIO_INPUTS);
-		deviceSampleRate = getSampleRate();
-		engineInputBuffer.clear();
-		engineOutputBuffer.clear();
-		// DEBUG("onStartStream %d %d %f", deviceNumInputs, deviceNumOutputs, deviceSampleRate);
-	}
-
-	void onStopStream() override {
-		deviceNumInputs = 0;
-		deviceNumOutputs = 0;
-		deviceSampleRate = 0.f;
-		engineInputBuffer.clear();
-		engineOutputBuffer.clear();
-		// DEBUG("onStopStream");
-	}
 };
 
 
@@ -432,7 +450,7 @@ struct AudioInterfaceWidget : ModuleWidget {
 
 			AudioWidget* audioWidget = createWidget<AudioWidget>(mm2px(Vec(3.2122073, 14.837339)));
 			audioWidget->box.size = mm2px(Vec(44, 28));
-			audioWidget->setAudioPort(module);
+			audioWidget->setAudioPort(module ? &module->port : NULL);
 			addChild(audioWidget);
 		}
 		else if (NUM_AUDIO_INPUTS == 16 && NUM_AUDIO_OUTPUTS == 16) {
@@ -496,7 +514,7 @@ struct AudioInterfaceWidget : ModuleWidget {
 
 			AudioWidget* audioWidget = createWidget<AudioWidget>(mm2px(Vec(2.57, 14.839)));
 			audioWidget->box.size = mm2px(Vec(91.382, 28.0));
-			audioWidget->setAudioPort(module);
+			audioWidget->setAudioPort(module ? &module->port : NULL);
 			addChild(audioWidget);
 		}
 		else if (NUM_AUDIO_INPUTS == 2 && NUM_AUDIO_OUTPUTS == 2) {
@@ -530,7 +548,7 @@ struct AudioInterfaceWidget : ModuleWidget {
 
 			AudioDeviceWidget* audioWidget = createWidget<AudioDeviceWidget>(mm2px(Vec(2.135, 14.259)));
 			audioWidget->box.size = mm2px(Vec(21.128, 6.725));
-			audioWidget->setAudioPort(module);
+			audioWidget->setAudioPort(module ? &module->port : NULL);
 			// Adjust deviceChoice position
 			audioWidget->deviceChoice->textOffset = Vec(6, 14);
 			addChild(audioWidget);