[Scummvm-cvs-logs] scummvm master -> d9e555afd5932c458f559034c6dca1df346ead4b
bluegr
bluegr at gmail.com
Tue Dec 25 22:59:42 CET 2012
This automated email contains information about 1 new commit which have been
pushed to the 'scummvm' repo located at https://github.com/scummvm/scummvm .
Summary:
d9e555afd5 MT32: Update the MT32 emulator to a newer munt commit
Commit: d9e555afd5932c458f559034c6dca1df346ead4b
https://github.com/scummvm/scummvm/commit/d9e555afd5932c458f559034c6dca1df346ead4b
Author: Filippos Karapetis (bluegr at gmail.com)
Date: 2012-12-25T13:48:25-08:00
Commit Message:
MT32: Update the MT32 emulator to a newer munt commit
Previous munt commit was f969d20 (Nov 15, 2012)
Current munt commit is 84b2819 (Dec 22, 2012)
We are still missing the changes from commit 788f7b1 onwards (Dec 22,
2012). There are bigger ROM access-related changes from that point,
which we'll have to integrate as well.
Changed paths:
A audio/softsynth/mt32/BReverbModel.cpp
A audio/softsynth/mt32/BReverbModel.h
audio/softsynth/mt32/AReverbModel.cpp
audio/softsynth/mt32/AReverbModel.h
audio/softsynth/mt32/DelayReverb.cpp
audio/softsynth/mt32/DelayReverb.h
audio/softsynth/mt32/Synth.cpp
audio/softsynth/mt32/Synth.h
audio/softsynth/mt32/module.mk
audio/softsynth/mt32/mt32emu.h
diff --git a/audio/softsynth/mt32/AReverbModel.cpp b/audio/softsynth/mt32/AReverbModel.cpp
index 151f6c2..ec24394 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"
-
-namespace MT32Emu {
-
-// Default reverb settings for modes 0-2
-static const unsigned int NUM_ALLPASSES = 6;
-static const unsigned int NUM_DELAYS = 5;
+#if MT32EMU_USE_REVERBMODEL == 1
-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};
-
-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};
+#include "AReverbModel.h"
-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};
+// 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
-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};
+namespace MT32Emu {
-RingBuffer::RingBuffer(Bit32u newsize) {
- index = 0;
- size = newsize;
+// 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[] = {0, 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[] = {0, 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[] = {0, 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() {
@@ -83,44 +116,51 @@ 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
- out = next();
+ // the previously stored value
+ float last = buffer[index];
- // store input
- buffer[index] = in;
+ // prepare input + feedback
+ float filterIn = in + next() * feedbackFactor;
- // return buffer output
- return out;
+ // store input + feedback processed by a low-pass filter
+ buffer[index] = filterFactor * last - filterIn;
}
-AReverbModel::AReverbModel(const AReverbSettings *useSettings) : allpasses(NULL), delays(NULL), currentSettings(useSettings) {
+float CombFilter::getOutputAt(const Bit32u outIndex) const {
+ return buffer[(size + index - outIndex) % size];
}
+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();
}
@@ -130,12 +170,14 @@ void AReverbModel::open(unsigned int /*sampleRate*/) {
// IIR filter values depend on sample rate as well
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 +192,78 @@ 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;
+ for (Bit32u i = 0; i < NUM_COMBS; i++) {
+ combs[i]->setFeedbackFactor(currentSettings.decayTimes[(i << 3) + (time & 7)] / 256.0f);
+ }
+ wetLevel = 0.5f * lpfAmp * currentSettings.wetLevels[(level & 7)] / 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++;
@@ -237,3 +273,5 @@ void AReverbModel::process(const float *inLeft, const float *inRight, float *out
}
}
+
+#endif
diff --git a/audio/softsynth/mt32/AReverbModel.h b/audio/softsynth/mt32/AReverbModel.h
index 3fae08c..d70e9ee 100644
--- a/audio/softsynth/mt32/AReverbModel.h
+++ b/audio/softsynth/mt32/AReverbModel.h
@@ -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
@@ -21,66 +21,67 @@
namespace MT32Emu {
struct AReverbSettings {
- const Bit32u *allpassSizes;
- const Bit32u *delaySizes;
- const float *decayTimes;
- const float *wetLevels;
- float filtVal;
- float damp1;
- float damp2;
+ const Bit32u * const allpassSizes;
+ const Bit32u * const combSizes;
+ const Bit32u * const outLPositions;
+ const Bit32u * const outRPositions;
+ const Bit32u * const filterFactor;
+ const Bit32u * const decayTimes;
+ const Bit32u * const wetLevels;
+ const Bit32u lpfAmp;
};
class RingBuffer {
protected:
float *buffer;
- Bit32u size;
+ const Bit32u size;
Bit32u index;
+
public:
- RingBuffer(Bit32u size);
+ RingBuffer(const Bit32u size);
virtual ~RingBuffer();
float next();
- bool isEmpty();
+ bool isEmpty() const;
void mute();
};
class AllpassFilter : public RingBuffer {
public:
- AllpassFilter(Bit32u size);
- float process(float in);
+ AllpassFilter(const Bit32u size);
+ float process(const float in);
};
-class Delay : public RingBuffer {
+class CombFilter : public RingBuffer {
+ float feedbackFactor;
+ float filterFactor;
+
public:
- Delay(Bit32u size);
- float process(float in);
+ CombFilter(const Bit32u size);
+ void process(const float in);
+ float getOutputAt(const Bit32u outIndex) const;
+ void setFeedbackFactor(const float useFeedbackFactor);
+ void setFilterFactor(const float useFilterFactor);
};
class AReverbModel : public ReverbModel {
AllpassFilter **allpasses;
- Delay **delays;
+ CombFilter **combs;
- const AReverbSettings *currentSettings;
- float decayTime;
+ const AReverbSettings ¤tSettings;
+ float lpfAmp;
float wetLevel;
- float filterhist1, filterhist2;
- float combhist;
void mute();
+
public:
- AReverbModel(const AReverbSettings *newSettings);
+ AReverbModel(const ReverbMode mode);
~AReverbModel();
void open(unsigned int sampleRate);
void close();
void setParameters(Bit8u time, Bit8u level);
void process(const float *inLeft, const float *inRight, float *outLeft, float *outRight, unsigned long numSamples);
bool isActive() const;
-
- static const AReverbSettings REVERB_MODE_0_SETTINGS;
- static const AReverbSettings REVERB_MODE_1_SETTINGS;
- static const AReverbSettings REVERB_MODE_2_SETTINGS;
};
-// Default reverb settings for modes 0-2
-
}
#endif
diff --git a/audio/softsynth/mt32/BReverbModel.cpp b/audio/softsynth/mt32/BReverbModel.cpp
new file mode 100644
index 0000000..b5f4467
--- /dev/null
+++ b/audio/softsynth/mt32/BReverbModel.cpp
@@ -0,0 +1,395 @@
+/* Copyright (C) 2003, 2004, 2005, 2006, 2008, 2009 Dean Beeler, Jerome Fisher
+ * 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
+ * the Free Software Foundation, either version 2.1 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "mt32emu.h"
+
+#if MT32EMU_USE_REVERBMODEL == 2
+
+#include "BReverbModel.h"
+
+// 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;
+
+static const Bit32u MODE_3_ADDITIONAL_DELAY = 1;
+static const Bit32u MODE_3_FEEDBACK_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 MODE_0_NUMBER_OF_ALLPASSES = 3;
+static const Bit32u MODE_0_ALLPASSES[] = {994, 729, 78};
+static const Bit32u MODE_0_NUMBER_OF_COMBS = 4; // Well, actually there are 3 comb filters, but the entrance LPF + delay can be processed via a hacked comb.
+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[] = {0xA0, 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_DRY_AMP[] = {0xA0, 0xA0, 0xA0, 0xA0, 0xB0, 0xB0, 0xB0, 0xD0};
+static const Bit32u MODE_0_WET_AMP[] = {0x10, 0x30, 0x50, 0x70, 0x90, 0xC0, 0xF0, 0xF0};
+static const Bit32u MODE_0_LPF_AMP = 0x60;
+
+static const Bit32u MODE_1_NUMBER_OF_ALLPASSES = 3;
+static const Bit32u MODE_1_ALLPASSES[] = {1324, 809, 176};
+static const Bit32u MODE_1_NUMBER_OF_COMBS = 4; // Same as for mode 0 above
+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[] = {0x80, 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_DRY_AMP[] = {0xA0, 0xA0, 0xB0, 0xB0, 0xB0, 0xB0, 0xB0, 0xE0};
+static const Bit32u MODE_1_WET_AMP[] = {0x10, 0x30, 0x50, 0x70, 0x90, 0xC0, 0xF0, 0xF0};
+static const Bit32u MODE_1_LPF_AMP = 0x60;
+
+static const Bit32u MODE_2_NUMBER_OF_ALLPASSES = 3;
+static const Bit32u MODE_2_ALLPASSES[] = {969, 644, 157};
+static const Bit32u MODE_2_NUMBER_OF_COMBS = 4; // Same as for mode 0 above
+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_DRY_AMP[] = {0xA0, 0xA0, 0xB0, 0xB0, 0xB0, 0xB0, 0xC0, 0xE0};
+static const Bit32u MODE_2_WET_AMP[] = {0x10, 0x30, 0x50, 0x70, 0x90, 0xC0, 0xF0, 0xF0};
+static const Bit32u MODE_2_LPF_AMP = 0x80;
+
+static const Bit32u MODE_3_NUMBER_OF_ALLPASSES = 0;
+static const Bit32u MODE_3_NUMBER_OF_COMBS = 1;
+static const Bit32u MODE_3_DELAY[] = {16000 + MODE_3_FEEDBACK_DELAY + PROCESS_DELAY + MODE_3_ADDITIONAL_DELAY};
+static const Bit32u MODE_3_OUTL[] = {400, 624, 960, 1488, 2256, 3472, 5280, 8000};
+static const Bit32u MODE_3_OUTR[] = {800, 1248, 1920, 2976, 4512, 6944, 10560, 16000};
+static const Bit32u MODE_3_COMB_FACTOR[] = {0x68};
+static const Bit32u MODE_3_COMB_FEEDBACK[] = {0x68, 0x60};
+static const Bit32u MODE_3_DRY_AMP[] = {0x20, 0x50, 0x50, 0x50, 0x50, 0x50, 0x50, 0x50};
+static const Bit32u MODE_3_WET_AMP[] = {0x18, 0x18, 0x28, 0x40, 0x60, 0x80, 0xA8, 0xF8};
+
+static const BReverbSettings REVERB_MODE_0_SETTINGS = {MODE_0_NUMBER_OF_ALLPASSES, MODE_0_ALLPASSES, MODE_0_NUMBER_OF_COMBS, MODE_0_COMBS, MODE_0_OUTL, MODE_0_OUTR, MODE_0_COMB_FACTOR, MODE_0_COMB_FEEDBACK, MODE_0_DRY_AMP, MODE_0_WET_AMP, MODE_0_LPF_AMP};
+static const BReverbSettings REVERB_MODE_1_SETTINGS = {MODE_1_NUMBER_OF_ALLPASSES, MODE_1_ALLPASSES, MODE_1_NUMBER_OF_COMBS, MODE_1_COMBS, MODE_1_OUTL, MODE_1_OUTR, MODE_1_COMB_FACTOR, MODE_1_COMB_FEEDBACK, MODE_1_DRY_AMP, MODE_1_WET_AMP, MODE_1_LPF_AMP};
+static const BReverbSettings REVERB_MODE_2_SETTINGS = {MODE_2_NUMBER_OF_ALLPASSES, MODE_2_ALLPASSES, MODE_2_NUMBER_OF_COMBS, MODE_2_COMBS, MODE_2_OUTL, MODE_2_OUTR, MODE_2_COMB_FACTOR, MODE_2_COMB_FEEDBACK, MODE_2_DRY_AMP, MODE_2_WET_AMP, MODE_2_LPF_AMP};
+static const BReverbSettings REVERB_MODE_3_SETTINGS = {MODE_3_NUMBER_OF_ALLPASSES, NULL, MODE_3_NUMBER_OF_COMBS, MODE_3_DELAY, MODE_3_OUTL, MODE_3_OUTR, MODE_3_COMB_FACTOR, MODE_3_COMB_FEEDBACK, MODE_3_DRY_AMP, MODE_3_WET_AMP, 0};
+
+static const BReverbSettings * const REVERB_SETTINGS[] = {&REVERB_MODE_0_SETTINGS, &REVERB_MODE_1_SETTINGS, &REVERB_MODE_2_SETTINGS, &REVERB_MODE_3_SETTINGS};
+
+// This algorithm tries to emulate exactly Boss multiplication operation (at least this is what we see on reverb RAM data lines).
+// Also LA32 is suspected to use the similar one to perform PCM interpolation and ring modulation.
+static Bit32s weirdMul(Bit32s a, Bit8u addMask, Bit8u carryMask) {
+ Bit8u mask = 0x80;
+ Bit32s res = 0;
+ for (int i = 0; i < 8; i++) {
+ Bit32s carry = (a < 0) && (mask & carryMask) > 0 ? a & 1 : 0;
+ a >>= 1;
+ res += (mask & addMask) > 0 ? a + carry : 0;
+ mask >>= 1;
+ }
+ return res;
+}
+
+RingBuffer::RingBuffer(Bit32u newsize) : size(newsize), index(0) {
+ buffer = new Bit16s[size];
+}
+
+RingBuffer::~RingBuffer() {
+ delete[] buffer;
+ buffer = NULL;
+}
+
+Bit32s RingBuffer::next() {
+ if (++index >= size) {
+ index = 0;
+ }
+ return buffer[index];
+}
+
+bool RingBuffer::isEmpty() const {
+ if (buffer == NULL) return true;
+
+ Bit16s *buf = buffer;
+ for (Bit32u i = 0; i < size; i++) {
+ if (*buf < -8 || *buf > 8) return false;
+ buf++;
+ }
+ return true;
+}
+
+void RingBuffer::mute() {
+ Bit16s *buf = buffer;
+ for (Bit32u i = 0; i < size; i++) {
+ *buf++ = 0;
+ }
+}
+
+AllpassFilter::AllpassFilter(const Bit32u useSize) : RingBuffer(useSize) {}
+
+Bit32s AllpassFilter::process(const Bit32s in) {
+ // This model corresponds to the allpass filter implementation of the real CM-32L device
+ // found from sample analysis
+
+ Bit16s bufferOut = next();
+
+ // store input - feedback / 2
+ buffer[index] = in - (bufferOut >> 1);
+
+ // return buffer output + feedforward / 2
+ return bufferOut + (buffer[index] >> 1);
+}
+
+CombFilter::CombFilter(const Bit32u useSize, const Bit32u useFilterFactor) : RingBuffer(useSize), filterFactor(useFilterFactor) {}
+
+void CombFilter::process(const Bit32s in) {
+ // This model corresponds to the comb filter implementation of the real CM-32L device
+
+ // the previously stored value
+ Bit32s last = buffer[index];
+
+ // prepare input + feedback
+ Bit32s filterIn = in + weirdMul(next(), feedbackFactor, 0xF0 /* Maybe 0x80 ? */);
+
+ // store input + feedback processed by a low-pass filter
+ buffer[index] = weirdMul(last, filterFactor, 0x40) - filterIn;
+}
+
+Bit32s CombFilter::getOutputAt(const Bit32u outIndex) const {
+ return buffer[(size + index - outIndex) % size];
+}
+
+void CombFilter::setFeedbackFactor(const Bit32u useFeedbackFactor) {
+ feedbackFactor = useFeedbackFactor;
+}
+
+DelayWithLowPassFilter::DelayWithLowPassFilter(const Bit32u useSize, const Bit32u useFilterFactor, const Bit32u useAmp)
+ : CombFilter(useSize, useFilterFactor), amp(useAmp) {}
+
+void DelayWithLowPassFilter::process(const Bit32s in) {
+ // the previously stored value
+ Bit32s last = buffer[index];
+
+ // move to the next index
+ next();
+
+ // low-pass filter process
+ Bit32s lpfOut = weirdMul(last, filterFactor, 0xFF) + in;
+
+ // store lpfOut multiplied by LPF amp factor
+ buffer[index] = weirdMul(lpfOut, amp, 0xFF);
+}
+
+TapDelayCombFilter::TapDelayCombFilter(const Bit32u useSize, const Bit32u useFilterFactor) : CombFilter(useSize, useFilterFactor) {}
+
+void TapDelayCombFilter::process(const Bit32s in) {
+ // the previously stored value
+ Bit32s last = buffer[index];
+
+ // move to the next index
+ next();
+
+ // prepare input + feedback
+ // Actually, the size of the filter varies with the TIME parameter, the feedback sample is taken from the position just below the right output
+ Bit32s filterIn = in + weirdMul(getOutputAt(outR + MODE_3_FEEDBACK_DELAY), feedbackFactor, 0xF0);
+
+ // store input + feedback processed by a low-pass filter
+ buffer[index] = weirdMul(last, filterFactor, 0xF0) - filterIn;
+}
+
+Bit32s TapDelayCombFilter::getLeftOutput() const {
+ return getOutputAt(outL + PROCESS_DELAY + MODE_3_ADDITIONAL_DELAY);
+}
+
+Bit32s TapDelayCombFilter::getRightOutput() const {
+ return getOutputAt(outR + PROCESS_DELAY + MODE_3_ADDITIONAL_DELAY);
+}
+
+void TapDelayCombFilter::setOutputPositions(const Bit32u useOutL, const Bit32u useOutR) {
+ outL = useOutL;
+ outR = useOutR;
+}
+
+BReverbModel::BReverbModel(const ReverbMode mode)
+ : allpasses(NULL), combs(NULL), currentSettings(*REVERB_SETTINGS[mode]), tapDelayMode(mode == REVERB_MODE_TAP_DELAY) {}
+
+BReverbModel::~BReverbModel() {
+ close();
+}
+
+void BReverbModel::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
+ if (currentSettings.numberOfAllpasses > 0) {
+ allpasses = new AllpassFilter*[currentSettings.numberOfAllpasses];
+ for (Bit32u i = 0; i < currentSettings.numberOfAllpasses; i++) {
+ allpasses[i] = new AllpassFilter(currentSettings.allpassSizes[i]);
+ }
+ }
+ combs = new CombFilter*[currentSettings.numberOfCombs];
+ if (tapDelayMode) {
+ *combs = new TapDelayCombFilter(*currentSettings.combSizes, *currentSettings.filterFactors);
+ } else {
+ combs[0] = new DelayWithLowPassFilter(currentSettings.combSizes[0], currentSettings.filterFactors[0], currentSettings.lpfAmp);
+ for (Bit32u i = 1; i < currentSettings.numberOfCombs; i++) {
+ combs[i] = new CombFilter(currentSettings.combSizes[i], currentSettings.filterFactors[i]);
+ }
+ }
+ mute();
+}
+
+void BReverbModel::close() {
+ if (allpasses != NULL) {
+ for (Bit32u i = 0; i < currentSettings.numberOfAllpasses; i++) {
+ if (allpasses[i] != NULL) {
+ delete allpasses[i];
+ allpasses[i] = NULL;
+ }
+ }
+ delete[] allpasses;
+ allpasses = NULL;
+ }
+ if (combs != NULL) {
+ for (Bit32u i = 0; i < currentSettings.numberOfCombs; i++) {
+ if (combs[i] != NULL) {
+ delete combs[i];
+ combs[i] = NULL;
+ }
+ }
+ delete[] combs;
+ combs = NULL;
+ }
+}
+
+void BReverbModel::mute() {
+ if (allpasses != NULL) {
+ for (Bit32u i = 0; i < currentSettings.numberOfAllpasses; i++) {
+ allpasses[i]->mute();
+ }
+ }
+ if (combs != NULL) {
+ for (Bit32u i = 0; i < currentSettings.numberOfCombs; i++) {
+ combs[i]->mute();
+ }
+ }
+}
+
+void BReverbModel::setParameters(Bit8u time, Bit8u level) {
+ if (combs == NULL) return;
+ level &= 7;
+ time &= 7;
+ if (tapDelayMode) {
+ TapDelayCombFilter *comb = static_cast<TapDelayCombFilter *> (*combs);
+ comb->setOutputPositions(currentSettings.outLPositions[time], currentSettings.outRPositions[time & 7]);
+ comb->setFeedbackFactor(currentSettings.feedbackFactors[((level < 3) || (time < 6)) ? 0 : 1]);
+ } else {
+ for (Bit32u i = 0; i < currentSettings.numberOfCombs; i++) {
+ combs[i]->setFeedbackFactor(currentSettings.feedbackFactors[(i << 3) + time]);
+ }
+ }
+ if (time == 0 && level == 0) {
+ dryAmp = wetLevel = 0;
+ } else {
+ dryAmp = currentSettings.dryAmps[level];
+ wetLevel = currentSettings.wetLevels[level];
+ }
+}
+
+bool BReverbModel::isActive() const {
+ for (Bit32u i = 0; i < currentSettings.numberOfAllpasses; i++) {
+ if (!allpasses[i]->isEmpty()) return true;
+ }
+ for (Bit32u i = 0; i < currentSettings.numberOfCombs; i++) {
+ if (!combs[i]->isEmpty()) return true;
+ }
+ return false;
+}
+
+void BReverbModel::process(const float *inLeft, const float *inRight, float *outLeft, float *outRight, unsigned long numSamples) {
+ Bit32s dry, link, outL1, outR1;
+
+ for (unsigned long i = 0; i < numSamples; i++) {
+ if (tapDelayMode) {
+ dry = Bit32s(*inLeft * 8192.0f) + Bit32s(*inRight * 8192.0f);
+ } else {
+ dry = Bit32s(*inLeft * 8192.0f) / 2 + Bit32s(*inRight * 8192.0f) / 2;
+ }
+
+ // Looks like dryAmp doesn't change in MT-32 but it does in CM-32L / LAPC-I
+ dry = weirdMul(dry, dryAmp, 0xFF);
+
+ if (tapDelayMode) {
+ TapDelayCombFilter *comb = static_cast<TapDelayCombFilter *> (*combs);
+ comb->process(dry);
+ *outLeft = weirdMul(comb->getLeftOutput(), wetLevel, 0xFF) / 8192.0f;
+ *outRight = weirdMul(comb->getRightOutput(), wetLevel, 0xFF) / 8192.0f;
+ } else {
+ // Get the last stored sample before processing in order not to loose it
+ link = combs[0]->getOutputAt(currentSettings.combSizes[0] - 1);
+
+ // Entrance LPF. Note, comb.process() differs a bit here.
+ combs[0]->process(dry);
+
+ // This introduces reverb noise which actually makes output from the real Boss chip nondeterministic
+ link = link - 1;
+ link = allpasses[0]->process(link);
+ link = allpasses[1]->process(link);
+ link = allpasses[2]->process(link);
+
+ // If the output position is equal to the comb size, get it now in order not to loose it
+ outL1 = combs[1]->getOutputAt(currentSettings.outLPositions[0] - 1);
+ outL1 += outL1 >> 1;
+
+ combs[1]->process(link);
+ combs[2]->process(link);
+ combs[3]->process(link);
+
+ link = combs[2]->getOutputAt(currentSettings.outLPositions[1]);
+ link += link >> 1;
+ link += outL1;
+ link += combs[3]->getOutputAt(currentSettings.outLPositions[2]);
+ *outLeft = weirdMul(link, wetLevel, 0xFF) / 8192.0f;
+
+ outR1 = combs[1]->getOutputAt(currentSettings.outRPositions[0]);
+ outR1 += outR1 >> 1;
+ link = combs[2]->getOutputAt(currentSettings.outRPositions[1]);
+ link += link >> 1;
+ link += outR1;
+ link += combs[3]->getOutputAt(currentSettings.outRPositions[2]);
+ *outRight = weirdMul(link, wetLevel, 0xFF) / 8192.0f;
+ }
+
+ inLeft++;
+ inRight++;
+ outLeft++;
+ outRight++;
+ }
+}
+
+}
+
+#endif
diff --git a/audio/softsynth/mt32/BReverbModel.h b/audio/softsynth/mt32/BReverbModel.h
new file mode 100644
index 0000000..7096ae0
--- /dev/null
+++ b/audio/softsynth/mt32/BReverbModel.h
@@ -0,0 +1,112 @@
+/* Copyright (C) 2003, 2004, 2005, 2006, 2008, 2009 Dean Beeler, Jerome Fisher
+ * 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
+ * the Free Software Foundation, either version 2.1 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef MT32EMU_B_REVERB_MODEL_H
+#define MT32EMU_B_REVERB_MODEL_H
+
+namespace MT32Emu {
+
+struct BReverbSettings {
+ const Bit32u numberOfAllpasses;
+ const Bit32u * const allpassSizes;
+ const Bit32u numberOfCombs;
+ const Bit32u * const combSizes;
+ const Bit32u * const outLPositions;
+ const Bit32u * const outRPositions;
+ const Bit32u * const filterFactors;
+ const Bit32u * const feedbackFactors;
+ const Bit32u * const dryAmps;
+ const Bit32u * const wetLevels;
+ const Bit32u lpfAmp;
+};
+
+class RingBuffer {
+protected:
+ Bit16s *buffer;
+ const Bit32u size;
+ Bit32u index;
+
+public:
+ RingBuffer(const Bit32u size);
+ virtual ~RingBuffer();
+ Bit32s next();
+ bool isEmpty() const;
+ void mute();
+};
+
+class AllpassFilter : public RingBuffer {
+public:
+ AllpassFilter(const Bit32u size);
+ Bit32s process(const Bit32s in);
+};
+
+class CombFilter : public RingBuffer {
+protected:
+ const Bit32u filterFactor;
+ Bit32u feedbackFactor;
+
+public:
+ CombFilter(const Bit32u size, const Bit32u useFilterFactor);
+ virtual void process(const Bit32s in); // Actually, no need to make it virtual, but for sure
+ Bit32s getOutputAt(const Bit32u outIndex) const;
+ void setFeedbackFactor(const Bit32u useFeedbackFactor);
+};
+
+class DelayWithLowPassFilter : public CombFilter {
+ Bit32u amp;
+
+public:
+ DelayWithLowPassFilter(const Bit32u useSize, const Bit32u useFilterFactor, const Bit32u useAmp);
+ void process(const Bit32s in);
+ void setFeedbackFactor(const Bit32u) {}
+};
+
+class TapDelayCombFilter : public CombFilter {
+ Bit32u outL;
+ Bit32u outR;
+
+public:
+ TapDelayCombFilter(const Bit32u useSize, const Bit32u useFilterFactor);
+ void process(const Bit32s in);
+ Bit32s getLeftOutput() const;
+ Bit32s getRightOutput() const;
+ void setOutputPositions(const Bit32u useOutL, const Bit32u useOutR);
+};
+
+class BReverbModel : public ReverbModel {
+ AllpassFilter **allpasses;
+ CombFilter **combs;
+
+ const BReverbSettings ¤tSettings;
+ const bool tapDelayMode;
+ Bit32u dryAmp;
+ Bit32u wetLevel;
+ void mute();
+
+public:
+ BReverbModel(const ReverbMode mode);
+ ~BReverbModel();
+ void open(unsigned int sampleRate);
+ void close();
+ void setParameters(Bit8u time, Bit8u level);
+ void process(const float *inLeft, const float *inRight, float *outLeft, float *outRight, unsigned long numSamples);
+ bool isActive() const;
+};
+
+}
+
+#endif
diff --git a/audio/softsynth/mt32/DelayReverb.cpp b/audio/softsynth/mt32/DelayReverb.cpp
index 23d25a5..77521b4 100644
--- a/audio/softsynth/mt32/DelayReverb.cpp
+++ b/audio/softsynth/mt32/DelayReverb.cpp
@@ -22,7 +22,7 @@
namespace MT32Emu {
-// CONFIRMED: The values below are found via analysis of digital samples. Checked with all time and level combinations.
+// CONFIRMED: The values below are found via analysis of digital samples and tracing reverb RAM address / data lines. Checked with all time and level combinations.
// Obviously:
// rightDelay = (leftDelay - 2) * 2 + 2
// echoDelay = rightDelay - 1
@@ -39,10 +39,11 @@ static const Bit32u REVERB_TIMINGS[8][3]= {
{8002, 16002, 16001}
};
-static const float REVERB_FADE[8] = {0.0f, -0.049400051f, -0.08220577f, -0.131861118f, -0.197344907f, -0.262956344f, -0.345162114f, -0.509508615f};
-const float REVERB_FEEDBACK67 = -0.629960524947437f; // = -EXP2F(-2 / 3)
-const float REVERB_FEEDBACK = -0.682034520443118f; // = -EXP2F(-53 / 96)
-const float LPF_VALUE = 0.594603558f; // = EXP2F(-0.75f)
+// Reverb amp is found as dryAmp * wetAmp
+static const Bit32u REVERB_AMP[8] = {0x20*0x18, 0x50*0x18, 0x50*0x28, 0x50*0x40, 0x50*0x60, 0x50*0x80, 0x50*0xA8, 0x50*0xF8};
+static const Bit32u REVERB_FEEDBACK67 = 0x60;
+static const Bit32u REVERB_FEEDBACK = 0x68;
+static const float LPF_VALUE = 0x68 / 256.0f;
DelayReverb::DelayReverb() {
buf = NULL;
@@ -97,20 +98,18 @@ void DelayReverb::recalcParameters() {
// Number of samples between a response and that response feeding back/echoing
delayFeedback = REVERB_TIMINGS[time][2] * sampleRate / 32000;
- if (time < 6) {
- feedback = REVERB_FEEDBACK;
+ if (level < 3 || time < 6) {
+ feedback = REVERB_FEEDBACK / 256.0f;
} else {
- feedback = REVERB_FEEDBACK67;
+ feedback = REVERB_FEEDBACK67 / 256.0f;
}
- // Fading speed, i.e. amplitude ratio of neighbor responses
- fade = REVERB_FADE[level];
+ // Overall output amp
+ amp = (level == 0 && time == 0) ? 0.0f : REVERB_AMP[level] / 65536.0f;
}
void DelayReverb::process(const float *inLeft, const float *inRight, float *outLeft, float *outRight, unsigned long numSamples) {
- if (buf == NULL) {
- return;
- }
+ if (buf == NULL) return;
for (unsigned int sampleIx = 0; sampleIx < numSamples; sampleIx++) {
// The ring buffer write index moves backwards; reads are all done with positive offsets.
@@ -120,10 +119,10 @@ void DelayReverb::process(const float *inLeft, const float *inRight, float *outL
Bit32u bufIxFeedback = (bufIx + delayFeedback) % bufSize;
// Attenuated input samples and feedback response are directly added to the current ring buffer location
- float sample = fade * (inLeft[sampleIx] + inRight[sampleIx]) + feedback * buf[bufIxFeedback];
+ float lpfIn = amp * (inLeft[sampleIx] + inRight[sampleIx]) + feedback * buf[bufIxFeedback];
// Single-pole IIR filter found on real devices
- buf[bufIx] = buf[bufIxPrev] + (sample - buf[bufIxPrev]) * LPF_VALUE;
+ buf[bufIx] = buf[bufIxPrev] * LPF_VALUE - lpfIn;
outLeft[sampleIx] = buf[bufIxLeft];
outRight[sampleIx] = buf[bufIxRight];
@@ -133,17 +132,13 @@ void DelayReverb::process(const float *inLeft, const float *inRight, float *outL
}
bool DelayReverb::isActive() const {
- // Quick hack: Return true iff all samples in the left buffer are the same and
- // all samples in the right buffers are the same (within the sample output threshold).
- if (buf == NULL) {
- return false;
- }
- float last = buf[0] * 8192.0f;
- for (unsigned int i = 1; i < bufSize; i++) {
- float s = (buf[i] * 8192.0f);
- if (fabs(s - last) > 1.0f) {
- return true;
- }
+ if (buf == NULL) return false;
+
+ float *b = buf;
+ float max = 0.001f;
+ for (Bit32u i = 0; i < bufSize; i++) {
+ if ((*b < -max) || (*b > max)) return true;
+ b++;
}
return false;
}
diff --git a/audio/softsynth/mt32/DelayReverb.h b/audio/softsynth/mt32/DelayReverb.h
index 7c030fb..e967ddc 100644
--- a/audio/softsynth/mt32/DelayReverb.h
+++ b/audio/softsynth/mt32/DelayReverb.h
@@ -35,7 +35,7 @@ private:
Bit32u delayRight;
Bit32u delayFeedback;
- float fade;
+ float amp;
float feedback;
void recalcParameters();
diff --git a/audio/softsynth/mt32/Synth.cpp b/audio/softsynth/mt32/Synth.cpp
index f37d76c..9b5518b 100644
--- a/audio/softsynth/mt32/Synth.cpp
+++ b/audio/softsynth/mt32/Synth.cpp
@@ -27,8 +27,10 @@
#include "mmath.h"
#include "PartialManager.h"
-#if MT32EMU_USE_AREVERBMODEL == 1
+#if MT32EMU_USE_REVERBMODEL == 1
#include "AReverbModel.h"
+#elif MT32EMU_USE_REVERBMODEL == 2
+#include "BReverbModel.h"
#else
#include "FreeverbModel.h"
#endif
@@ -145,17 +147,23 @@ Synth::Synth() {
reverbEnabled = true;
reverbOverridden = false;
-#if MT32EMU_USE_AREVERBMODEL == 1
- reverbModels[0] = new AReverbModel(&AReverbModel::REVERB_MODE_0_SETTINGS);
- reverbModels[1] = new AReverbModel(&AReverbModel::REVERB_MODE_1_SETTINGS);
- reverbModels[2] = new AReverbModel(&AReverbModel::REVERB_MODE_2_SETTINGS);
+#if MT32EMU_USE_REVERBMODEL == 1
+ reverbModels[REVERB_MODE_ROOM] = new AReverbModel(REVERB_MODE_ROOM);
+ reverbModels[REVERB_MODE_HALL] = new AReverbModel(REVERB_MODE_HALL);
+ reverbModels[REVERB_MODE_PLATE] = new AReverbModel(REVERB_MODE_PLATE);
+ reverbModels[REVERB_MODE_TAP_DELAY] = new DelayReverb();
+#elif MT32EMU_USE_REVERBMODEL == 2
+ reverbModels[REVERB_MODE_ROOM] = new BReverbModel(REVERB_MODE_ROOM);
+ reverbModels[REVERB_MODE_HALL] = new BReverbModel(REVERB_MODE_HALL);
+ reverbModels[REVERB_MODE_PLATE] = new BReverbModel(REVERB_MODE_PLATE);
+ reverbModels[REVERB_MODE_TAP_DELAY] = new BReverbModel(REVERB_MODE_TAP_DELAY);
#else
- reverbModels[0] = new FreeverbModel(0.76f, 0.687770909f, 0.63f, 0, 0.5f);
- reverbModels[1] = new FreeverbModel(2.0f, 0.712025098f, 0.86f, 1, 0.5f);
- reverbModels[2] = new FreeverbModel(0.4f, 0.939522749f, 0.38f, 2, 0.05f);
+ reverbModels[REVERB_MODE_ROOM] = new FreeverbModel(0.76f, 0.687770909f, 0.63f, 0, 0.5f);
+ reverbModels[REVERB_MODE_HALL] = new FreeverbModel(2.0f, 0.712025098f, 0.86f, 1, 0.5f);
+ reverbModels[REVERB_MODE_PLATE] = new FreeverbModel(0.4f, 0.939522749f, 0.38f, 2, 0.05f);
+ reverbModels[REVERB_MODE_TAP_DELAY] = new DelayReverb();
#endif
- reverbModels[3] = new DelayReverb();
reverbModel = NULL;
setDACInputMode(DACInputMode_NICE);
setOutputGain(1.0f);
@@ -315,7 +323,7 @@ LoadResult Synth::loadPCMROM(const char *filename) {
return LoadResult_Unreadable;
}
LoadResult rc = LoadResult_OK;
- for (int i = 0; i < pcmROMSize; i++) {
+ for (unsigned int i = 0; i < pcmROMSize; i++) {
Bit8u s = file->readByte();
Bit8u c = file->readByte();
@@ -350,7 +358,7 @@ LoadResult Synth::loadPCMROM(const char *filename) {
bool Synth::initPCMList(Bit16u mapAddress, Bit16u count) {
ControlROMPCMStruct *tps = (ControlROMPCMStruct *)&controlROMData[mapAddress];
for (int i = 0; i < count; i++) {
- int rAddr = tps[i].pos * 0x800;
+ unsigned int rAddr = tps[i].pos * 0x800;
int rLenExp = (tps[i].len & 0x70) >> 4;
int rLen = 0x800 << rLenExp;
if (rAddr + rLen > pcmROMSize) {
diff --git a/audio/softsynth/mt32/Synth.h b/audio/softsynth/mt32/Synth.h
index a30df62..2534b7a 100644
--- a/audio/softsynth/mt32/Synth.h
+++ b/audio/softsynth/mt32/Synth.h
@@ -179,6 +179,13 @@ enum MemoryRegionType {
MR_PatchTemp, MR_RhythmTemp, MR_TimbreTemp, MR_Patches, MR_Timbres, MR_System, MR_Display, MR_Reset
};
+enum ReverbMode {
+ REVERB_MODE_ROOM,
+ REVERB_MODE_HALL,
+ REVERB_MODE_PLATE,
+ REVERB_MODE_TAP_DELAY
+};
+
class MemoryRegion {
private:
Synth *synth;
@@ -315,7 +322,7 @@ private:
const ControlROMMap *controlROMMap;
Bit8u controlROMData[CONTROL_ROM_SIZE];
float *pcmROMData;
- int pcmROMSize; // This is in 16-bit samples, therefore half the number of bytes in the ROM
+ unsigned int pcmROMSize; // This is in 16-bit samples, therefore half the number of bytes in the ROM
Bit8s chantable[32];
diff --git a/audio/softsynth/mt32/module.mk b/audio/softsynth/mt32/module.mk
index 995e450..c0ee363 100644
--- a/audio/softsynth/mt32/module.mk
+++ b/audio/softsynth/mt32/module.mk
@@ -2,6 +2,7 @@ MODULE := audio/softsynth/mt32
MODULE_OBJS := \
AReverbModel.o \
+ BReverbModel.o \
DelayReverb.o \
FreeverbModel.o \
LA32Ramp.o \
diff --git a/audio/softsynth/mt32/mt32emu.h b/audio/softsynth/mt32/mt32emu.h
index f10bc1f..a5a72b0 100644
--- a/audio/softsynth/mt32/mt32emu.h
+++ b/audio/softsynth/mt32/mt32emu.h
@@ -84,9 +84,10 @@
// If zero, keeps reverb buffers for all modes around all the time to avoid allocating/freeing in the critical path.
#define MT32EMU_REDUCE_REVERB_MEMORY 1
-// 0: Use standard Freeverb
-// 1: Use AReverb (currently not properly tuned)
-#define MT32EMU_USE_AREVERBMODEL 0
+// 0: Use legacy Freeverb
+// 1: Use Accurate Reverb model aka AReverb
+// 2: Use Bit-perfect Boss Reverb model aka BReverb (for developers, not much practical use)
+#define MT32EMU_USE_REVERBMODEL 1
namespace MT32Emu
{
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