Use dsp:: namespace. Automatically include dsp headers in rack.hpp.

include/dsp/functions.hpp → include/dsp/common.hpp

@@ -3,6 +3,11 @@
 
 
 namespace rack {
+namespace dsp {
+
+
+static const float FREQ_C4 = 261.6256f;
+static const float FREQ_A4 = 440.0000f;
 
 
 /** The normalized sinc function. https://en.wikipedia.org/wiki/Sinc_function */
@@ -42,13 +47,14 @@ inline float exponentialBipolar(float b, float x) {
 	return (std::pow(b, x) - std::pow(b, -x)) / a;
 }
 
-inline float gainToDb(float gain) {
-	return std::log10(gain) * 20.f;
+inline float amplitudeToDb(float amp) {
+	return std::log10(amp) * 20.f;
 }
 
-inline float dbToGain(float db) {
+inline float dbToAmplitude(float db) {
 	return std::pow(10.f, db / 20.f);
 }
 
 
+} // namespace dsp
 } // namespace rack

include/dsp/decimator.hpp

@@ -1,2 +0,0 @@
-#pragma once
-#include "resampler.hpp"

include/dsp/digital.hpp

@@ -1,8 +1,9 @@
 #pragma once
-#include "math.hpp"
+#include "dsp/common.hpp"
 
 
 namespace rack {
+namespace dsp {
 
 
 /** Turns HIGH when value reaches 1.f, turns LOW when value reaches 0.f. */
@@ -102,4 +103,5 @@ struct PulseGenerator {
 };
 
 
+} // namespace dsp
 } // namespace rack

include/dsp/filter.hpp

@@ -1,8 +1,10 @@
 #pragma once
-#include "math.hpp"
+#include "dsp/common.hpp"
 
 
 namespace rack {
+namespace dsp {
+
 
 struct RCFilter {
 	float c = 0.f;
@@ -81,4 +83,5 @@ struct ExponentialFilter {
 };
 
 
+} // namespace dsp
 } // namespace rack

include/dsp/fir.hpp

@@ -1,9 +1,11 @@
 #pragma once
-#include "dsp/functions.hpp"
+#include "dsp/common.hpp"
 #include <pffft.h>
 
 
 namespace rack {
+namespace dsp {
+
 
 /** Performs a direct sum convolution */
 inline float convolveNaive(const float *in, const float *kernel, int len) {
@@ -145,4 +147,5 @@ struct RealTimeConvolver {
 };
 
 
+} // namespace dsp
 } // namespace rack

include/dsp/frame.hpp

@@ -1,8 +1,10 @@
 #pragma once
-#include <stdlib.h>
+#include "dsp/common.hpp"
 
 
 namespace rack {
+namespace dsp {
+
 
 /** Useful for storing arrays of samples in ring buffers and casting them to `float*` to be used by interleaved processors, like SampleRateConverter */
 template <size_t CHANNELS>
@@ -10,4 +12,6 @@ struct Frame {
 	float samples[CHANNELS];
 };
 
+
+} // namespace dsp
 } // namespace rack

include/dsp/minblep.hpp

@@ -1,8 +1,10 @@
 #pragma once
-#include "math.hpp"
+#include "dsp/common.hpp"
 
 
 namespace rack {
+namespace dsp {
+
 
 // Pre-made minBLEP samples in minBLEP.cpp
 extern const float minblep_16_32[];
@@ -33,4 +35,6 @@ struct MinBLEP {
 	}
 };
 
+
+} // namespace dsp
 } // namespace rack

include/dsp/ode.hpp

@@ -1,8 +1,10 @@
 #pragma once
+#include "dsp/common.hpp"
 
 
 namespace rack {
-namespace ode {
+namespace dsp {
+
 
 /** The callback function `f` in each of these stepping functions must have the signature
 
@@ -84,5 +86,6 @@ void stepRK4(float t, float dt, float x[], int len, F f) {
 	}
 }
 
-} // namespace ode
+
+} // namespace dsp
 } // namespace rack

include/dsp/resampler.hpp

@@ -1,13 +1,16 @@
 #pragma once
-#include "frame.hpp"
-#include "ringbuffer.hpp"
-#include "fir.hpp"
+#include "dsp/common.hpp"
+#include "dsp/frame.hpp"
+#include "dsp/ringbuffer.hpp"
+#include "dsp/fir.hpp"
 #include <assert.h>
 #include <string.h>
 #include <speex/speex_resampler.h>
 
 
 namespace rack {
+namespace dsp {
+
 
 template<int CHANNELS>
 struct SampleRateConverter {
@@ -170,4 +173,5 @@ struct Upsampler {
 };
 
 
+} // namespace dsp
 } // namespace rack

include/dsp/ringbuffer.hpp

@@ -1,9 +1,11 @@
 #pragma once
-#include "common.hpp"
+#include "dsp/common.hpp"
 #include <string.h>
 
 
 namespace rack {
+namespace dsp {
+
 
 /** A simple cyclic buffer.
 S must be a power of 2.
@@ -196,4 +198,6 @@ struct AppleRingBuffer {
 	}
 };
 
+
+} // namespace dsp
 } // namespace rack

include/dsp/samplerate.hpp

@@ -1,2 +0,0 @@
-#pragma once
-#include "resampler.hpp"

include/dsp/vumeter.hpp

@@ -1,8 +1,9 @@
 #pragma once
-#include "math.hpp"
+#include "dsp/common.hpp"
 
 
 namespace rack {
+namespace dsp {
 
 
 struct VUMeter {
@@ -28,4 +29,5 @@ struct VUMeter {
 };
 
 
+} // namespace dsp
 } // namespace rack

include/rack.hpp

@@ -80,6 +80,16 @@
 #include "plugin/Model.hpp"
 #include "plugin/callbacks.hpp"
 
+#include "dsp/common.hpp"
+#include "dsp/digital.hpp"
+#include "dsp/filter.hpp"
+#include "dsp/fir.hpp"
+#include "dsp/frame.hpp"
+#include "dsp/minblep.hpp"
+#include "dsp/ode.hpp"
+#include "dsp/resampler.hpp"
+#include "dsp/ringbuffer.hpp"
+#include "dsp/vumeter.hpp"
 
 namespace rack {
 

include/rack0.hpp

@@ -147,4 +147,19 @@ DEPRECATED inline float engineGetSampleTime() {
 	return context()->engine->getSampleTime();
 }
 
+////////////////////
+// dsp
+////////////////////
+
+using namespace dsp;
+
+inline float gainToDb(float gain) {
+	return dsp::amplitudeToDb(gain);
+}
+
+inline float dbToGain(float db) {
+	return dsp::dbToAmplitude(db);
+}
+
+
 } // namespace rack

src/Core/AudioInterface.cpp

@@ -1,7 +1,5 @@
 #include "Core.hpp"
 #include "audio.hpp"
-#include "dsp/resampler.hpp"
-#include "dsp/ringbuffer.hpp"
 #include <mutex>
 #include <chrono>
 #include <thread>
@@ -23,9 +21,9 @@ struct AudioInterfaceIO : audio::IO {
 	std::mutex audioMutex;
 	std::condition_variable audioCv;
 	// Audio thread produces, engine thread consumes
-	DoubleRingBuffer<Frame<AUDIO_INPUTS>, (1<<15)> inputBuffer;
+	dsp::DoubleRingBuffer<dsp::Frame<AUDIO_INPUTS>, (1<<15)> inputBuffer;
 	// Audio thread consumes, engine thread produces
-	DoubleRingBuffer<Frame<AUDIO_OUTPUTS>, (1<<15)> outputBuffer;
+	dsp::DoubleRingBuffer<dsp::Frame<AUDIO_OUTPUTS>, (1<<15)> outputBuffer;
 	bool active = false;
 
 	~AudioInterfaceIO() {
@@ -46,7 +44,7 @@ struct AudioInterfaceIO : audio::IO {
 			for (int i = 0; i < frames; i++) {
 				if (inputBuffer.full())
 					break;
-				Frame<AUDIO_INPUTS> inputFrame;
+				dsp::Frame<AUDIO_INPUTS> inputFrame;
 				memset(&inputFrame, 0, sizeof(inputFrame));
 				memcpy(&inputFrame, &input[numInputs * i], numInputs * sizeof(float));
 				inputBuffer.push(inputFrame);
@@ -62,7 +60,7 @@ struct AudioInterfaceIO : audio::IO {
 			if (audioCv.wait_for(lock, timeout, cond)) {
 				// Consume audio block
 				for (int i = 0; i < frames; i++) {
-					Frame<AUDIO_OUTPUTS> f = outputBuffer.shift();
+					dsp::Frame<AUDIO_OUTPUTS> f = outputBuffer.shift();
 					for (int j = 0; j < numOutputs; j++) {
 						output[numOutputs*i + j] = clamp(f.samples[j], -1.f, 1.f);
 					}
@@ -112,12 +110,12 @@ struct AudioInterface : Module {
 	int lastNumOutputs = -1;
 	int lastNumInputs = -1;
 
-	SampleRateConverter<AUDIO_INPUTS> inputSrc;
-	SampleRateConverter<AUDIO_OUTPUTS> outputSrc;
+	dsp::SampleRateConverter<AUDIO_INPUTS> inputSrc;
+	dsp::SampleRateConverter<AUDIO_OUTPUTS> outputSrc;
 
 	// in rack's sample rate
-	DoubleRingBuffer<Frame<AUDIO_INPUTS>, 16> inputBuffer;
-	DoubleRingBuffer<Frame<AUDIO_OUTPUTS>, 16> outputBuffer;
+	dsp::DoubleRingBuffer<dsp::Frame<AUDIO_INPUTS>, 16> inputBuffer;
+	dsp::DoubleRingBuffer<dsp::Frame<AUDIO_OUTPUTS>, 16> outputBuffer;
 
 	AudioInterface() {
 		setup(NUM_PARAMS, NUM_INPUTS, NUM_OUTPUTS, NUM_LIGHTS);
@@ -177,7 +175,7 @@ void AudioInterface::step() {
 	}
 
 	// Take input from buffer
-	Frame<AUDIO_INPUTS> inputFrame;
+	dsp::Frame<AUDIO_INPUTS> inputFrame;
 	if (!inputBuffer.empty()) {
 		inputFrame = inputBuffer.shift();
 	}
@@ -195,7 +193,7 @@ void AudioInterface::step() {
 	if (audioIO.active && audioIO.numOutputs > 0) {
 		// Get and push output SRC frame
 		if (!outputBuffer.full()) {
-			Frame<AUDIO_OUTPUTS> outputFrame;
+			dsp::Frame<AUDIO_OUTPUTS> outputFrame;
 			for (int i = 0; i < AUDIO_OUTPUTS; i++) {
 				outputFrame.samples[i] = inputs[AUDIO_INPUT + i].getVoltage() / 10.f;
 			}

src/Core/MIDICCToCVInterface.cpp

@@ -1,6 +1,5 @@
 #include "Core.hpp"
 #include "midi.hpp"
-#include "dsp/filter.hpp"
 
 
 struct MIDICCToCVInterface : Module {
@@ -20,7 +19,7 @@ struct MIDICCToCVInterface : Module {
 
 	midi::InputQueue midiInput;
 	int8_t ccs[128];
-	ExponentialFilter ccFilters[16];
+	dsp::ExponentialFilter ccFilters[16];
 
 	int learningId = -1;
 	int learnedCcs[16] = {};

src/Core/MIDIToCVInterface.cpp

@@ -1,7 +1,5 @@
 #include "Core.hpp"
 #include "midi.hpp"
-#include "dsp/digital.hpp"
-#include "dsp/filter.hpp"
 
 #include <algorithm>
 
@@ -35,14 +33,14 @@ struct MIDIToCVInterface : Module {
 	midi::InputQueue midiInput;
 
 	uint8_t mod = 0;
-	ExponentialFilter modFilter;
+	dsp::ExponentialFilter modFilter;
 	uint16_t pitch = 8192;
-	ExponentialFilter pitchFilter;
-	PulseGenerator retriggerPulse;
-	PulseGenerator clockPulses[2];
-	PulseGenerator startPulse;
-	PulseGenerator stopPulse;
-	PulseGenerator continuePulse;
+	dsp::ExponentialFilter pitchFilter;
+	dsp::PulseGenerator retriggerPulse;
+	dsp::PulseGenerator clockPulses[2];
+	dsp::PulseGenerator startPulse;
+	dsp::PulseGenerator stopPulse;
+	dsp::PulseGenerator continuePulse;
 	int clock = 0;
 	int divisions[2];
 

src/Core/QuadMIDIToCVInterface.cpp

@@ -1,6 +1,5 @@
 #include "Core.hpp"
 #include "midi.hpp"
-#include "dsp/digital.hpp"
 
 #include <algorithm>