1 files changed, 152 insertions, 112 deletions
diff --git a/audio/softsynth/mt32/AReverbModel.cpp b/audio/softsynth/mt32/AReverbModel.cpp
index 4ee6c87943..595b286a31 100644
--- a/audio/softsynth/mt32/AReverbModel.cpp
+++ b/audio/softsynth/mt32/AReverbModel.cpp
@@ -1,5 +1,5 @@
 /* Copyright (C) 2003, 2004, 2005, 2006, 2008, 2009 Dean Beeler, Jerome Fisher
- * Copyright (C) 2011 Dean Beeler, Jerome Fisher, Sergey V. Mikayev
+ * Copyright (C) 2011, 2012 Dean Beeler, Jerome Fisher, Sergey V. Mikayev
  *
  *  This program is free software: you can redistribute it and/or modify
  *  it under the terms of the GNU Lesser General Public License as published by
@@ -16,64 +16,97 @@
  */
 
 #include "mt32emu.h"
-#include "AReverbModel.h"
-
-using namespace MT32Emu;
-
-// Default reverb settings for modes 0-2
-
-static const unsigned int NUM_ALLPASSES = 6;
-static const unsigned int NUM_DELAYS = 5;
 
-static const Bit32u MODE_0_ALLPASSES[] = {729, 78, 394, 994, 1250, 1889};
-static const Bit32u MODE_0_DELAYS[] = {846, 4, 1819, 778, 346};
-static const float MODE_0_TIMES[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f, 0.6f, 0.9f};
-static const float MODE_0_LEVELS[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f, 0.6f, 1.01575f};
+#if MT32EMU_USE_REVERBMODEL == 1
 
-static const Bit32u MODE_1_ALLPASSES[] = {176, 809, 1324, 1258};
-static const Bit32u MODE_1_DELAYS[] = {2262, 124, 974, 2516, 356};
-static const float MODE_1_TIMES[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f, 0.6f, 0.95f};
-static const float MODE_1_LEVELS[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f, 0.6f, 1.01575f};
-
-static const Bit32u MODE_2_ALLPASSES[] = {78, 729, 994, 389};
-static const Bit32u MODE_2_DELAYS[] = {846, 4, 1819, 778, 346};
-static const float MODE_2_TIMES[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f, 0.6f, 0.7f};
-static const float MODE_2_LEVELS[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f, 0.6f, 0.7f};
-
-const AReverbSettings AReverbModel::REVERB_MODE_0_SETTINGS = {MODE_0_ALLPASSES, MODE_0_DELAYS, MODE_0_TIMES, MODE_0_LEVELS, 0.687770909f, 0.5f, 0.5f};
-const AReverbSettings AReverbModel::REVERB_MODE_1_SETTINGS = {MODE_1_ALLPASSES, MODE_1_DELAYS, MODE_1_TIMES, MODE_1_LEVELS, 0.712025098f, 0.375f, 0.625f};
-const AReverbSettings AReverbModel::REVERB_MODE_2_SETTINGS = {MODE_2_ALLPASSES, MODE_2_DELAYS, MODE_2_TIMES, MODE_2_LEVELS, 0.939522749f, 0.0f, 0.0f};
+#include "AReverbModel.h"
 
-RingBuffer::RingBuffer(Bit32u newsize) {
-	index = 0;
-	size = newsize;
+// Analysing of state of reverb RAM address lines gives exact sizes of the buffers of filters used. This also indicates that
+// the reverb model implemented in the real devices consists of three series allpass filters preceded by a non-feedback comb (or a delay with a LPF)
+// and followed by three parallel comb filters
+
+namespace MT32Emu {
+
+// Because LA-32 chip makes it's output available to process by the Boss chip with a significant delay,
+// the Boss chip puts to the buffer the LA32 dry output when it is ready and performs processing of the _previously_ latched data.
+// Of course, the right way would be to use a dedicated variable for this, but our reverb model is way higher level,
+// so we can simply increase the input buffer size.
+static const Bit32u PROCESS_DELAY = 1;
+
+// Default reverb settings for modes 0-2. These correspond to CM-32L / LAPC-I "new" reverb settings. MT-32 reverb is a bit different.
+// Found by tracing reverb RAM data lines (thanks go to Lord_Nightmare & balrog).
+
+static const Bit32u NUM_ALLPASSES = 3;
+static const Bit32u NUM_COMBS = 4; // Well, actually there are 3 comb filters, but the entrance LPF + delay can be perfectly processed via a comb here.
+
+static const Bit32u MODE_0_ALLPASSES[] = {994, 729, 78};
+static const Bit32u MODE_0_COMBS[] = {705 + PROCESS_DELAY, 2349, 2839, 3632};
+static const Bit32u MODE_0_OUTL[] = {2349, 141, 1960};
+static const Bit32u MODE_0_OUTR[] = {1174, 1570, 145};
+static const Bit32u MODE_0_COMB_FACTOR[] = {0x3C, 0x60, 0x60, 0x60};
+static const Bit32u MODE_0_COMB_FEEDBACK[] = {0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+                                              0x28, 0x48, 0x60, 0x78, 0x80, 0x88, 0x90, 0x98,
+                                              0x28, 0x48, 0x60, 0x78, 0x80, 0x88, 0x90, 0x98,
+                                              0x28, 0x48, 0x60, 0x78, 0x80, 0x88, 0x90, 0x98};
+static const Bit32u MODE_0_LEVELS[] = {10*1, 10*3, 10*5, 10*7, 11*9, 11*12, 11*15, 13*15};
+static const Bit32u MODE_0_LPF_AMP = 6;
+
+static const Bit32u MODE_1_ALLPASSES[] = {1324, 809, 176};
+static const Bit32u MODE_1_COMBS[] = {961 + PROCESS_DELAY, 2619, 3545, 4519};
+static const Bit32u MODE_1_OUTL[] = {2618, 1760, 4518};
+static const Bit32u MODE_1_OUTR[] = {1300, 3532, 2274};
+static const Bit32u MODE_1_COMB_FACTOR[] = {0x30, 0x60, 0x60, 0x60};
+static const Bit32u MODE_1_COMB_FEEDBACK[] = {0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+											  0x28, 0x48, 0x60, 0x70, 0x78, 0x80, 0x90, 0x98,
+											  0x28, 0x48, 0x60, 0x78, 0x80, 0x88, 0x90, 0x98,
+											  0x28, 0x48, 0x60, 0x78, 0x80, 0x88, 0x90, 0x98};
+static const Bit32u MODE_1_LEVELS[] = {10*1, 10*3, 11*5, 11*7, 11*9, 11*12, 11*15, 14*15};
+static const Bit32u MODE_1_LPF_AMP = 6;
+
+static const Bit32u MODE_2_ALLPASSES[] = {969, 644, 157};
+static const Bit32u MODE_2_COMBS[] = {116 + PROCESS_DELAY, 2259, 2839, 3539};
+static const Bit32u MODE_2_OUTL[] = {2259, 718, 1769};
+static const Bit32u MODE_2_OUTR[] = {1136, 2128, 1};
+static const Bit32u MODE_2_COMB_FACTOR[] = {0, 0x20, 0x20, 0x20};
+static const Bit32u MODE_2_COMB_FEEDBACK[] = {0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+                                              0x30, 0x58, 0x78, 0x88, 0xA0, 0xB8, 0xC0, 0xD0,
+                                              0x30, 0x58, 0x78, 0x88, 0xA0, 0xB8, 0xC0, 0xD0,
+                                              0x30, 0x58, 0x78, 0x88, 0xA0, 0xB8, 0xC0, 0xD0};
+static const Bit32u MODE_2_LEVELS[] = {10*1, 10*3, 11*5, 11*7, 11*9, 11*12, 12*15, 14*15};
+static const Bit32u MODE_2_LPF_AMP = 8;
+
+static const AReverbSettings REVERB_MODE_0_SETTINGS = {MODE_0_ALLPASSES, MODE_0_COMBS, MODE_0_OUTL, MODE_0_OUTR, MODE_0_COMB_FACTOR, MODE_0_COMB_FEEDBACK, MODE_0_LEVELS, MODE_0_LPF_AMP};
+static const AReverbSettings REVERB_MODE_1_SETTINGS = {MODE_1_ALLPASSES, MODE_1_COMBS, MODE_1_OUTL, MODE_1_OUTR, MODE_1_COMB_FACTOR, MODE_1_COMB_FEEDBACK, MODE_1_LEVELS, MODE_1_LPF_AMP};
+static const AReverbSettings REVERB_MODE_2_SETTINGS = {MODE_2_ALLPASSES, MODE_2_COMBS, MODE_2_OUTL, MODE_2_OUTR, MODE_2_COMB_FACTOR, MODE_2_COMB_FEEDBACK, MODE_2_LEVELS, MODE_2_LPF_AMP};
+
+static const AReverbSettings * const REVERB_SETTINGS[] = {&REVERB_MODE_0_SETTINGS, &REVERB_MODE_1_SETTINGS, &REVERB_MODE_2_SETTINGS, &REVERB_MODE_0_SETTINGS};
+
+RingBuffer::RingBuffer(const Bit32u newsize) : size(newsize), index(0) {
 	buffer = new float[size];
 }
 
 RingBuffer::~RingBuffer() {
 	delete[] buffer;
 	buffer = NULL;
-	size = 0;
 }
 
 float RingBuffer::next() {
-	index++;
-	if (index >= size) {
+	if (++index >= size) {
 		index = 0;
 	}
 	return buffer[index];
 }
 
-bool RingBuffer::isEmpty() {
+bool RingBuffer::isEmpty() const {
 	if (buffer == NULL) return true;
 
 	float *buf = buffer;
-	float total = 0;
+	float max = 0.001f;
 	for (Bit32u i = 0; i < size; i++) {
-		total += (*buf < 0 ? -*buf : *buf);
+		if ((*buf < -max) || (*buf > max)) return false;
 		buf++;
 	}
-	return ((total / size) < .0002 ? true : false);
+	return true;
 }
 
 void RingBuffer::mute() {
@@ -83,59 +116,66 @@ void RingBuffer::mute() {
 	}
 }
 
-AllpassFilter::AllpassFilter(Bit32u useSize) : RingBuffer(useSize) {
-}
-
-Delay::Delay(Bit32u useSize) : RingBuffer(useSize) {
-}
+AllpassFilter::AllpassFilter(const Bit32u useSize) : RingBuffer(useSize) {}
 
-float AllpassFilter::process(float in) {
-	// This model corresponds to the allpass filter implementation in the real CM-32L device
+float AllpassFilter::process(const float in) {
+	// This model corresponds to the allpass filter implementation of the real CM-32L device
 	// found from sample analysis
 
-	float out;
-
-	out = next();
+	const float bufferOut = next();
 
 	// store input - feedback / 2
-	buffer[index] = in - 0.5f * out;
+	buffer[index] = in - 0.5f * bufferOut;
 
 	// return buffer output + feedforward / 2
-	return out + 0.5f * buffer[index];
+	return bufferOut + 0.5f * buffer[index];
 }
 
-float Delay::process(float in) {
-	// Implements a very simple delay
+CombFilter::CombFilter(const Bit32u useSize) : RingBuffer(useSize) {}
 
-	float out;
+void CombFilter::process(const float in) {
+	// This model corresponds to the comb filter implementation of the real CM-32L device
+	// found from sample analysis
+
+	// the previously stored value
+	float last = buffer[index];
 
-	out = next();
+	// prepare input + feedback
+	float filterIn = in + next() * feedbackFactor;
 
-	// store input
-	buffer[index] = in;
+	// store input + feedback processed by a low-pass filter
+	buffer[index] = filterFactor * last - filterIn;
+}
 
-	// return buffer output
-	return out;
+float CombFilter::getOutputAt(const Bit32u outIndex) const {
+	return buffer[(size + index - outIndex) % size];
 }
 
-AReverbModel::AReverbModel(const AReverbSettings *useSettings) : allpasses(NULL), delays(NULL), currentSettings(useSettings) {
+void CombFilter::setFeedbackFactor(const float useFeedbackFactor) {
+	feedbackFactor = useFeedbackFactor;
 }
 
+void CombFilter::setFilterFactor(const float useFilterFactor) {
+	filterFactor = useFilterFactor;
+}
+
+AReverbModel::AReverbModel(const ReverbMode mode) : allpasses(NULL), combs(NULL), currentSettings(*REVERB_SETTINGS[mode]) {}
+
 AReverbModel::~AReverbModel() {
 	close();
 }
 
-void AReverbModel::open(unsigned int /*sampleRate*/) {
-	// FIXME: filter sizes must be multiplied by sample rate to 32000Hz ratio
-	// IIR filter values depend on sample rate as well
+void AReverbModel::open() {
 	allpasses = new AllpassFilter*[NUM_ALLPASSES];
 	for (Bit32u i = 0; i < NUM_ALLPASSES; i++) {
-		allpasses[i] = new AllpassFilter(currentSettings->allpassSizes[i]);
+		allpasses[i] = new AllpassFilter(currentSettings.allpassSizes[i]);
 	}
-	delays = new Delay*[NUM_DELAYS];
-	for (Bit32u i = 0; i < NUM_DELAYS; i++) {
-		delays[i] = new Delay(currentSettings->delaySizes[i]);
+	combs = new CombFilter*[NUM_COMBS];
+	for (Bit32u i = 0; i < NUM_COMBS; i++) {
+		combs[i] = new CombFilter(currentSettings.combSizes[i]);
+		combs[i]->setFilterFactor(currentSettings.filterFactor[i] / 256.0f);
 	}
+	lpfAmp = currentSettings.lpfAmp / 16.0f;
 	mute();
 }
 
@@ -150,84 +190,80 @@ void AReverbModel::close() {
 		delete[] allpasses;
 		allpasses = NULL;
 	}
-	if (delays != NULL) {
-		for (Bit32u i = 0; i < NUM_DELAYS; i++) {
-			if (delays[i] != NULL) {
-				delete delays[i];
-				delays[i] = NULL;
+	if (combs != NULL) {
+		for (Bit32u i = 0; i < NUM_COMBS; i++) {
+			if (combs[i] != NULL) {
+				delete combs[i];
+				combs[i] = NULL;
 			}
 		}
-		delete[] delays;
-		delays = NULL;
+		delete[] combs;
+		combs = NULL;
 	}
 }
 
 void AReverbModel::mute() {
+	if (allpasses == NULL || combs == NULL) return;
 	for (Bit32u i = 0; i < NUM_ALLPASSES; i++) {
 		allpasses[i]->mute();
 	}
-	for (Bit32u i = 0; i < NUM_DELAYS; i++) {
-		delays[i]->mute();
+	for (Bit32u i = 0; i < NUM_COMBS; i++) {
+		combs[i]->mute();
 	}
-	filterhist1 = 0;
-	filterhist2 = 0;
-	combhist = 0;
 }
 
 void AReverbModel::setParameters(Bit8u time, Bit8u level) {
 // FIXME: wetLevel definitely needs ramping when changed
 // Although, most games don't set reverb level during MIDI playback
-	decayTime = currentSettings->decayTimes[time];
-	wetLevel = currentSettings->wetLevels[level];
+	if (combs == NULL) return;
+	level &= 7;
+	time &= 7;
+	for (Bit32u i = 0; i < NUM_COMBS; i++) {
+		combs[i]->setFeedbackFactor(currentSettings.decayTimes[(i << 3) + time] / 256.0f);
+	}
+	wetLevel = (level == 0 && time == 0) ? 0.0f : 0.5f * lpfAmp * currentSettings.wetLevels[level] / 256.0f;
 }
 
 bool AReverbModel::isActive() const {
-	bool bActive = false;
 	for (Bit32u i = 0; i < NUM_ALLPASSES; i++) {
-		bActive |= !allpasses[i]->isEmpty();
+		if (!allpasses[i]->isEmpty()) return true;
 	}
-	for (Bit32u i = 0; i < NUM_DELAYS; i++) {
-		bActive |= !delays[i]->isEmpty();
+	for (Bit32u i = 0; i < NUM_COMBS; i++) {
+		if (!combs[i]->isEmpty()) return true;
 	}
-	return bActive;
+	return false;
 }
 
 void AReverbModel::process(const float *inLeft, const float *inRight, float *outLeft, float *outRight, unsigned long numSamples) {
-// Three series allpass filters followed by a delay, fourth allpass filter and another delay
-	float dry, link, outL1, outL2, outR1, outR2;
+	float dry, link, outL1;
 
 	for (unsigned long i = 0; i < numSamples; i++) {
-		dry = *inLeft + *inRight;
+		dry = wetLevel * (*inLeft + *inRight);
 
-		// Implementation of 2-stage IIR single-pole low-pass filter
-		// found at the entrance of reverb processing on real devices
-		filterhist1 += (dry - filterhist1) * currentSettings->filtVal;
-		filterhist2 += (filterhist1 - filterhist2) * currentSettings->filtVal;
+		// Get the last stored sample before processing in order not to loose it
+		link = combs[0]->getOutputAt(currentSettings.combSizes[0] - 1);
 
-		link = allpasses[0]->process(-filterhist2);
-		link = allpasses[1]->process(link);
+		combs[0]->process(-dry);
 
-		// this implements a comb filter cross-linked with the fourth allpass filter
-		link += combhist * decayTime;
+		link = allpasses[0]->process(link);
+		link = allpasses[1]->process(link);
 		link = allpasses[2]->process(link);
-		link = delays[0]->process(link);
-		outL1 = link;
-		link = allpasses[3]->process(link);
-		link = delays[1]->process(link);
-		outR1 = link;
-		link = allpasses[4]->process(link);
-		link = delays[2]->process(link);
-		outL2 = link;
-		link = allpasses[5]->process(link);
-		link = delays[3]->process(link);
-		outR2 = link;
-		link = delays[4]->process(link);
-
-		// comb filter end point
-		combhist = combhist * currentSettings->damp1 + link * currentSettings->damp2;
-
-		*outLeft = (outL1 + outL2) * wetLevel;
-		*outRight = (outR1 + outR2) * wetLevel;
+
+		// If the output position is equal to the comb size, get it now in order not to loose it
+		outL1 = 1.5f * combs[1]->getOutputAt(currentSettings.outLPositions[0] - 1);
+
+		combs[1]->process(link);
+		combs[2]->process(link);
+		combs[3]->process(link);
+
+		link = outL1 + 1.5f * combs[2]->getOutputAt(currentSettings.outLPositions[1]);
+		link += combs[3]->getOutputAt(currentSettings.outLPositions[2]);
+		*outLeft = link;
+
+		link = 1.5f * combs[1]->getOutputAt(currentSettings.outRPositions[0]);
+		link += 1.5f * combs[2]->getOutputAt(currentSettings.outRPositions[1]);
+		link += combs[3]->getOutputAt(currentSettings.outRPositions[2]);
+		*outRight = link;
 
 		inLeft++;
 		inRight++;
@@ -235,3 +271,7 @@ void AReverbModel::process(const float *inLeft, const float *inRight, float *out
 		outRight++;
 	}
 }
+
+}
+
+#endif