Several small fixes.

Added newer third-party libraries.

MicroDexed.ino

@@ -1276,7 +1276,7 @@ void handleProgramChange(uint8_t inChannel, uint8_t inProgram) {
   }
 }
 
-void handleSystemExclusive(byte* sysex, unsigned int len) {
+void handleSystemExclusive(uint8_t* sysex, unsigned int len) {
   int16_t sysex_return;
 
   for (uint8_t instance_id = 0; instance_id < NUM_DEXED; instance_id++) {
@@ -2124,7 +2124,7 @@ void set_voiceconfig_params(uint8_t instance_id) {
   microdexed_peak_mixer.gain(instance_id, 1.0);
 
   // Controller
-  MicroDexed[instance_id]->setMaxNotes(configuration.dexed[instance_id].polyphony);
+  //MicroDexed[instance_id]->setMaxNotes(configuration.dexed[instance_id].polyphony);
   MicroDexed[instance_id]->setPBController(configuration.dexed[instance_id].pb_range, configuration.dexed[instance_id].pb_step);
   MicroDexed[instance_id]->setMWController(configuration.dexed[instance_id].mw_range, configuration.dexed[instance_id].mw_assign, configuration.dexed[instance_id].mw_mode);
   MicroDexed[instance_id]->setFCController(configuration.dexed[instance_id].fc_range, configuration.dexed[instance_id].fc_assign, configuration.dexed[instance_id].fc_mode);
@@ -2347,7 +2347,7 @@ float volume_transform(float in) {
   return powf(in, VOLUME_TRANSFORM_EXP);
 }
 
-uint32_t crc32(byte* calc_start, uint16_t calc_bytes)  // base code from https://www.arduino.cc/en/Tutorial/EEPROMCrc
+uint32_t crc32(uint8_t* calc_start, uint16_t calc_bytes)  // base code from https://www.arduino.cc/en/Tutorial/EEPROMCrc
 {
   const uint32_t crc_table[16] = {
     0x00000000, 0x1db71064, 0x3b6e20c8, 0x26d930ac,
@@ -2357,7 +2357,7 @@ uint32_t crc32(byte* calc_start, uint16_t calc_bytes)  // base code from https:/
   };
   uint32_t crc = ~0L;
 
-  for (byte* index = calc_start; index < (calc_start + calc_bytes); ++index) {
+  for (uint8_t* index = calc_start; index < (calc_start + calc_bytes); ++index) {
     crc = crc_table[(crc ^ *index) & 0x0f] ^ (crc >> 4);
     crc = crc_table[(crc ^ (*index >> 4)) & 0x0f] ^ (crc >> 4);
     crc = ~crc;

UI.hpp

@@ -1673,7 +1673,7 @@ void getNoteName(char* noteName, uint8_t noteNumber) {
   uint8_t oct_index = noteNumber - 12;
 
   noteNumber -= 21;
-  snprintf_P(noteName, sizeof(noteName), PSTR("%2s%1d"), notes[noteNumber % 12], oct_index / 12);
+  snprintf_P(noteName, sizeof(noteName), PSTR("%2S%1d"), notes[noteNumber % 12], oct_index / 12);
 }
 
 void UI_func_lowest_note(uint8_t param) {
@@ -5529,7 +5529,7 @@ void UI_func_sysex_receive_bank(uint8_t param) {
 #endif
             char tmp[CONFIG_FILENAME_LEN];
             strlcpy(tmp, receive_bank_filename, sizeof(tmp));
-            snprintf_P(receive_bank_filename, sizeof(receive_bank_filename), PSTR("/%d/%s.syx"), bank_number, tmp);
+            snprintf_P(receive_bank_filename, sizeof(receive_bank_filename), PSTR("/%2d/%12s.syx"), bank_number, tmp);
 #ifdef DEBUG
             Serial.print(F("Receiving into bank "));
             Serial.print(bank_number);

config.h

@@ -263,7 +263,7 @@
 #define VOICE_NAME_LEN 12  // 11 (plus '\0')
 #define FILENAME_LEN BANK_NAME_LEN + VOICE_NAME_LEN
 #define CONFIG_FILENAME_LEN 50
-#define DRUM_NAME_LEN 16
+#define DRUM_NAME_LEN 17
 #define PERFORMANCE_NAME_LEN 11
 
 #define FAV_CONFIG_PATH "FAVCFG"

dexed_sd.h

@@ -40,7 +40,7 @@ extern uint8_t voice;
 extern uint8_t ui_state;
 extern uint8_t ui_main_state;
 extern config_t configuration;
-extern uint32_t crc32(byte* calc_start, uint16_t calc_bytes);
+extern uint32_t crc32(uint8_t* calc_start, uint16_t calc_uint8_ts);
 extern void set_fx_params(void);
 extern void set_voiceconfig_params(uint8_t instance_id);
 extern void set_sys_params(void);

third-party/Synth_Dexed/EngineMkI.cpp

@@ -0,0 +1,349 @@
+/*
+ * Copyright (C) 2015-2017 Pascal Gauthier.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software Foundation,
+ * Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
+ *
+ * The code is based on ppplay https://github.com/stohrendorf/ppplay and opl3
+ * math documentation :
+ * https://github.com/gtaylormb/opl3_fpga/blob/master/docs/opl3math/opl3math.pdf
+ *
+ */
+
+#include "EngineMkI.h"
+
+#define _USE_MATH_DEFINES
+#include <cmath>
+#include <cstdlib>
+
+#include "sin.h"
+#include "exp2.h"
+
+#ifdef DEBUG
+    #include "time.h"
+    //#define MKIDEBUG
+#endif
+
+const int32_t __attribute__ ((aligned(16))) zeros[_N_] = {0};
+
+static const uint16_t NEGATIVE_BIT = 0x8000;
+static const uint16_t ENV_BITDEPTH = 14;
+
+static const uint16_t SINLOG_BITDEPTH = 10;
+static const uint16_t SINLOG_TABLESIZE = 1<<SINLOG_BITDEPTH;
+static uint16_t sinLogTable[SINLOG_TABLESIZE];
+
+static const uint16_t SINEXP_BITDEPTH = 10;
+static const uint16_t SINEXP_TABLESIZE = 1<<SINEXP_BITDEPTH;
+static uint16_t sinExpTable[SINEXP_TABLESIZE];
+
+const uint16_t ENV_MAX = 1<<ENV_BITDEPTH;
+
+static inline uint16_t sinLog(uint16_t phi) {
+    const uint16_t SINLOG_TABLEFILTER = SINLOG_TABLESIZE-1;
+    const uint16_t index = (phi & SINLOG_TABLEFILTER);
+    
+    switch( ( phi & (SINLOG_TABLESIZE * 3) ) ) {
+        case 0:
+            return sinLogTable[index];
+        case SINLOG_TABLESIZE:
+            return sinLogTable[index ^ SINLOG_TABLEFILTER];
+        case SINLOG_TABLESIZE * 2 :
+            return sinLogTable[index] | NEGATIVE_BIT;
+        default:
+            return sinLogTable[index ^ SINLOG_TABLEFILTER] | NEGATIVE_BIT;
+    }
+}
+
+EngineMkI::EngineMkI() {
+    float bitReso = SINLOG_TABLESIZE;
+    
+    for(int32_t i=0;i<SINLOG_TABLESIZE;i++) {
+        float x1 = sin(((0.5+i)/bitReso) * M_PI/2.0);
+        sinLogTable[i] = round(-1024 * log2(x1));
+    }
+    
+    bitReso = SINEXP_TABLESIZE;
+    for(int32_t i=0;i<SINEXP_TABLESIZE;i++) {
+        float x1 = (pow(2, float(i)/bitReso)-1) * 4096;
+        sinExpTable[i] = round(x1);
+    }
+    
+#ifdef MKIDEBUG
+    uint8_t buffer[4096];
+    int32_t pos = 0;
+    
+    TRACE("****************************************");
+    for(int32_t i=0;i<SINLOG_TABLESIZE;i++) {
+        pos += sprintf(buffer+pos, "%d ", sinLogTable[i]);
+        if ( pos > 90 ) {
+            TRACE("SINLOGTABLE: %s" ,buffer);
+            buffer[0] = 0;
+            pos = 0;
+        }
+    }
+    TRACE("SINLOGTABLE: %s", buffer);
+    buffer[0] = 0;
+    pos = 0;
+    TRACE("----------------------------------------");    
+    for(int32_t i=0;i<SINEXP_TABLESIZE;i++) {
+        pos += sprintf(buffer+pos, "%d ", sinExpTable[i]);
+        if ( pos > 90 ) {
+            TRACE("SINEXTTABLE: %s" ,buffer);
+            buffer[0] = 0;
+            pos = 0;
+        }
+    }
+    TRACE("SINEXTTABLE: %s", buffer);
+    TRACE("****************************************");
+#endif
+}
+
+inline int32_t mkiSin(int32_t phase, uint16_t env) {
+    uint16_t expVal = sinLog(phase >> (22 - SINLOG_BITDEPTH)) + (env);
+    //int16_t expValShow = expVal;
+    
+    const bool isSigned = expVal & NEGATIVE_BIT;
+    expVal &= ~NEGATIVE_BIT;
+    
+    const uint16_t SINEXP_FILTER = 0x3FF;
+    uint16_t result = 4096 + sinExpTable[( expVal & SINEXP_FILTER ) ^ SINEXP_FILTER];
+    
+    //uint16_t resultB4 = result;
+    result >>= ( expVal >> 10 ); // exp
+    
+#ifdef MKIDEBUG
+    if ( ( time(NULL) % 5 ) == 0 ) {
+        if ( expValShow < 0 ) {
+            expValShow = (expValShow + 0x7FFF) * -1;
+        }
+        //TRACE(",%d,%d,%d,%d,%d,%d", phase >> (22 - SINLOG_BITDEPTH), env, expValShow, ( expVal & SINEXP_FILTER ) ^ SINEXP_FILTER, resultB4, result);
+    }
+#endif
+    
+    if( isSigned )
+        return (-result - 1) << 13;
+    else
+        return result << 13;
+}
+
+void EngineMkI::compute(int32_t *output, const int32_t *input,
+                        int32_t phase0, int32_t freq,
+                        int32_t gain1, int32_t gain2, bool add) {
+    int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
+    int32_t gain = gain1;
+    int32_t phase = phase0;
+    const int32_t *adder = add ? output : zeros;
+
+    for (uint8_t i = 0; i < _N_; i++) {
+        gain += dgain;
+        int32_t y = mkiSin((phase+input[i]), gain);
+        output[i] = y + adder[i];
+        phase += freq;
+    }
+}
+
+void EngineMkI::compute_pure(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, bool add) {
+    int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
+    int32_t gain = gain1;
+    int32_t phase = phase0;
+    const int32_t *adder = add ? output : zeros;
+    
+    for (uint8_t i = 0; i < _N_; i++) {
+        gain += dgain;
+        int32_t y = mkiSin(phase , gain);
+        output[i] = y + adder[i];
+        phase += freq;
+    }
+}
+
+void EngineMkI::compute_fb(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, int32_t *fb_buf, int32_t fb_shift, bool add) {
+    int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
+    int32_t gain = gain1;
+    int32_t phase = phase0;
+    const int32_t *adder = add ? output : zeros;
+    int32_t y0 = fb_buf[0];
+    int32_t y = fb_buf[1];
+    
+    for (uint8_t i = 0; i < _N_; i++) {
+        gain += dgain;
+        int32_t scaled_fb = (y0 + y) >> (fb_shift + 1);
+        y0 = y;
+        y = mkiSin((phase+scaled_fb), gain);
+        output[i] = y + adder[i];
+        phase += freq;
+    }
+    
+    fb_buf[0] = y0;
+    fb_buf[1] = y;
+}
+
+// exclusively used for ALGO 6 with feedback
+void EngineMkI::compute_fb2(int32_t *output, FmOpParams *parms, int32_t gain01, int32_t gain02, int32_t *fb_buf, int32_t fb_shift) {
+    int32_t dgain[2];
+    int32_t gain[2];
+    int32_t phase[2];
+    int32_t y0 = fb_buf[0];
+    int32_t y = fb_buf[1];
+    
+    phase[0] = parms[0].phase;
+    phase[1] = parms[1].phase;
+
+    parms[1].gain_out = (ENV_MAX-(parms[1].level_in >> (28-ENV_BITDEPTH)));
+
+    gain[0] = gain01;
+    gain[1] = parms[1].gain_out == 0 ? (ENV_MAX-1) : parms[1].gain_out;
+
+    dgain[0] = (gain02 - gain01 + (_N_ >> 1)) >> LG_N;
+    dgain[1] = (parms[1].gain_out - (parms[1].gain_out == 0 ? (ENV_MAX-1) : parms[1].gain_out));
+    
+    for (uint8_t i = 0; i < _N_; i++) {
+        int32_t scaled_fb = (y0 + y) >> (fb_shift + 1);
+        
+        // op 0
+        gain[0] += dgain[0];
+        y0 = y;
+        y = mkiSin(phase[0]+scaled_fb, gain[0]);
+        phase[0] += parms[0].freq;
+        
+        // op 1
+        gain[1] += dgain[1];
+        y = mkiSin(phase[1]+y, gain[1]);
+        phase[1] += parms[1].freq;
+        
+        output[i] = y;
+    }
+    fb_buf[0] = y0;
+    fb_buf[1] = y;
+}
+
+// exclusively used for ALGO 4 with feedback
+void EngineMkI::compute_fb3(int32_t *output, FmOpParams *parms, int32_t gain01, int32_t gain02, int32_t *fb_buf, int32_t fb_shift) {
+    int32_t dgain[3];
+    int32_t gain[3];
+    int32_t phase[3];
+    int32_t y0 = fb_buf[0];
+    int32_t y = fb_buf[1];
+    
+    phase[0] = parms[0].phase;
+    phase[1] = parms[1].phase;
+    phase[2] = parms[2].phase;
+    
+    parms[1].gain_out = (ENV_MAX-(parms[1].level_in >> (28-ENV_BITDEPTH)));
+    parms[2].gain_out = (ENV_MAX-(parms[2].level_in >> (28-ENV_BITDEPTH)));
+    
+    gain[0] = gain01;
+    gain[1] = parms[1].gain_out == 0 ? (ENV_MAX-1) : parms[1].gain_out;
+    gain[2] = parms[2].gain_out == 0 ? (ENV_MAX-1) : parms[2].gain_out;
+
+    dgain[0] = (gain02 - gain01 + (_N_ >> 1)) >> LG_N;
+    dgain[1] = (parms[1].gain_out - (parms[1].gain_out == 0 ? (ENV_MAX-1) : parms[1].gain_out));
+    dgain[2] = (parms[2].gain_out - (parms[2].gain_out == 0 ? (ENV_MAX-1) : parms[2].gain_out));
+    
+    
+    for (uint8_t i = 0; i < _N_; i++) {
+        int32_t scaled_fb = (y0 + y) >> (fb_shift + 1);
+        
+        // op 0
+        gain[0] += dgain[0];
+        y0 = y;
+        y = mkiSin(phase[0]+scaled_fb, gain[0]);
+        phase[0] += parms[0].freq;
+        
+        // op 1
+        gain[1] += dgain[1];
+        y = mkiSin(phase[1]+y, gain[1]);
+        phase[1] += parms[1].freq;
+        
+        // op 2
+        gain[2] += dgain[2];
+        y = mkiSin(phase[2]+y, gain[2]);
+        phase[2] += parms[2].freq;
+        
+        output[i] = y;
+    }
+    fb_buf[0] = y0;
+    fb_buf[1] = y;
+}
+
+void EngineMkI::render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_shift) {
+    const int32_t kLevelThresh = ENV_MAX-100;
+    FmAlgorithm alg = algorithms[algorithm];
+    bool has_contents[3] = { true, false, false };
+    bool fb_on = feedback_shift < 16;
+
+    switch(algorithm) {
+        case 3 : case 5 :
+            if ( fb_on )
+                alg.ops[0] = 0xc4;
+    }
+    
+    for (int32_t op = 0; op < 6; op++) {
+        int32_t flags = alg.ops[op];
+        bool add = (flags & OUT_BUS_ADD) != 0;
+        FmOpParams &param = params[op];
+        int32_t inbus = (flags >> 4) & 3;
+        int32_t outbus = flags & 3;
+        int32_t *outptr = (outbus == 0) ? output : buf_[outbus - 1].get();
+        int32_t gain1 = param.gain_out == 0 ? (ENV_MAX-1) : param.gain_out;
+        int32_t gain2 = ENV_MAX-(param.level_in >> (28-ENV_BITDEPTH));
+        param.gain_out = gain2;
+        
+        if (gain1 <= kLevelThresh || gain2 <= kLevelThresh) {
+            
+            if (!has_contents[outbus]) {
+                add = false;
+            }
+            
+            if (inbus == 0 || !has_contents[inbus]) {
+                // PG: this is my 'dirty' implementation of FB for 2 and 3 operators...
+                if ((flags & 0xc0) == 0xc0 && fb_on) {
+                    switch ( algorithm ) {
+                        // three operator feedback, process exception for ALGO 4
+                        case 3 :
+                            compute_fb3(outptr, params, gain1, gain2, fb_buf, std::min((feedback_shift+2), (int32_t)16));
+                            params[1].phase += params[1].freq << LG_N; // hack, we already processed op-5 - op-4
+                            params[2].phase += params[2].freq << LG_N; // yuk yuk
+                            op += 2; // ignore the 2 other operators
+                            break;
+                        // two operator feedback, process exception for ALGO 6
+                        case 5 :
+                            compute_fb2(outptr, params, gain1, gain2, fb_buf, std::min((feedback_shift+2), (int32_t)16));
+                            params[1].phase += params[1].freq << LG_N;  // yuk, hack, we already processed op-5
+                            op++; // ignore next operator;
+                            break;
+                        case 31 :
+                            // one operator feedback, process exception for ALGO 32
+                            compute_fb(outptr, param.phase, param.freq, gain1, gain2, fb_buf, std::min((feedback_shift+2), (int32_t)16), add);
+                            break;
+                        default:
+                            // one operator feedback, normal process
+                            compute_fb(outptr, param.phase, param.freq, gain1, gain2, fb_buf, feedback_shift, add);
+                            break;
+                    }
+                } else {
+                    compute_pure(outptr, param.phase, param.freq, gain1, gain2, add);
+                }
+            } else {
+                compute(outptr, buf_[inbus - 1].get(), param.phase, param.freq, gain1, gain2, add);
+            }
+            
+            has_contents[outbus] = true;
+        } else if (!add) {
+            has_contents[outbus] = false;
+        }
+        param.phase += param.freq << LG_N;
+    }
+}
+

third-party/Synth_Dexed/EngineMkI.h

@@ -0,0 +1,44 @@
+/*
+ * Copyright 2014 Pascal Gauthier.
+ * Copyright 2012 Google Inc.
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ENGINEMKI_H_INCLUDED
+#define ENGINEMKI_H_INCLUDED
+
+#include "aligned_buf.h"
+#include "fm_op_kernel.h"
+#include "controllers.h"
+#include "fm_core.h"
+
+class EngineMkI : public FmCore {
+public:
+    EngineMkI();
+    ~EngineMkI() {};
+    void render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_shift) override;
+    
+    void compute(int32_t *output, const int32_t *input, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, bool add);
+    
+    void compute_pure(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, bool add);
+    
+    void compute_fb(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, int32_t *fb_buf, int32_t fb_gain, bool add);
+    
+    void compute_fb2(int32_t *output, FmOpParams *params, int32_t gain01, int32_t gain02, int32_t *fb_buf, int32_t fb_shift);
+    
+    void compute_fb3(int32_t *output, FmOpParams *params, int32_t gain01, int32_t gain02, int32_t *fb_buf, int32_t fb_shift);
+};
+
+
+#endif  // ENGINEMKI_H_INCLUDED

third-party/Synth_Dexed/EngineMsfa.cpp

@@ -0,0 +1,66 @@
+/*
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+//using namespace std;
+
+#include "synth.h"
+#include "exp2.h"
+#include "fm_op_kernel.h"
+#include "EngineMsfa.h"
+
+void EngineMsfa::render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_shift) {
+  const int32_t kLevelThresh = 1120;
+  const FmAlgorithm alg = algorithms[algorithm];
+  bool has_contents[3] = { true, false, false };
+  for (uint8_t op = 0; op < 6; op++) {
+    int32_t flags = alg.ops[op];
+    bool add = (flags & OUT_BUS_ADD) != 0;
+    FmOpParams &param = params[op];
+    int32_t inbus = (flags >> 4) & 3;
+    int32_t outbus = flags & 3;
+    int32_t *outptr = (outbus == 0) ? output : buf_[outbus - 1].get();
+    int32_t gain1 = param.gain_out;
+    int32_t gain2 = Exp2::lookup(param.level_in - (14 * (1 << 24)));
+    param.gain_out = gain2;
+
+    if (gain1 >= kLevelThresh || gain2 >= kLevelThresh) {
+      if (!has_contents[outbus]) {
+        add = false;
+      }
+      if (inbus == 0 || !has_contents[inbus]) {
+        // todo: more than one op in a feedback loop
+        if ((flags & 0xc0) == 0xc0 && feedback_shift < 16) {
+          // cout << op << " fb " << inbus << outbus << add << endl;
+          FmOpKernel::compute_fb(outptr, param.phase, param.freq,
+                                 gain1, gain2,
+                                 fb_buf, feedback_shift, add);
+        } else {
+          // cout << op << " pure " << inbus << outbus << add << endl;
+          FmOpKernel::compute_pure(outptr, param.phase, param.freq,
+                                   gain1, gain2, add);
+        }
+      } else {
+        // cout << op << " normal " << inbus << outbus << " " << param.freq << add << endl;
+        FmOpKernel::compute(outptr, buf_[inbus - 1].get(),
+                            param.phase, param.freq, gain1, gain2, add);
+      }
+      has_contents[outbus] = true;
+    } else if (!add) {
+      has_contents[outbus] = false;
+    }
+    param.phase += param.freq << LG_N;
+  }
+}

third-party/Synth_Dexed/EngineMsfa.h

@@ -0,0 +1,29 @@
+/*
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#pragma once
+
+#include "fm_op_kernel.h"
+#include "aligned_buf.h"
+#include "controllers.h"
+#include "fm_core.h"
+
+class EngineMsfa : public FmCore {
+  public:
+    EngineMsfa() {};
+    ~EngineMsfa() {};
+    void render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_gain) override;
+};

third-party/Synth_Dexed/EngineOpl.cpp

@@ -0,0 +1,209 @@
+/*
+ * Copyright (C) 2014  Pascal Gauthier.
+ * Copyright (C) 2012  Steffen Ohrendorf <steffen.ohrendorf@gmx.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software Foundation,
+ * Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
+ *
+ * Original Java Code: Copyright (C) 2008 Robson Cozendey <robson@cozendey.com>
+ *
+ * Some code based on forum posts in: http://forums.submarine.org.uk/phpBB/viewforum.php?f=9,
+ * Copyright (C) 2010-2013 by carbon14 and opl3
+ *
+ */
+
+#include "EngineOpl.h"
+
+const int32_t __attribute__ ((aligned(16))) zeros[_N_] = {0};
+
+uint16_t SignBit = 0x8000;
+
+uint16_t sinLogTable[256] = {
+    2137, 1731, 1543, 1419, 1326, 1252, 1190, 1137, 1091, 1050, 1013, 979, 949, 920, 894, 869,
+    846, 825, 804, 785, 767, 749, 732, 717, 701, 687, 672, 659, 646, 633, 621, 609,
+    598, 587, 576, 566, 556, 546, 536, 527, 518, 509, 501, 492, 484, 476, 468, 461,
+    453, 446, 439, 432, 425, 418, 411, 405, 399, 392, 386, 380, 375, 369, 363, 358,
+    352, 347, 341, 336, 331, 326, 321, 316, 311, 307, 302, 297, 293, 289, 284, 280,
+    276, 271, 267, 263, 259, 255, 251, 248, 244, 240, 236, 233, 229, 226, 222, 219,
+    215, 212, 209, 205, 202, 199, 196, 193, 190, 187, 184, 181, 178, 175, 172, 169,
+    167, 164, 161, 159, 156, 153, 151, 148, 146, 143, 141, 138, 136, 134, 131, 129,
+    127, 125, 122, 120, 118, 116, 114, 112, 110, 108, 106, 104, 102, 100, 98, 96,
+    94, 92, 91, 89, 87, 85, 83, 82, 80, 78, 77, 75, 74, 72, 70, 69,
+    67, 66, 64, 63, 62, 60, 59, 57, 56, 55, 53, 52, 51, 49, 48, 47,
+    46, 45, 43, 42, 41, 40, 39, 38, 37, 36, 35, 34, 33, 32, 31, 30,
+    29, 28, 27, 26, 25, 24, 23, 23, 22, 21, 20, 20, 19, 18, 17, 17,
+    16, 15, 15, 14, 13, 13, 12, 12, 11, 10, 10, 9, 9, 8, 8, 7,
+    7, 7, 6, 6, 5, 5, 5, 4, 4, 4, 3, 3, 3, 2, 2, 2,
+    2, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0
+};
+
+uint16_t sinExpTable[256] = {
+    0, 3, 6, 8, 11, 14, 17, 20, 22, 25, 28, 31, 34, 37, 40, 42,
+    45, 48, 51, 54, 57, 60, 63, 66, 69, 72, 75, 78, 81, 84, 87, 90,
+    93, 96, 99, 102, 105, 108, 111, 114, 117, 120, 123, 126, 130, 133, 136, 139,
+    142, 145, 148, 152, 155, 158, 161, 164, 168, 171, 174, 177, 181, 184, 187, 190,
+    194, 197, 200, 204, 207, 210, 214, 217, 220, 224, 227, 231, 234, 237, 241, 244,
+    248, 251, 255, 258, 262, 265, 268, 272, 276, 279, 283, 286, 290, 293, 297, 300,
+    304, 308, 311, 315, 318, 322, 326, 329, 333, 337, 340, 344, 348, 352, 355, 359,
+    363, 367, 370, 374, 378, 382, 385, 389, 393, 397, 401, 405, 409, 412, 416, 420,
+    424, 428, 432, 436, 440, 444, 448, 452, 456, 460, 464, 468, 472, 476, 480, 484,
+    488, 492, 496, 501, 505, 509, 513, 517, 521, 526, 530, 534, 538, 542, 547, 551,
+    555, 560, 564, 568, 572, 577, 581, 585, 590, 594, 599, 603, 607, 612, 616, 621,
+    625, 630, 634, 639, 643, 648, 652, 657, 661, 666, 670, 675, 680, 684, 689, 693,
+    698, 703, 708, 712, 717, 722, 726, 731, 736, 741, 745, 750, 755, 760, 765, 770,
+    774, 779, 784, 789, 794, 799, 804, 809, 814, 819, 824, 829, 834, 839, 844, 849,
+    854, 859, 864, 869, 874, 880, 885, 890, 895, 900, 906, 911, 916, 921, 927, 932,
+    937, 942, 948, 953, 959, 964, 969, 975, 980, 986, 991, 996, 1002, 1007, 1013, 1018
+};
+
+inline uint16_t sinLog(uint16_t phi) {
+    const uint8_t index = (phi & 0xff);
+
+    switch( ( phi & 0x0300 ) ) {
+        case 0x0000:
+            // rising quarter wave  Shape A
+            return sinLogTable[index];
+        case 0x0100:
+            // falling quarter wave  Shape B
+            return sinLogTable[index ^ 0xFF];
+        case 0x0200:
+            // rising quarter wave -ve  Shape C
+            return sinLogTable[index] | SignBit;
+        default:
+            // falling quarter wave -ve  Shape D
+            return sinLogTable[index ^ 0xFF] | SignBit;
+    }
+}
+
+// 16 env units are ~3dB and halve the output
+/**
+ * @brief OPL Sine Wave calculation
+ * @param[in] phase Wave phase (0..1023)
+ * @param[in] env Envelope value (0..511)
+ * @warning @a env will not be checked for correct values.
+ */
+inline int16_t oplSin( uint16_t phase, uint16_t env ) {
+    uint16_t expVal = sinLog(phase) + (env << 3);
+    const bool isSigned = expVal & SignBit;
+
+    expVal &= ~SignBit;
+    // expVal: 0..2137+511*8 = 0..6225
+    // result: 0..1018+1024
+    uint32_t result = 0x0400 + sinExpTable[( expVal & 0xff ) ^ 0xFF];
+    result <<= 1;
+    result >>= ( expVal >> 8 ); // exp
+
+    if( isSigned ) {
+        // -1 for one's complement
+        return -result - 1;
+    } else {
+        return result;
+    }    
+}
+
+void EngineOpl::compute(int32_t *output, const int32_t *input, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, bool add) {
+    int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
+    int32_t gain = gain1;
+    int32_t phase = phase0;
+    const int32_t *adder = add ? output : zeros;
+
+    for (uint8_t i = 0; i < _N_; i++) {
+        gain += dgain;
+        int32_t y = oplSin((phase+input[i]) >> 14, gain);
+        output[i] = (y << 14) + adder[i];
+        phase += freq;
+    }
+}
+
+void EngineOpl::compute_pure(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, bool add) {
+    int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
+    int32_t gain = gain1;
+    int32_t phase = phase0;
+    const int32_t *adder = add ? output : zeros;
+
+    for (uint8_t i = 0; i < _N_; i++) {
+        gain += dgain;
+        int32_t y = oplSin(phase >> 14, gain);
+        output[i] = (y << 14) + adder[i];
+        phase += freq;
+    }
+}
+
+void EngineOpl::compute_fb(int32_t *output, int32_t phase0, int32_t freq,
+                                    int32_t gain1, int32_t gain2,
+                              int32_t *fb_buf, int32_t fb_shift, bool add) {
+    int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
+    int32_t gain = gain1;
+    int32_t phase = phase0;
+    const int32_t *adder = add ? output : zeros;
+    int32_t y0 = fb_buf[0];
+    int32_t y = fb_buf[1];
+    
+    for (uint8_t i = 0; i < _N_; i++) {
+        gain += dgain;
+        int32_t scaled_fb = (y0 + y) >> (fb_shift + 1);
+        y0 = y;
+        y = oplSin((phase+scaled_fb) >> 14, gain) << 14;
+        output[i] = y + adder[i];
+        phase += freq;
+    }
+    
+    fb_buf[0] = y0;
+    fb_buf[1] = y;
+}
+
+
+void EngineOpl::render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_shift) {
+    const int32_t kLevelThresh = 507;  // really ????
+    const FmAlgorithm alg = algorithms[algorithm];
+    bool has_contents[3] = { true, false, false };
+    for (uint8_t op = 0; op < 6; op++) {
+        int32_t flags = alg.ops[op];
+        bool add = (flags & OUT_BUS_ADD) != 0;
+        FmOpParams &param = params[op];
+        int32_t inbus = (flags >> 4) & 3;
+        int32_t outbus = flags & 3;
+        int32_t *outptr = (outbus == 0) ? output : buf_[outbus - 1].get();
+        int32_t gain1 = param.gain_out == 0 ? 511 : param.gain_out;
+        int32_t gain2 = 512-(param.level_in >> 19);
+        param.gain_out = gain2;
+        
+        if (gain1 <= kLevelThresh || gain2 <= kLevelThresh) {
+            if (!has_contents[outbus]) {
+                add = false;
+            }
+            if (inbus == 0 || !has_contents[inbus]) {
+                // todo: more than one op in a feedback loop
+                if ((flags & 0xc0) == 0xc0 && feedback_shift < 16) {
+                    // cout << op << " fb " << inbus << outbus << add << endl;
+                    compute_fb(outptr, param.phase, param.freq,
+                                           gain1, gain2,
+                                           fb_buf, feedback_shift, add);
+                } else {
+                    // cout << op << " pure " << inbus << outbus << add << endl;
+                    compute_pure(outptr, param.phase, param.freq,
+                                             gain1, gain2, add);
+                }
+            } else {
+                // cout << op << " normal " << inbus << outbus << " " << param.freq << add << endl;
+                compute(outptr, buf_[inbus - 1].get(),
+                                    param.phase, param.freq, gain1, gain2, add);
+            }
+            has_contents[outbus] = true;
+        } else if (!add) {
+            has_contents[outbus] = false;
+        }
+        param.phase += param.freq << LG_N;
+    }
+}

third-party/Synth_Dexed/EngineOpl.h

@@ -0,0 +1,39 @@
+/**
+ *
+ * Copyright (c) 2014 Pascal Gauthier.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software Foundation,
+ * Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
+ *
+ */
+
+#ifndef ENGINEOPL_H_INCLUDED
+#define ENGINEOPL_H_INCLUDED
+
+#include "aligned_buf.h"
+#include "fm_op_kernel.h"
+#include "controllers.h"
+#include "fm_core.h"
+
+class EngineOpl : public FmCore {
+public:
+    EngineOpl() {};
+    ~EngineOpl() {};
+    void render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_shift) override;
+    void compute(int32_t *output, const int32_t *input, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, bool add);
+    void compute_pure(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, bool add);
+    void compute_fb(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, int32_t *fb_buf, int32_t fb_gain, bool add);
+};
+
+#endif  // ENGINEOPL_H_INCLUDED

third-party/Synth_Dexed/PluginFx.cpp

@@ -0,0 +1,233 @@
+/**
+
+   Copyright (c) 2013-2014 Pascal Gauthier.
+   Copyright (c) 2013-2014 Filatov Vadim.
+
+   Filter taken from the Obxd project :
+     https://github.com/2DaT/Obxd
+
+   This program is free software; you can redistribute it and/or modify
+   it under the terms of the GNU General Public License as published by
+   the Free Software Foundation; either version 3 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 General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software Foundation,
+   Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
+
+*/
+
+#define _USE_MATH_DEFINES
+#include <math.h>
+#include "PluginFx.h"
+#include "synth.h"
+
+const float dc = 1e-18;
+
+inline static float tptpc(float& state, float inp, float cutoff) {
+  float v = (inp - state) * cutoff / (1 + cutoff);
+  float res = v + state;
+  state = res + v;
+  return res;
+}
+
+inline static float tptlpupw(float & state , float inp , float cutoff , float srInv) {
+  cutoff = (cutoff * srInv) * M_PI;
+  float v = (inp - state) * cutoff / (1 + cutoff);
+  float res = v + state;
+  state = res + v;
+  return res;
+}
+
+//static float linsc(float param,const float min,const float max) {
+//    return (param) * (max - min) + min;
+//}
+
+static float logsc(float param, const float min, const float max, const float rolloff = 19.0f) {
+  return ((EXP_FUNC(param * LOG_FUNC(rolloff + 1)) - 1.0f) / (rolloff)) * (max - min) + min;
+}
+
+PluginFx::PluginFx() {
+  Cutoff = 1.0;
+  Reso = 0.0;
+  Gain = 1.0;
+}
+
+void PluginFx::init(uint16_t sr) {
+  mm = 0;
+  s1 = s2 = s3 = s4 = c = d = 0;
+  R24 = 0;
+
+  mmch = (int)(mm * 3);
+  mmt = mm * 3 - mmch;
+
+  sampleRate = sr;
+  sampleRateInv = 1 / sampleRate;
+#if defined(ARM_SQRT_FUNC)
+  float rcrate; ARM_SQRT_FUNC(44000 / sampleRate, &rcrate);
+#else
+  float rcrate = SQRT_FUNC((44000 / sampleRate));
+#endif
+  rcor24 = (970.0 / 44000) * rcrate;
+  rcor24Inv = 1 / rcor24;
+
+  bright = tanf((sampleRate * 0.5f - 10) * M_PI * sampleRateInv);
+
+  R = 1;
+  rcor = (480.0 / 44000) * rcrate;
+  rcorInv = 1 / rcor;
+  bandPassSw = false;
+
+  pCutoff = -1;
+  pReso = -1;
+
+  dc_r = 1.0 - (126.0 / sr);
+  dc_id = 0;
+  dc_od = 0;
+}
+
+inline float PluginFx::NR24(float sample, float g, float lpc) {
+  float ml = 1 / (1 + g);
+  float S = (lpc * (lpc * (lpc * s1 + s2) + s3) + s4) * ml;
+  float G = lpc * lpc * lpc * lpc;
+  float y = (sample - R24 * S) / (1 + R24 * G);
+  return y + 1e-8;
+};
+
+inline float PluginFx::NR(float sample, float g) {
+  float y = ((sample - R * s1 * 2 - g * s1  - s2) / (1 + g * (2 * R + g))) + dc;
+  return y;
+}
+
+void PluginFx::process(float *work, uint16_t sampleSize) {
+  // very basic DC filter
+  float t_fd = work[0];
+  work[0] = work[0] - dc_id + dc_r * dc_od;
+  dc_id = t_fd;
+  for (int i = 1; i < sampleSize; i++) {
+    t_fd = work[i];
+    work[i] = work[i] - dc_id + dc_r * work[i - 1];
+    dc_id = t_fd;
+  }
+
+  dc_od = work[sampleSize - 1];
+
+  // Gain
+  if (Gain == 0.0)
+  {
+    for (uint16_t i = 0; i < sampleSize; i++ )
+      work[i] = 0.0;
+  }
+  else if ( Gain != 1.0)
+  {
+    for (uint16_t i = 0; i < sampleSize; i++ )
+      work[i] *= Gain;
+  }
+
+  // don't apply the LPF if the cutoff is to maximum
+  if ( Cutoff == 1.0 )
+    return;
+
+  if ( Cutoff != pCutoff || Reso != pReso ) {
+    rReso = (0.991 - logsc(1 - Reso, 0, 0.991));
+    R24 = 3.5 * rReso;
+
+    float cutoffNorm = logsc(Cutoff, 60, 19000);
+    rCutoff = (float)tanf(cutoffNorm * sampleRateInv * M_PI);
+
+    pCutoff = Cutoff;
+    pReso = Reso;
+
+    R = 1 - rReso;
+  }
+
+  // THIS IS MY FAVORITE 4POLE OBXd filter
+
+  // maybe smooth this value
+  float g = rCutoff;
+  float lpc = g / (1 + g);
+
+  for (uint16_t i = 0; i < sampleSize; i++ ) {
+    float s = work[i];
+    s = s - 0.45 * tptlpupw(c, s, 15, sampleRateInv);
+    s = tptpc(d, s, bright);
+
+    float y0 = NR24(s, g, lpc);
+
+    //first low pass in cascade
+    float v = (y0 - s1) * lpc;
+    float res = v + s1;
+    s1 = res + v;
+
+    //damping
+    s1 = atanf(s1 * rcor24) * rcor24Inv;
+    float y1 = res;
+    float y2 = tptpc(s2, y1, g);
+    float y3 = tptpc(s3, y2, g);
+    float y4 = tptpc(s4, y3, g);
+    float mc = 0.0;
+
+    switch (mmch) {
+      case 0:
+        mc = ((1 - mmt) * y4 + (mmt) * y3);
+        break;
+      case 1:
+        mc = ((1 - mmt) * y3 + (mmt) * y2);
+        break;
+      case 2:
+        mc = ((1 - mmt) * y2 + (mmt) * y1);
+        break;
+      case 3:
+        mc = y1;
+        break;
+    }
+
+    //half volume comp
+    work[i] = mc * (1 + R24 * 0.45);
+  }
+}
+
+/*
+
+  // THIS IS THE 2POLE FILTER
+
+  for(int i=0; i < sampleSize; i++ ) {
+  float s = work[i];
+  s = s - 0.45*tptlpupw(c,s,15,sampleRateInv);
+  s = tptpc(d,s,bright);
+
+  //float v = ((sample- R * s1*2 - g2*s1 - s2)/(1+ R*g1*2 + g1*g2));
+  float v = NR(s,g);
+  float y1 = v*g + s1;
+  //damping
+  s1 = atanf(s1 * rcor) * rcorInv;
+
+  float y2 = y1*g + s2;
+  s2 = y2 + y1*g;
+
+  float mc;
+  if(!bandPassSw)
+  mc = (1-mm)*y2 + (mm)*v;
+  else
+  {
+
+  mc =2 * ( mm < 0.5 ?
+  ((0.5 - mm) * y2 + (mm) * y1):
+  ((1-mm) * y1 + (mm-0.5) * v)
+  );
+  }
+
+  work[i] = mc;
+  }
+
+*/
+
+float PluginFx::getGain(void)
+{
+  return (Gain);
+}

third-party/Synth_Dexed/PluginFx.h

@@ -0,0 +1,72 @@
+/**
+
+   Copyright (c) 2013 Pascal Gauthier.
+
+   This program is free software; you can redistribute it and/or modify
+   it under the terms of the GNU General Public License as published by
+   the Free Software Foundation; either version 3 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 General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software Foundation,
+   Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
+
+*/
+
+#pragma once
+
+#include "stdint.h"
+
+class PluginFx {
+    float s1, s2, s3, s4;
+    float sampleRate;
+    float sampleRateInv;
+    float d, c;
+    float R24;
+    float rcor24, rcor24Inv;
+    float bright;
+
+    // 24 db multimode
+    float mm;
+    float mmt;
+    int32_t mmch;
+    inline float NR24(float sample, float g, float lpc);
+
+    // preprocess values taken the UI
+    float rCutoff;
+    float rReso;
+    float rGain;
+
+    // thread values; if these are different from the UI,
+    // it needs to be recalculated.
+    float pReso;
+    float pCutoff;
+    float pGain;
+
+    // I am still keeping the 2pole w/multimode filter
+    inline float NR(float sample, float g);
+    bool bandPassSw;
+    float rcor, rcorInv;
+    int32_t R;
+
+    float dc_id;
+    float dc_od;
+    float dc_r;
+
+  public:
+    PluginFx();
+
+    // this is set directly by the ui / parameter
+    float Cutoff;
+    float Reso;
+    float Gain;
+
+    void init(uint16_t sampleRate);
+    void process(float *work, uint16_t sampleSize);
+    float getGain(void);
+};

third-party/Synth_Dexed/aligned_buf.h

@@ -0,0 +1,37 @@
+/*
+   Copyright 2013 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+// A convenient wrapper for buffers with alignment constraints
+
+// Note that if we were on C++11, we'd use aligned_storage or somesuch.
+
+#ifndef __ALIGNED_BUF_H
+#define __ALIGNED_BUF_H
+
+#include <stdlib.h>
+#include <stdint.h>
+
+template<typename T, size_t size, size_t alignment = 16>
+class AlignedBuf {
+  public:
+    T *get() {
+      return (T *)((((intptr_t)storage_) + alignment - 1) & -alignment);
+    }
+  private:
+    unsigned char storage_[size * sizeof(T) + alignment];
+};
+
+#endif

third-party/Synth_Dexed/compressor.h

@@ -0,0 +1,429 @@
+/* From https://github.com/chipaudette/OpenAudio_ArduinoLibrary */
+
+/*
+   AudioEffectCompressor
+
+   Created: Chip Audette, Dec 2016 - Jan 2017
+   Purpose; Apply dynamic range compression to the audio stream.
+            Assumes floating-point data.
+
+   This processes a single stream fo audio data (ie, it is mono)
+
+   MIT License.  use at your own risk.
+*/
+
+#ifndef _COMPRESSOR_H
+#define _COMPRESSOR_H
+
+#ifndef TEENSYDUINO
+
+#include <arm_math.h> //ARM DSP extensions.  https://www.keil.com/pack/doc/CMSIS/DSP/html/index.html
+#include "synth.h"
+
+/*
+static const float zeroblock_f32[] = {
+    0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
+#if AUDIO_BLOCK_SAMPLES > 16
+    0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
+#endif
+#if AUDIO_BLOCK_SAMPLES > 32
+    0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
+#endif
+#if AUDIO_BLOCK_SAMPLES > 48
+    0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
+#endif
+#if AUDIO_BLOCK_SAMPLES > 64
+    0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
+#endif
+#if AUDIO_BLOCK_SAMPLES > 80
+    0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
+#endif
+#if AUDIO_BLOCK_SAMPLES > 96
+    0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
+#endif
+#if AUDIO_BLOCK_SAMPLES > 112
+    0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
+#endif
+#if AUDIO_BLOCK_SAMPLES > 128
+#error AUDIO_BLOCK_SAMPLES > 128 is a problem for this class
+#endif
+};
+*/
+
+class Compressor
+{
+  //GUI: inputs:1, outputs:1  //this line used for automatic generation of GUI node
+  public:
+    //constructor
+    Compressor(const float sample_rate_Hz) {
+	  //setDefaultValues(AUDIO_SAMPLE_RATE);   resetStates();
+	  setDefaultValues(sample_rate_Hz);
+          resetStates();
+    };
+	
+    void setDefaultValues(const float sample_rate_Hz) {
+      setThresh_dBFS(-20.0f);     //set the default value for the threshold for compression
+      setCompressionRatio(5.0f);  //set the default copression ratio
+      setAttack_sec(0.005f, sample_rate_Hz);  //default to this value
+      setRelease_sec(0.200f, sample_rate_Hz); //default to this value
+      setHPFilterCoeff();  enableHPFilter(true);  //enable the HP filter to remove any DC offset from the audio
+    }
+
+
+    //here's the method that does all the work
+    void doCompression(float *audio_block, uint16_t len) {
+      //Serial.println("AudioEffectGain_F32: updating.");  //for debugging.
+      if (!audio_block) {
+        printf("No audio_block available for Compressor!\n");
+        return;
+      }
+
+      //apply a high-pass filter to get rid of the DC offset
+      if (use_HP_prefilter)
+        arm_biquad_cascade_df1_f32(&hp_filt_struct, audio_block, audio_block, len);
+      
+      //apply the pre-gain...a negative gain value will disable
+      if (pre_gain > 0.0f)
+        arm_scale_f32(audio_block, pre_gain, audio_block, len); //use ARM DSP for speed!
+
+      //calculate the level of the audio (ie, calculate a smoothed version of the signal power)
+      float* audio_level_dB_block = float[len]; 
+      if(!audio_level_dB_block)
+      {
+        printf("Cannot allocate memory for \"audio_level_dB_block\" - stopping\n");
+        while(1);
+      }
+
+      //arm_copy_f32(zeroblock_f32,audio_level_dB_block,len);
+
+      if(audio_level_dB_block)
+      	calcAudioLevel_dB(audio_block, audio_level_dB_block, len); //returns through audio_level_dB_block
+
+      //compute the desired gain based on the observed audio level
+      float* gain_block=new float[len]; 
+      if(!gain_block)
+      {
+        printf("Cannot allocate memory for \"gain_block\" - stopping\n");
+        while(1);
+      }
+
+      //arm_copy_f32(zeroblock_f32,gain_block,len);
+
+      if(gain_block)
+      {
+      	calcGain(audio_level_dB_block, gain_block, len);  //returns through gain_block
+
+        //apply the desired gain...store the processed audio back into audio_block
+        arm_mult_f32(audio_block, gain_block, audio_block, len);
+      }
+
+      //release memory
+      if(audio_level_dB_block)
+	delete audio_level_dB_block;
+      if(gain_block)
+        delete gain_block;
+    }
+
+    // Here's the method that estimates the level of the audio (in dB)
+    // It squares the signal and low-pass filters to get a time-averaged
+    // signal power.  It then 
+    void calcAudioLevel_dB(float *wav_block, float *level_dB_block, uint16_t len) { 
+    	
+      // calculate the instantaneous signal power (square the signal)
+      float* wav_pow_block=new float[len]; 
+      if(!wav_pow_block)
+      {
+        printf("Cannot allocate memory for \"wav_pow_block\" - stopping\n");
+        while(1);
+      }
+
+      //arm_copy_f32(zeroblock_f32,wav_pow_block,len);
+
+      arm_mult_f32(wav_block, wav_block, wav_pow_block, len);
+
+      // low-pass filter and convert to dB
+      float c1 = level_lp_const, c2 = 1.0f - c1; //prepare constants
+      for (uint16_t i = 0; i < len; i++) {
+        // first-order low-pass filter to get a running estimate of the average power
+        wav_pow_block[i] = c1*prev_level_lp_pow + c2*wav_pow_block[i];
+        
+        // save the state of the first-order low-pass filter
+        prev_level_lp_pow = wav_pow_block[i]; 
+
+        //now convert the signal power to dB (but not yet multiplied by 10.0)
+        level_dB_block[i] = log10f_approx(wav_pow_block[i]);
+      }
+
+      //limit the amount that the state of the smoothing filter can go toward negative infinity
+      if (prev_level_lp_pow < (1.0E-13)) prev_level_lp_pow = 1.0E-13;  //never go less than -130 dBFS 
+
+      //scale the wav_pow_block by 10.0 to complete the conversion to dB
+      arm_scale_f32(level_dB_block, 10.0f, level_dB_block, len); //use ARM DSP for speed!
+
+      //release memory and return
+      if(wav_pow_block)
+        delete wav_pow_block;
+
+      return; //output is passed through level_dB_block
+    }
+
+    //This method computes the desired gain from the compressor, given an estimate
+    //of the signal level (in dB)
+    void calcGain(float *audio_level_dB_block, float *gain_block,uint16_t len) { 
+    
+      //first, calculate the instantaneous target gain based on the compression ratio
+      float* inst_targ_gain_dB_block=new float[len];
+      if(!inst_targ_gain_dB_block)
+      {
+	printf("Cannot allocate memory for \"inst_targ_gain_dB_block\" - stopping\n");
+	while(1);
+      }
+      //arm_copy_f32(zeroblock_f32,inst_targ_gain_dB_block,len);
+
+      calcInstantaneousTargetGain(audio_level_dB_block, inst_targ_gain_dB_block,len);
+    
+      //second, smooth in time (attack and release) by stepping through each sample
+      float *gain_dB_block = new float[len]; 
+      if(!gain_dB_block)
+      {
+	printf("Cannot allocate memory for \"gain_dB_block\" - stopping\n");
+	while(1);
+      }
+      //arm_copy_f32(zeroblock_f32,gain_dB_block,len);
+
+      calcSmoothedGain_dB(inst_targ_gain_dB_block,gain_dB_block, len);
+
+      //finally, convert from dB to linear gain: gain = 10^(gain_dB/20);  (ie this takes care of the sqrt, too!)
+      arm_scale_f32(gain_dB_block, 1.0f/20.0f, gain_dB_block, len);  //divide by 20 
+      for (uint16_t i = 0; i < len; i++) gain_block[i] = pow10f(gain_dB_block[i]); //do the 10^(x)
+      
+
+      //release memory and return
+      if(inst_targ_gain_dB_block)
+        delete inst_targ_gain_dB_block;
+      if(gain_dB_block)
+        delete gain_dB_block;
+
+      return;  //output is passed through gain_block
+    }
+      
+    //Compute the instantaneous desired gain, including the compression ratio and
+    //threshold for where the comrpession kicks in
+    void calcInstantaneousTargetGain(float *audio_level_dB_block, float *inst_targ_gain_dB_block, uint16_t len) {
+      
+      // how much are we above the compression threshold?
+      float* above_thresh_dB_block=new float[len];
+      if(!above_thresh_dB_block)
+      {
+        printf("Cannot allocate memory for \"above_thresh_dB_block\" - stopping\n");
+        while(1);
+      }
+
+      //arm_copy_f32(zeroblock_f32,above_thresh_dB_block,len);
+
+      arm_offset_f32(audio_level_dB_block,  //CMSIS DSP for "add a constant value to all elements"
+        -thresh_dBFS,                         //this is the value to be added
+        above_thresh_dB_block,          //this is the output
+        len);  
+
+      // scale by the compression ratio...this is what the output level should be (this is our target level)
+      arm_scale_f32(above_thresh_dB_block,    //CMSIS DSP for "multiply all elements by a constant value"
+           1.0f / comp_ratio,                       //this is the value to be multiplied 
+           inst_targ_gain_dB_block,           //this is the output
+           len); 
+
+      // compute the instantaneous gain...which is the difference between the target level and the original level
+      arm_sub_f32(inst_targ_gain_dB_block,  //CMSIS DSP for "subtract two vectors element-by-element"
+           above_thresh_dB_block,           //this is the vector to be subtracted
+           inst_targ_gain_dB_block,         //this is the output
+           len);
+
+      // limit the target gain to attenuation only (this part of the compressor should not make things louder!)
+      for (uint16_t i=0; i < len; i++) {
+        if (inst_targ_gain_dB_block[i] > 0.0f) inst_targ_gain_dB_block[i] = 0.0f;
+      }
+
+      // release memory before returning
+      if(above_thresh_dB_block)
+        delete above_thresh_dB_block;
+
+      return;  //output is passed through inst_targ_gain_dB_block
+    }
+
+    //this method applies the "attack" and "release" constants to smooth the
+    //target gain level through time.
+    void calcSmoothedGain_dB(float *inst_targ_gain_dB_block, float *gain_dB_block, uint16_t len) {
+      float gain_dB;
+      float one_minus_attack_const = 1.0f - attack_const;
+      float one_minus_release_const = 1.0f - release_const;
+      for (uint16_t i = 0; i < len; i++) {
+        gain_dB = inst_targ_gain_dB_block[i];
+
+        //smooth the gain using the attack or release constants
+        if (gain_dB < prev_gain_dB) {  //are we in the attack phase?
+          gain_dB_block[i] = attack_const*prev_gain_dB + one_minus_attack_const*gain_dB;
+        } else {   //or, we're in the release phase
+          gain_dB_block[i] = release_const*prev_gain_dB + one_minus_release_const*gain_dB;
+        }
+
+        //save value for the next time through this loop
+        prev_gain_dB = gain_dB_block[i];
+      }
+
+      return;  //the output here is gain_block
+    }
+
+
+    //methods to set parameters of this module
+    void resetStates(void) {
+      prev_level_lp_pow = 1.0f;
+      prev_gain_dB = 0.0f;
+      
+      //initialize the HP filter.  (This also resets the filter states,)
+      arm_biquad_cascade_df1_init_f32(&hp_filt_struct, hp_nstages, hp_coeff, hp_state);
+    }
+    void setPreGain(float g) {  pre_gain = g;  }
+    void setPreGain_dB(float gain_dB) { setPreGain(pow(10.0, gain_dB / 20.0));  }
+    void setCompressionRatio(float cr) {
+      comp_ratio = std::max(0.001f, cr); //limit to positive values
+      updateThresholdAndCompRatioConstants();
+    }
+    void setAttack_sec(float a, float fs_Hz) {
+      attack_sec = a;
+      attack_const = expf(-1.0f / (attack_sec * fs_Hz)); //expf() is much faster than exp()
+
+      //also update the time constant for the envelope extraction
+      setLevelTimeConst_sec(std::min(attack_sec,release_sec) / 5.0, fs_Hz);  //make the level time-constant one-fifth the gain time constants
+    } 
+    void setRelease_sec(float r, float fs_Hz) {
+      release_sec = r;
+      release_const = expf(-1.0f / (release_sec * fs_Hz)); //expf() is much faster than exp()
+
+      //also update the time constant for the envelope extraction
+      setLevelTimeConst_sec(std::min(attack_sec,release_sec) / 5.0, fs_Hz);  //make the level time-constant one-fifth the gain time constants
+    }
+    void setLevelTimeConst_sec(float t_sec, float fs_Hz) {
+      const float min_t_sec = 0.002f;  //this is the minimum allowed value
+      level_lp_sec = std::max(min_t_sec,t_sec);
+      level_lp_const = expf(-1.0f / (level_lp_sec * fs_Hz)); //expf() is much faster than exp()
+    }
+    void setThresh_dBFS(float val) { 
+      thresh_dBFS = val;
+      setThreshPow(pow(10.0, thresh_dBFS / 10.0));
+    }
+    void enableHPFilter(boolean flag) { use_HP_prefilter = flag; };
+
+    //methods to return information about this module
+    float getPreGain_dB(void) { return 20.0 * log10f_approx(pre_gain);  }
+    float getAttack_sec(void) {  return attack_sec; }
+    float getRelease_sec(void) {  return release_sec; }
+    float getLevelTimeConst_sec(void) { return level_lp_sec; }
+    float getThresh_dBFS(void) { return thresh_dBFS; }
+    float getCompressionRatio(void) { return comp_ratio; }
+    float getCurrentLevel_dBFS(void) { return 10.0* log10f_approx(prev_level_lp_pow); }
+    float getCurrentGain_dB(void) { return prev_gain_dB; }
+
+    void setHPFilterCoeff_N2IIR_Matlab(float b[], float a[]){
+      //https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
+      //Use matlab to compute the coeff for HP at 20Hz: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
+      hp_coeff[0] = b[0];   hp_coeff[1] = b[1];  hp_coeff[2] = b[2]; //here are the matlab "b" coefficients
+      hp_coeff[3] = -a[1];  hp_coeff[4] = -a[2];  //the DSP needs the "a" terms to have opposite sign vs Matlab    	
+    }
+    
+  private:
+    //state-related variables
+    float *inputQueueArray_f32[1]; //memory pointer for the input to this module
+    float prev_level_lp_pow = 1.0;
+    float prev_gain_dB = 0.0; //last gain^2 used
+
+    //HP filter state-related variables
+    arm_biquad_casd_df1_inst_f32 hp_filt_struct;
+    static const uint8_t hp_nstages = 1;
+    float hp_coeff[5 * hp_nstages] = {1.0, 0.0, 0.0, 0.0, 0.0}; //no filtering. actual filter coeff set later
+    float hp_state[4 * hp_nstages];
+    void setHPFilterCoeff(void) {
+      //https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
+      //Use matlab to compute the coeff for HP at 20Hz: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
+      float b[] = {9.979871156751189e-01,    -1.995974231350238e+00, 9.979871156751189e-01};  //from Matlab
+      float a[] = { 1.000000000000000e+00,    -1.995970179642828e+00,    9.959782830576472e-01};  //from Matlab
+      setHPFilterCoeff_N2IIR_Matlab(b, a);
+      //hp_coeff[0] = b[0];   hp_coeff[1] = b[1];  hp_coeff[2] = b[2]; //here are the matlab "b" coefficients
+      //hp_coeff[3] = -a[1];  hp_coeff[4] = -a[2];  //the DSP needs the "a" terms to have opposite sign vs Matlab
+    }
+
+
+    //private parameters related to gain calculation
+    float attack_const, release_const, level_lp_const; //used in calcGain().  set by setAttack_sec() and setRelease_sec();
+    float comp_ratio_const, thresh_pow_FS_wCR;  //used in calcGain();  set in updateThresholdAndCompRatioConstants()
+    void updateThresholdAndCompRatioConstants(void) {
+      comp_ratio_const = 1.0f-(1.0f / comp_ratio);
+      thresh_pow_FS_wCR = powf(thresh_pow_FS, comp_ratio_const);    
+    }
+
+    //settings
+    float attack_sec, release_sec, level_lp_sec; 
+    float thresh_dBFS = 0.0;  //threshold for compression, relative to digital full scale
+    float thresh_pow_FS = 1.0f;  //same as above, but not in dB
+    void setThreshPow(float t_pow) { 
+      thresh_pow_FS = t_pow;
+      updateThresholdAndCompRatioConstants();
+    }
+    float comp_ratio = 1.0;  //compression ratio
+    float pre_gain = -1.0;  //gain to apply before the compression.  negative value disables
+    boolean use_HP_prefilter;
+    
+    
+    // Accelerate the powf(10.0,x) function
+    static float pow10f(float x) {
+      //return powf(10.0f,x)   //standard, but slower
+      return expf(2.302585092994f*x);  //faster:  exp(log(10.0f)*x)
+    }
+
+    // Accelerate the log10f(x)  function?
+    static float log10f_approx(float x) {
+      //return log10f(x);   //standard, but slower
+      return log2f_approx(x)*0.3010299956639812f; //faster:  log2(x)/log2(10)
+    }
+    
+    /* ----------------------------------------------------------------------
+    ** Fast approximation to the log2() function.  It uses a two step
+    ** process.  First, it decomposes the floating-point number into
+    ** a fractional component F and an exponent E.  The fraction component
+    ** is used in a polynomial approximation and then the exponent added
+    ** to the result.  A 3rd order polynomial is used and the result
+    ** when computing db20() is accurate to 7.984884e-003 dB.
+    ** ------------------------------------------------------------------- */
+    //https://community.arm.com/tools/f/discussions/4292/cmsis-dsp-new-functionality-proposal/22621#22621
+    //float log2f_approx_coeff[4] = {1.23149591368684f, -4.11852516267426f, 6.02197014179219f, -3.13396450166353f};
+    static float log2f_approx(float X) {
+      //float *C = &log2f_approx_coeff[0];
+      float Y;
+      float F;
+      int32_t E;
+    
+      // This is the approximation to log2()
+      F = frexpf(fabsf(X), &E);
+      //  Y = C[0]*F*F*F + C[1]*F*F + C[2]*F + C[3] + E;
+      //Y = *C++;
+      Y = 1.23149591368684f;
+      Y *= F;
+      //Y += (*C++);
+      Y += -4.11852516267426f;
+      Y *= F;
+      //Y += (*C++);
+      Y += 6.02197014179219f;
+      Y *= F;
+      //Y += (*C++);
+      Y += -3.13396450166353f;
+      Y += E;
+    
+      return(Y);
+    }
+    
+    
+};
+
+#else
+#warning USING TEENSYDUINO SO INTERNAL COMPRESSOR IS DISABLED!
+#endif // TEENSYDUINO
+#endif

third-party/Synth_Dexed/controllers.h

@@ -0,0 +1,152 @@
+/*
+   Copyright 2013 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#ifndef __CONTROLLERS_H
+#define __CONTROLLERS_H
+
+#include <stdint.h>
+#include "synth.h"
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+#include "fm_core.h"
+
+// State of MIDI controllers
+const uint8_t kControllerPitch = 0;
+const uint8_t kControllerPitchRange = 1;
+const uint8_t kControllerPitchStep = 2;
+const uint8_t kControllerPortamentoGlissando = 3;
+
+class FmMod {
+  public:
+    uint8_t range;
+    bool pitch;
+    bool amp;
+    bool eg;
+    uint8_t ctrl_mode;
+
+    FmMod()
+    {
+      range = 0;
+      ctrl_mode = 0;
+      pitch = false;
+      amp = false;
+      eg = false;
+    }
+
+    void setRange(uint8_t r)
+    {
+      range = r < 0 || r > 99 ? 0 : r;
+    }
+
+    uint8_t getRange(void)
+    {
+      return (range);
+    }
+
+    void setTarget(uint8_t assign)
+    {
+      assign = assign < 0 && assign > 7 ? 0 : assign;
+      pitch = assign & 1; // PITCH
+      amp = assign & 2; // AMP
+      eg = assign & 4; // EG
+    }
+
+    uint8_t getTarget(void)
+    {
+      return (pitch & amp & eg);
+    }
+
+    void setMode(uint8_t m)
+    {
+      ctrl_mode = m > MIDI_CONTROLLER_MODE_MAX ? 0 : m;
+    }
+};
+
+class Controllers {
+    void applyMod(int cc, FmMod &mod)
+    {
+      uint8_t total = 0;
+      float range = mod.range / 100.0;
+
+      switch (mod.ctrl_mode)
+      {
+        case 0:
+          total = uint8_t(float(cc) * range); // LINEAR mode
+          break;
+        case 1:
+          total = uint8_t(127.0 * range - (float(cc) * range)); // REVERSE mode
+          break;
+        case 2:
+          total = uint8_t(range * float(cc) + (1.0 - range) * 127.0); // DIRECT BC mode by Thierry (opus.quatre)
+          break;
+      }
+
+      if (mod.amp)
+        amp_mod = std::max(amp_mod, total);
+
+      if (mod.pitch)
+        pitch_mod = std::max(pitch_mod, total);
+
+      if (mod.eg)
+        eg_mod = std::max(eg_mod, total);
+    }
+
+  public:
+    int32_t values_[4];
+
+    uint8_t amp_mod;
+    uint8_t pitch_mod;
+    uint8_t eg_mod;
+
+    uint8_t aftertouch_cc;
+    uint8_t breath_cc;
+    uint8_t foot_cc;
+    uint8_t modwheel_cc;
+    bool portamento_enable_cc;
+    int32_t portamento_cc;
+    bool portamento_gliss_cc;
+    int32_t masterTune;
+
+    uint8_t opSwitch;
+
+    FmMod wheel;
+    FmMod foot;
+    FmMod breath;
+    FmMod at;
+
+    Controllers() {
+      amp_mod = 0;
+      pitch_mod = 0;
+      eg_mod = 0;
+    }
+
+    void refresh() {
+      amp_mod = pitch_mod = eg_mod = 0;
+
+      applyMod(modwheel_cc, wheel);
+      applyMod(breath_cc, breath);
+      applyMod(foot_cc, foot);
+      applyMod(aftertouch_cc, at);
+
+      if ( ! ((wheel.eg || foot.eg) || (breath.eg || at.eg)) )
+        eg_mod = 127;
+    }
+
+    class FmCore* core;
+};
+
+#endif

third-party/Synth_Dexed/dexed.h

@@ -0,0 +1,391 @@
+/*
+   MicroDexed
+
+   MicroDexed is a port of the Dexed sound engine
+   (https://github.com/asb2m10/dexed) for the Teensy-3.5/3.6/4.x with audio shield.
+   Dexed ist heavily based on https://github.com/google/music-synthesizer-for-android
+
+   (c)2018-2021 H. Wirtz <wirtz@parasitstudio.de>
+
+   This program is free software; you can redistribute it and/or modify
+   it under the terms of the GNU General Public License as published by
+   the Free Software Foundation; either version 3 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 General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software Foundation,
+   Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
+
+*/
+
+//#define DISABLE_DEXED_COMPRESSOR 1
+
+#ifndef DEXED_H_INCLUDED
+#define DEXED_H_INCLUDED
+
+#include <stdint.h>
+#include <cstdlib>
+#include <arm_math.h>
+#if defined(TEENSYDUINO)
+#include <Audio.h>
+#endif
+#include "fm_op_kernel.h"
+#include "synth.h"
+#include "env.h"
+#include "aligned_buf.h"
+#include "pitchenv.h"
+#include "controllers.h"
+#include "dx7note.h"
+#include "lfo.h"
+#include "PluginFx.h"
+#include "compressor.h"
+#include "EngineMsfa.h"
+#include "EngineMkI.h"
+#include "EngineOpl.h"
+
+#define NUM_VOICE_PARAMETERS 156
+
+struct ProcessorVoice {
+  uint8_t midi_note;
+  uint8_t velocity;
+  int16_t porta;
+  bool keydown;
+  bool sustained;
+  bool live;
+  uint32_t key_pressed_timer;
+  Dx7Note *dx7_note;
+};
+
+enum DexedVoiceOPParameters {
+  DEXED_OP_EG_R1,           // 0
+  DEXED_OP_EG_R2,           // 1
+  DEXED_OP_EG_R3,           // 2
+  DEXED_OP_EG_R4,           // 3
+  DEXED_OP_EG_L1,           // 4
+  DEXED_OP_EG_L2,           // 5
+  DEXED_OP_EG_L3,           // 6
+  DEXED_OP_EG_L4,           // 7
+  DEXED_OP_LEV_SCL_BRK_PT,  // 8
+  DEXED_OP_SCL_LEFT_DEPTH,  // 9
+  DEXED_OP_SCL_RGHT_DEPTH,  // 10
+  DEXED_OP_SCL_LEFT_CURVE,  // 11
+  DEXED_OP_SCL_RGHT_CURVE,  // 12
+  DEXED_OP_OSC_RATE_SCALE,  // 13
+  DEXED_OP_AMP_MOD_SENS,    // 14
+  DEXED_OP_KEY_VEL_SENS,    // 15
+  DEXED_OP_OUTPUT_LEV,      // 16
+  DEXED_OP_OSC_MODE,        // 17
+  DEXED_OP_FREQ_COARSE,     // 18
+  DEXED_OP_FREQ_FINE,       // 19
+  DEXED_OP_OSC_DETUNE       // 20
+};
+
+#define DEXED_VOICE_OFFSET 126
+enum DexedVoiceParameters {
+  DEXED_PITCH_EG_R1,        // 0
+  DEXED_PITCH_EG_R2,        // 1
+  DEXED_PITCH_EG_R3,        // 2
+  DEXED_PITCH_EG_R4,        // 3
+  DEXED_PITCH_EG_L1,        // 4
+  DEXED_PITCH_EG_L2,        // 5
+  DEXED_PITCH_EG_L3,        // 6
+  DEXED_PITCH_EG_L4,        // 7
+  DEXED_ALGORITHM,          // 8
+  DEXED_FEEDBACK,           // 9
+  DEXED_OSC_KEY_SYNC,       // 10
+  DEXED_LFO_SPEED,          // 11
+  DEXED_LFO_DELAY,          // 12
+  DEXED_LFO_PITCH_MOD_DEP,  // 13
+  DEXED_LFO_AMP_MOD_DEP,    // 14
+  DEXED_LFO_SYNC,           // 15
+  DEXED_LFO_WAVE,           // 16
+  DEXED_LFO_PITCH_MOD_SENS, // 17
+  DEXED_TRANSPOSE,          // 18
+  DEXED_NAME                // 19
+};
+
+enum ADSR {
+  ATTACK,
+  DECAY,
+  SUSTAIN,
+  RELEASE
+};
+
+enum OPERATORS {
+  OP1,
+  OP2,
+  OP3,
+  OP4,
+  OP5,
+  OP6
+};
+
+enum CONTROLLER_ASSIGN {
+  NONE,
+  PITCH,
+  AMP,
+  PITCH_AMP,
+  EG,
+  PITCH_EG,
+  AMP_EG,
+  PITCH_AMP_EG
+};
+
+enum PORTAMENTO_MODE {
+  RETAIN,
+  FOLLOW
+};
+
+enum ON_OFF {
+  OFF,
+  ON
+};
+
+enum VELOCITY_SCALES {
+  MIDI_VELOCITY_SCALING_OFF,
+  MIDI_VELOCITY_SCALING_DX7,
+  MIDI_VELOCITY_SCALING_DX7II
+};
+
+enum ENGINES {
+  MSFA,
+  MKI,
+  OPL
+};
+
+// GLOBALS
+
+//==============================================================================
+
+class Dexed
+{
+  public:
+    Dexed(uint8_t maxnotes, uint16_t rate);
+    ~Dexed();
+
+    // Global methods
+    void activate(void);
+    void deactivate(void);
+    bool getMonoMode(void);
+    void setMonoMode(bool mode);
+    void setNoteRefreshMode(bool mode);
+    uint8_t getMaxNotes(void);
+    void doRefreshVoice(void);
+    void setOPAll(uint8_t ops);
+    bool decodeVoice(uint8_t* data, uint8_t* encoded_data);
+    bool encodeVoice(uint8_t* encoded_data);
+    bool getVoiceData(uint8_t* data_copy);
+    void setVoiceDataElement(uint8_t address, uint8_t value);
+    uint8_t getVoiceDataElement(uint8_t address);
+    void loadInitVoice(void);
+    void loadVoiceParameters(uint8_t* data);
+    uint8_t getNumNotesPlaying(void);
+    uint32_t getXRun(void);
+    uint16_t getRenderTimeMax(void);
+    void resetRenderTimeMax(void);
+    void ControllersRefresh(void);
+    void setVelocityScale(uint8_t offset, uint8_t max);
+    void getVelocityScale(uint8_t* offset, uint8_t* max);
+    void setVelocityScale(uint8_t setup);
+    void setMaxNotes(uint8_t n);
+    void setEngineType(uint8_t engine);
+    uint8_t getEngineType(void);
+    FmCore* getEngineAddress(void);
+#ifndef TEENSYDUINO
+    void setCompressor(bool comp);
+    bool getCompressor(void);
+    void setCompressorPreGain_dB(float pre_gain);
+    void setCompressorAttack_sec(float attack_sec);
+    void setCompressorRelease_sec(float release_sec);
+    void setCompressorThresh_dBFS(float thresh_dBFS);
+    void setCompressionRatio(float comp_ratio);
+    float getCompressorPreGain_dB(void);
+    float getCompressorAttack_sec(void);
+    float getCompressorRelease_sec(void);
+    float getCompressorThresh_dBFS(void);
+    float getCompressionRatio(void);
+#endif
+    int16_t checkSystemExclusive(const uint8_t* sysex, const uint16_t len);
+
+    // Sound methods
+    void keyup(uint8_t pitch);
+    void keydown(uint8_t pitch, uint8_t velo);
+    void setSustain(bool sustain);
+    bool getSustain(void);
+    void panic(void);
+    void notesOff(void);
+    void resetControllers(void);
+    void setMasterTune(int8_t mastertune);
+    int8_t getMasterTune(void);
+    void setPortamento(uint8_t portamento_mode, uint8_t portamento_glissando, uint8_t portamento_time);
+    void setPortamentoMode(uint8_t portamento_mode);
+    uint8_t getPortamentoMode(void);
+    void setPortamentoGlissando(uint8_t portamento_glissando);
+    uint8_t getPortamentoGlissando(void);
+    void setPortamentoTime(uint8_t portamento_time);
+    uint8_t getPortamentoTime(void);
+    void setPBController(uint8_t pb_range, uint8_t pb_step);
+    void setMWController(uint8_t mw_range, uint8_t mw_assign, uint8_t mw_mode);
+    void setFCController(uint8_t fc_range, uint8_t fc_assign, uint8_t fc_mode);
+    void setBCController(uint8_t bc_range, uint8_t bc_assign, uint8_t bc_mode);
+    void setATController(uint8_t at_range, uint8_t at_assign, uint8_t at_mode);
+    void setModWheel(uint8_t value);
+    uint8_t getModWheel(void);
+    void setBreathController(uint8_t value);
+    uint8_t getBreathController(void);
+    void setFootController(uint8_t value);
+    uint8_t getFootController(void);
+    void setAftertouch(uint8_t value);
+    uint8_t getAftertouch(void);
+    void setPitchbend(uint8_t value1, uint8_t value2);
+    void setPitchbend(int16_t value);
+    void setPitchbend(uint16_t value);
+    int16_t getPitchbend(void);
+    void setPitchbendRange(uint8_t range);
+    uint8_t getPitchbendRange(void);
+    void setPitchbendStep(uint8_t step);
+    uint8_t getPitchbendStep(void);
+    void setModWheelRange(uint8_t range);
+    uint8_t getModWheelRange(void);
+    void setModWheelTarget(uint8_t target);
+    uint8_t getModWheelTarget(void);
+    void setFootControllerRange(uint8_t range);
+    uint8_t getFootControllerRange(void);
+    void setFootControllerTarget(uint8_t target);
+    uint8_t getFootControllerTarget(void);
+    void setBreathControllerRange(uint8_t range);
+    uint8_t getBreathControllerRange(void);
+    void setBreathControllerTarget(uint8_t target);
+    uint8_t getBreathControllerTarget(void);
+    void setAftertouchRange(uint8_t range);
+    uint8_t getAftertouchRange(void);
+    void setAftertouchTarget(uint8_t target);
+    uint8_t getAftertouchTarget(void);
+    void setFilterCutoff(float cutoff);
+    float getFilterCutoff(void);
+    void setFilterResonance(float resonance);
+    float getFilterResonance(void);
+    void setGain(float gain);
+    float getGain(void);
+
+    // Voice configuration methods
+    void setOPRateAll(uint8_t rate);
+    void setOPLevelAll(uint8_t level);
+    void setOPRateAllCarrier(uint8_t step, uint8_t rate);
+    void setOPLevelAllCarrier(uint8_t step, uint8_t level);
+    void setOPRateAllModulator(uint8_t step, uint8_t rate);
+    void setOPLevelAllModulator(uint8_t step, uint8_t level);
+    void setOPRate(uint8_t op, uint8_t step, uint8_t rate);
+    uint8_t getOPRate(uint8_t op, uint8_t step);
+    void setOPLevel(uint8_t op, uint8_t step, uint8_t level);
+    uint8_t getOPLevel(uint8_t op, uint8_t step);
+    void setOPKeyboardLevelScalingBreakPoint(uint8_t op, uint8_t level);
+    uint8_t getOPKeyboardLevelScalingBreakPoint(uint8_t op);
+    void setOPKeyboardLevelScalingDepthLeft(uint8_t op, uint8_t depth);
+    uint8_t getOPKeyboardLevelScalingDepthLeft(uint8_t op);
+    void setOPKeyboardLevelScalingDepthRight(uint8_t op, uint8_t depth);
+    uint8_t getOPKeyboardLevelScalingDepthRight(uint8_t op);
+    void setOPKeyboardLevelScalingCurveLeft(uint8_t op, uint8_t curve);
+    uint8_t getOPKeyboardLevelScalingCurveLeft(uint8_t op);
+    void setOPKeyboardLevelScalingCurveRight(uint8_t op, uint8_t curve);
+    uint8_t getOPKeyboardLevelScalingCurveRight(uint8_t op);
+    void setOPKeyboardRateScale(uint8_t op, uint8_t scale);
+    uint8_t getOPKeyboardRateScale(uint8_t op);
+    void setOPAmpModulationSensity(uint8_t op, uint8_t sensitivity);
+    uint8_t getOPAmpModulationSensity(uint8_t op);
+    void setOPKeyboardVelocitySensity(uint8_t op, uint8_t sensitivity);
+    uint8_t getOPKeyboardVelocitySensity(uint8_t op);
+    void setOPOutputLevel(uint8_t op, uint8_t level);
+    uint8_t getOPOutputLevel(uint8_t op);
+    void setOPMode(uint8_t op, uint8_t mode);
+    uint8_t getOPMode(uint8_t op);
+    void setOPFrequencyCoarse(uint8_t op, uint8_t frq_coarse);
+    uint8_t getOPFrequencyCoarse(uint8_t op);
+    void setOPFrequencyFine(uint8_t op, uint8_t frq_fine);
+    uint8_t getOPFrequencyFine(uint8_t op);
+    void setOPDetune(uint8_t op, uint8_t detune);
+    uint8_t getOPDetune(uint8_t op);
+    void setPitchRate(uint8_t step, uint8_t rate);
+    uint8_t getPitchRate(uint8_t step);
+    void setPitchLevel(uint8_t step, uint8_t level);
+    uint8_t getPitchLevel(uint8_t step);
+    void setAlgorithm(uint8_t algorithm);
+    uint8_t getAlgorithm(void);
+    void setFeedback(uint8_t feedback);
+    uint8_t getFeedback(void);
+    void setOscillatorSync(bool sync);
+    bool getOscillatorSync(void);
+    void setLFOSpeed(uint8_t speed);
+    uint8_t getLFOSpeed(void);
+    void setLFODelay(uint8_t delay);
+    uint8_t getLFODelay(void);
+    void setLFOPitchModulationDepth(uint8_t depth);
+    uint8_t getLFOPitchModulationDepth(void);
+    void setLFOAmpModulationDepth(uint8_t delay);
+    uint8_t getLFOAmpModulationDepth(void);
+    void setLFOSync(bool sync);
+    bool getLFOSync(void);
+    void setLFOWaveform(uint8_t waveform);
+    uint8_t getLFOWaveform(void);
+    void setLFOPitchModulationSensitivity(uint8_t sensitivity);
+    uint8_t getLFOPitchModulationSensitivity(void);
+    void setTranspose(uint8_t transpose);
+    uint8_t getTranspose(void);
+    void setName(char* name);
+    void getName(char* buffer);
+
+    ProcessorVoice* voices;
+
+  protected:
+    uint8_t init_voice[NUM_VOICE_PARAMETERS] = {
+      99, 99, 99, 99, 99, 99, 99, 00, 33, 00, 00, 00, 00, 00, 00, 00, 00, 00, 01, 00, 00, // OP6 eg_rate_1-4, level_1-4, kbd_lev_scl_brk_pt, kbd_lev_scl_lft_depth, kbd_lev_scl_rht_depth, kbd_lev_scl_lft_curve, kbd_lev_scl_rht_curve, kbd_rate_scaling, amp_mod_sensitivity, key_vel_sensitivity, operator_output_level, osc_mode, osc_freq_coarse, osc_freq_fine, osc_detune
+      99, 99, 99, 99, 99, 99, 99, 00, 33, 00, 00, 00, 00, 00, 00, 00, 00, 00, 01, 00, 00, // OP5
+      99, 99, 99, 99, 99, 99, 99, 00, 33, 00, 00, 00, 00, 00, 00, 00, 00, 00, 01, 00, 00, // OP4
+      99, 99, 99, 99, 99, 99, 99, 00, 33, 00, 00, 00, 00, 00, 00, 00, 00, 00, 01, 00, 00, // OP4
+      99, 99, 99, 99, 99, 99, 99, 00, 33, 00, 00, 00, 00, 00, 00, 00, 00, 00, 01, 00, 00, // OP4
+      99, 99, 99, 99, 99, 99, 99, 00, 33, 00, 00, 00, 00, 00, 00, 00, 99, 00, 01, 00, 00, // OP4
+      99, 99, 99, 99, 50, 50, 50, 50,                                                     // 4 * pitch EG rates, 4 * pitch EG level
+      01, 00, 01,                                                                         // algorithm, feedback, osc sync
+      35, 00, 00, 00, 01, 00,                                                             // lfo speed, lfo delay, lfo pitch_mod_depth, lfo_amp_mod_depth, lfo_sync, lfo_waveform
+      03, 48,                                                                             // pitch_mod_sensitivity, transpose
+      73, 78, 73, 84, 32, 86, 79, 73, 67, 69                                              // 10 * char for name ("INIT VOICE")
+    };
+    float samplerate;
+    uint8_t data[NUM_VOICE_PARAMETERS];
+    uint8_t max_notes;
+    uint8_t used_notes;
+    PluginFx fx;
+    Controllers controllers;
+    int32_t lastKeyDown;
+    uint32_t xrun;
+    uint16_t render_time_max;
+    int16_t currentNote;
+    bool sustain;
+    float vuSignal;
+    bool monoMode;
+    bool noteRefreshMode;
+    bool refreshVoice;
+    uint8_t engineType;
+    VoiceStatus voiceStatus;
+    Lfo lfo;
+    EngineMsfa* engineMsfa;
+    EngineMkI* engineMkI;
+    EngineOpl* engineOpl;
+    void getSamples(float* buffer, uint16_t n_samples);
+    void getSamples(int16_t* buffer, uint16_t n_samples);
+    void compress(float* wav_in, float* wav_out, uint16_t n, float threshold, float slope, uint16_t sr,  float tla, float twnd, float tatt, float trel);
+    bool use_compressor;
+    uint8_t velocity_offset;
+    uint8_t velocity_max;
+    float velocity_diff;
+#ifndef TEENSYDUINO
+    Compressor* compressor;
+#endif
+};
+
+#endif

third-party/Synth_Dexed/dx7note.cpp

@@ -0,0 +1,388 @@
+/*
+   Copyright 2016-2017 Pascal Gauthier.
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#include <math.h>
+#include <stdlib.h>
+#include "synth.h"
+#include "freqlut.h"
+#include "exp2.h"
+#include "controllers.h"
+#include "dx7note.h"
+#include "porta.h"
+//#include <cmath>
+
+const int FEEDBACK_BITDEPTH = 8;
+
+int32_t midinote_to_logfreq(int midinote) {
+  //const int32_t base = 50857777;  // (1 << 24) * (log(440) / log(2) - 69/12)
+  const int32_t base = 50857777;  // (1 << 24) * (LOG_FUNC(440) / LOG_FUNC(2) - 69/12)
+  const int32_t step = (1 << 24) / 12;
+  return base + step * midinote;
+}
+
+int32_t logfreq_round2semi(int freq) {
+  const int base = 50857777;  // (1 << 24) * (log(440) / log(2) - 69/12)
+  const int step = (1 << 24) / 12;
+  const int rem = (freq - base) % step;
+  return freq - rem;
+}
+
+const int32_t coarsemul[] = {
+  -16777216, 0, 16777216, 26591258, 33554432, 38955489, 43368474, 47099600,
+  50331648, 53182516, 55732705, 58039632, 60145690, 62083076, 63876816,
+  65546747, 67108864, 68576247, 69959732, 71268397, 72509921, 73690858,
+  74816848, 75892776, 76922906, 77910978, 78860292, 79773775, 80654032,
+  81503396, 82323963, 83117622
+};
+
+int32_t osc_freq(int midinote, int mode, int coarse, int fine, int detune) {
+  // TODO: pitch randomization
+  int32_t logfreq;
+  if (mode == 0) {
+    logfreq = midinote_to_logfreq(midinote);
+    // could use more precision, closer enough for now. those numbers comes from my DX7
+    //FRAC_NUM detuneRatio = 0.0209 * exp(-0.396 * (((float)logfreq) / (1 << 24))) / 7;
+    FRAC_NUM detuneRatio = 0.0209 * EXP_FUNC(-0.396 * (((float)logfreq) / (1 << 24))) / 7;
+    logfreq += detuneRatio * logfreq * (detune - 7);
+
+    logfreq += coarsemul[coarse & 31];
+    if (fine) {
+      // (1 << 24) / log(2)
+      //logfreq += (int32_t)floor(24204406.323123 * log(1 + 0.01 * fine) + 0.5);
+      logfreq += (int32_t)floor(24204406.323123 * LOG_FUNC(1 + 0.01 * fine) + 0.5);
+    }
+
+    // // This was measured at 7.213Hz per count at 9600Hz, but the exact
+    // // value is somewhat dependent on midinote. Close enough for now.
+    // //logfreq += 12606 * (detune -7);
+  } else {
+    // ((1 << 24) * log(10) / log(2) * .01) << 3
+    logfreq = (4458616 * ((coarse & 3) * 100 + fine)) >> 3;
+    logfreq += detune > 7 ? 13457 * (detune - 7) : 0;
+  }
+  return logfreq;
+}
+
+const uint8_t velocity_data[64] = {
+  0, 70, 86, 97, 106, 114, 121, 126, 132, 138, 142, 148, 152, 156, 160, 163,
+  166, 170, 173, 174, 178, 181, 184, 186, 189, 190, 194, 196, 198, 200, 202,
+  205, 206, 209, 211, 214, 216, 218, 220, 222, 224, 225, 227, 229, 230, 232,
+  233, 235, 237, 238, 240, 241, 242, 243, 244, 246, 246, 248, 249, 250, 251,
+  252, 253, 254
+};
+
+// See "velocity" section of notes. Returns velocity delta in microsteps.
+int ScaleVelocity(int velocity, int sensitivity) {
+  int clamped_vel = std::max(0, std::min(127, velocity));
+  int vel_value = velocity_data[clamped_vel >> 1] - 239;
+  int scaled_vel = ((sensitivity * vel_value + 7) >> 3) << 4;
+  return scaled_vel;
+}
+
+int ScaleRate(int midinote, int sensitivity) {
+  int x = std::min(31, std::max(0, midinote / 3 - 7));
+  int qratedelta = (sensitivity * x) >> 3;
+#ifdef SUPER_PRECISE
+  int rem = x & 7;
+  if (sensitivity == 3 && rem == 3) {
+    qratedelta -= 1;
+  } else if (sensitivity == 7 && rem > 0 && rem < 4) {
+    qratedelta += 1;
+  }
+#endif
+  return qratedelta;
+}
+
+const uint8_t exp_scale_data[] = {
+  0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 11, 14, 16, 19, 23, 27, 33, 39, 47, 56, 66,
+  80, 94, 110, 126, 142, 158, 174, 190, 206, 222, 238, 250
+};
+
+int ScaleCurve(int group, int depth, int curve) {
+  int scale;
+  if (curve == 0 || curve == 3) {
+    // linear
+    scale = (group * depth * 329) >> 12;
+  } else {
+    // exponential
+    int n_scale_data = sizeof(exp_scale_data);
+    int raw_exp = exp_scale_data[std::min(group, n_scale_data - 1)];
+    scale = (raw_exp * depth * 329) >> 15;
+  }
+  if (curve < 2) {
+    scale = -scale;
+  }
+  return scale;
+}
+
+int ScaleLevel(int midinote, int break_pt, int left_depth, int right_depth,
+               int left_curve, int right_curve) {
+  int offset = midinote - break_pt - 17;
+  if (offset >= 0) {
+    return ScaleCurve((offset + 1) / 3, right_depth, right_curve);
+  } else {
+    return ScaleCurve(-(offset - 1) / 3, left_depth, left_curve);
+  }
+}
+
+static const uint8_t pitchmodsenstab[] = {
+  0, 10, 20, 33, 55, 92, 153, 255
+};
+
+// 0, 66, 109, 255
+static const uint32_t ampmodsenstab[] = {
+  0, 4342338, 7171437, 16777216
+};
+
+Dx7Note::Dx7Note() {
+  for (int op = 0; op < 6; op++) {
+    params_[op].phase = 0;
+    params_[op].gain_out = 0;
+  }
+}
+
+//void Dx7Note::init(const uint8_t patch[156], int midinote, int velocity) {
+void Dx7Note::init(const uint8_t patch[156], int midinote, int velocity, int srcnote, int porta, const Controllers *ctrls) {
+  int rates[4];
+  int levels[4];
+  for (int op = 0; op < 6; op++) {
+    int off = op * 21;
+    for (int i = 0; i < 4; i++) {
+      rates[i] = patch[off + i];
+      levels[i] = patch[off + 4 + i];
+    }
+    int outlevel = patch[off + 16];
+    outlevel = Env::scaleoutlevel(outlevel);
+    int level_scaling = ScaleLevel(midinote, patch[off + 8], patch[off + 9],
+                                   patch[off + 10], patch[off + 11], patch[off + 12]);
+    outlevel += level_scaling;
+    outlevel = std::min(127, outlevel);
+    outlevel = outlevel << 5;
+    outlevel += ScaleVelocity(velocity, patch[off + 15]);
+    outlevel = std::max(0, outlevel);
+    int rate_scaling = ScaleRate(midinote, patch[off + 13]);
+    env_[op].init((const int32_t*)rates, (const int32_t*)levels, outlevel, rate_scaling);
+
+    int mode = patch[off + 17];
+    int coarse = patch[off + 18];
+    int fine = patch[off + 19];
+    int detune = patch[off + 20];
+    int32_t freq = osc_freq(midinote, mode, coarse, fine, detune);
+    opMode[op] = mode;
+    basepitch_[op] = freq;
+    porta_curpitch_[op] = freq;
+    ampmodsens_[op] = ampmodsenstab[patch[off + 14] & 3];
+
+    if (porta >= 0)
+      porta_curpitch_[op] = osc_freq(srcnote, mode, coarse, fine, detune);
+  }
+  for (int i = 0; i < 4; i++) {
+    rates[i] = patch[126 + i];
+    levels[i] = patch[130 + i];
+  }
+  pitchenv_.set((const int32_t *)rates, (const int32_t*)levels);
+  algorithm_ = patch[134];
+  int feedback = patch[135];
+  fb_shift_ = feedback != 0 ? FEEDBACK_BITDEPTH - feedback : 16;
+  pitchmoddepth_ = (patch[139] * 165) >> 6;
+  pitchmodsens_ = pitchmodsenstab[patch[143] & 7];
+  ampmoddepth_ = (patch[140] * 165) >> 6;
+  porta_rateindex_ = (porta < 128) ? porta : 127;
+  porta_gliss_ = ctrls->values_[kControllerPortamentoGlissando];
+}
+
+void Dx7Note::compute(int32_t *buf, int32_t lfo_val, int32_t lfo_delay, const Controllers *ctrls) {
+  // ==== PITCH ====
+  uint32_t pmd = pitchmoddepth_ * lfo_delay;  // Q32
+  int32_t senslfo = pitchmodsens_ * (lfo_val - (1 << 23));
+  int32_t pmod_1 = (((int64_t) pmd) * (int64_t) senslfo) >> 39;
+  pmod_1 = abs(pmod_1);
+  int32_t pmod_2 = (int32_t)(((int64_t)ctrls->pitch_mod * (int64_t)senslfo) >> 14);
+  pmod_2 = abs(pmod_2);
+  int32_t pitch_mod = std::max(pmod_1, pmod_2);
+  pitch_mod = pitchenv_.getsample() + (pitch_mod * (senslfo < 0 ? -1 : 1));
+
+  // ---- PITCH BEND ----
+  int pitchbend = ctrls->values_[kControllerPitch];
+  int32_t pb = (pitchbend - 0x2000);
+  if (pb != 0) {
+    if (ctrls->values_[kControllerPitchStep] == 0) {
+      pb = ((float) (pb << 11)) * ((float) ctrls->values_[kControllerPitchRange]) / 12.0;
+    } else {
+      int stp = 12 / ctrls->values_[kControllerPitchStep];
+      pb = pb * stp / 8191;
+      pb = (pb * (8191 / stp)) << 11;
+    }
+  }
+  int32_t pitch_base = pb + ctrls->masterTune;
+  pitch_mod += pitch_base;
+
+  // ==== AMP MOD ====
+  lfo_val = (1 << 24) - lfo_val;
+  uint32_t amod_1 = (uint32_t)(((int64_t) ampmoddepth_ * (int64_t) lfo_delay) >> 8); // Q24 :D
+  amod_1 = (uint32_t)(((int64_t) amod_1 * (int64_t) lfo_val) >> 24);
+  uint32_t amod_2 = (uint32_t)(((int64_t) ctrls->amp_mod * (int64_t) lfo_val) >> 7); // Q?? :|
+  uint32_t amd_mod = std::max(amod_1, amod_2);
+
+  // ==== EG AMP MOD ====
+  uint32_t amod_3 = (ctrls->eg_mod + 1) << 17;
+  amd_mod = std::max((1 << 24) - amod_3, amd_mod);
+
+  // ==== OP RENDER ====
+  for (int op = 0; op < 6; op++) {
+    // if ( ctrls->opSwitch[op] == '0' )  {
+    if (!(ctrls->opSwitch & (1 << op)))  {
+      env_[op].getsample(); // advance the envelop even if it is not playing
+      params_[op].level_in = 0;
+    } else {
+      //int32_t gain = pow(2, 10 + level * (1.0 / (1 << 24)));
+
+      int32_t basepitch = basepitch_[op];
+
+      if ( opMode[op] )
+        params_[op].freq = Freqlut::lookup(basepitch + pitch_base);
+      else {
+        if ( porta_rateindex_ >= 0 ) {
+          basepitch = porta_curpitch_[op];
+          if ( porta_gliss_ )
+            basepitch = logfreq_round2semi(basepitch);
+        }
+        params_[op].freq = Freqlut::lookup(basepitch + pitch_mod);
+      }
+
+      int32_t level = env_[op].getsample();
+      if (ampmodsens_[op] != 0) {
+        uint32_t sensamp = (uint32_t)(((uint64_t) amd_mod) * ((uint64_t) ampmodsens_[op]) >> 24);
+
+        // TODO: mehhh.. this needs some real tuning.
+        //uint32_t pt = exp(((float)sensamp) / 262144 * 0.07 + 12.2);
+        uint32_t pt = EXP_FUNC(((float)sensamp) / 262144 * 0.07 + 12.2);
+        uint32_t ldiff = (uint32_t)(((uint64_t)level) * (((uint64_t)pt << 4)) >> 28);
+        level -= ldiff;
+      }
+      params_[op].level_in = level;
+    }
+  }
+
+  // ==== PORTAMENTO ====
+  int porta = porta_rateindex_;
+  if ( porta >= 0 ) {
+    int32_t rate = Porta::rates[porta];
+    for (int op = 0; op < 6; op++) {
+      int32_t cur = porta_curpitch_[op];
+      int32_t dst = basepitch_[op];
+
+      bool going_up = cur < dst;
+      int32_t newpitch = cur + (going_up ? +rate : -rate);
+
+      if ( going_up ? (cur > dst) : (cur < dst) )
+        newpitch = dst;
+
+      porta_curpitch_[op] = newpitch;
+    }
+  }
+
+  ctrls->core->render(buf, params_, algorithm_, fb_buf_, fb_shift_);
+}
+
+void Dx7Note::keyup() {
+  for (int op = 0; op < 6; op++) {
+    env_[op].keydown(false);
+  }
+  pitchenv_.keydown(false);
+}
+
+void Dx7Note::update(const uint8_t patch[156], int midinote, int velocity, int porta, const Controllers *ctrls) {
+  int rates[4];
+  int levels[4];
+  for (int op = 0; op < 6; op++) {
+    int off = op * 21;
+    int mode = patch[off + 17];
+    int coarse = patch[off + 18];
+    int fine = patch[off + 19];
+    int detune = patch[off + 20];
+    int32_t freq = osc_freq(midinote, mode, coarse, fine, detune);
+    basepitch_[op] = freq;
+    porta_curpitch_[op] = freq;
+    ampmodsens_[op] = ampmodsenstab[patch[off + 14] & 3];
+    opMode[op] = mode;
+
+    for (int i = 0; i < 4; i++) {
+      rates[i] = patch[off + i];
+      levels[i] = patch[off + 4 + i];
+    }
+    int outlevel = patch[off + 16];
+    outlevel = Env::scaleoutlevel(outlevel);
+    int level_scaling = ScaleLevel(midinote, patch[off + 8], patch[off + 9],
+                                   patch[off + 10], patch[off + 11], patch[off + 12]);
+    outlevel += level_scaling;
+    outlevel = std::min(127, outlevel);
+    outlevel = outlevel << 5;
+    outlevel += ScaleVelocity(velocity, patch[off + 15]);
+    outlevel = std::max(0, outlevel);
+    int rate_scaling = ScaleRate(midinote, patch[off + 13]);
+    env_[op].update((const int32_t*)rates, (const int32_t*)levels, (int32_t)outlevel, rate_scaling);
+  }
+  algorithm_ = patch[134];
+  int feedback = patch[135];
+  fb_shift_ = feedback != 0 ? FEEDBACK_BITDEPTH - feedback : 16;
+  pitchmoddepth_ = (patch[139] * 165) >> 6;
+  pitchmodsens_ = pitchmodsenstab[patch[143] & 7];
+  ampmoddepth_ = (patch[140] * 165) >> 6;
+  porta_rateindex_ = (porta < 128) ? porta : 127;
+  porta_gliss_ = ctrls->values_[kControllerPortamentoGlissando];
+
+}
+
+void Dx7Note::peekVoiceStatus(VoiceStatus &status) {
+  for (int i = 0; i < 6; i++) {
+    status.amp[i] = Exp2::lookup(params_[i].level_in - (14 * (1 << 24)));
+    env_[i].getPosition(&status.ampStep[i]);
+  }
+  pitchenv_.getPosition(&status.pitchStep);
+}
+
+/**
+   Used in monophonic mode to transfer voice state from different notes
+*/
+void Dx7Note::transferState(Dx7Note &src) {
+  for (int i = 0; i < 6; i++) {
+    env_[i].transfer(src.env_[i]);
+    params_[i].gain_out = src.params_[i].gain_out;
+    params_[i].phase = src.params_[i].phase;
+  }
+}
+
+void Dx7Note::transferSignal(Dx7Note &src) {
+  for (int i = 0; i < 6; i++) {
+    params_[i].gain_out = src.params_[i].gain_out;
+    params_[i].phase = src.params_[i].phase;
+  }
+}
+
+void Dx7Note::transferPortamento(Dx7Note &src) {
+  for (int i = 0; i < 6; i++) {
+    porta_curpitch_[i] = src.porta_curpitch_[i];
+  }
+}
+
+void Dx7Note::oscSync() {
+  for (int i = 0; i < 6; i++) {
+    params_[i].gain_out = 0;
+    params_[i].phase = 0;
+  }
+}

third-party/Synth_Dexed/dx7note.h

@@ -0,0 +1,82 @@
+/*
+   Copyright 2016-2017 Pascal Gauthier.
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#ifndef SYNTH_DX7NOTE_H_
+#define SYNTH_DX7NOTE_H_
+
+// This is the logic to put together a note from the MIDI description
+// and run the low-level modules.
+
+// It will continue to evolve a bit, as note-stealing logic, scaling,
+// and real-time control of parameters live here.
+
+#include <stdint.h>
+#include "env.h"
+#include "pitchenv.h"
+#include "fm_core.h"
+
+struct VoiceStatus {
+  uint32_t amp[6];
+  char ampStep[6];
+  char pitchStep;
+};
+
+class Dx7Note {
+  public:
+    Dx7Note();
+    void init(const uint8_t patch[156], int midinote, int velocity, int srcnote, int porta, const Controllers *ctrls);
+
+    // Note: this _adds_ to the buffer. Interesting question whether it's
+    // worth it...
+    void compute(int32_t *buf, int32_t lfo_val, int32_t lfo_delay, const Controllers *ctrls);
+
+    void keyup();
+
+    // TODO: some way of indicating end-of-note. Maybe should be a return
+    // value from the compute method? (Having a count return from keyup
+    // is also tempting, but if there's a dynamic parameter change after
+    // keyup, that won't work.
+
+    // PG:add the update
+    void update(const uint8_t patch[156], int midinote, int velocity, int porta, const Controllers *ctrls);
+    void peekVoiceStatus(VoiceStatus &status);
+    void transferState(Dx7Note& src);
+    void transferSignal(Dx7Note &src);
+    void transferPortamento(Dx7Note &src);
+    void oscSync();
+
+  private:
+    Env env_[6];
+    FmOpParams params_[6];
+    PitchEnv pitchenv_;
+    int32_t basepitch_[6];
+    int32_t fb_buf_[2]={0 ,0};
+    int32_t fb_shift_;
+    int32_t ampmodsens_[6];
+    int32_t opMode[6];
+
+    int ampmoddepth_;
+    int algorithm_;
+    int pitchmoddepth_;
+    int pitchmodsens_;
+
+    int porta_rateindex_;
+    int porta_gliss_;
+    int32_t porta_curpitch_[6];
+};
+
+#endif

third-party/Synth_Dexed/env.cpp

@@ -0,0 +1,190 @@
+/*
+   Copyright 2017 Pascal Gauthier.
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+//using namespace std;
+
+#include <math.h>
+
+#include "synth.h"
+#include "env.h"
+
+uint32_t Env::sr_multiplier = (1 << 24);
+
+const int32_t levellut[] = {
+  0, 5, 9, 13, 17, 20, 23, 25, 27, 29, 31, 33, 35, 37, 39, 41, 42, 43, 45, 46
+};
+
+#ifdef ACCURATE_ENVELOPE
+const int32_t statics[] = {
+  1764000, 1764000, 1411200, 1411200, 1190700, 1014300, 992250,
+  882000, 705600, 705600, 584325, 507150, 502740, 441000, 418950,
+  352800, 308700, 286650, 253575, 220500, 220500, 176400, 145530,
+  145530, 125685, 110250, 110250, 88200, 88200, 74970, 61740,
+  61740, 55125, 48510, 44100, 37485, 31311, 30870, 27562, 27562,
+  22050, 18522, 17640, 15435, 14112, 13230, 11025, 9261, 9261, 7717,
+  6615, 6615, 5512, 5512, 4410, 3969, 3969, 3439, 2866, 2690, 2249,
+  1984, 1896, 1808, 1411, 1367, 1234, 1146, 926, 837, 837, 705,
+  573, 573, 529, 441, 441
+  // and so on, I stopped measuring after R=76 (needs to be double-checked anyway)
+};
+#endif
+
+void Env::init_sr(double sampleRate) {
+  sr_multiplier = (44100.0 / sampleRate) * (1 << 24);
+}
+
+void Env::init(const int32_t r[4], const int32_t l[4], int32_t ol, int32_t rate_scaling) {
+  for (int i = 0; i < 4; i++) {
+    rates_[i] = r[i];
+    levels_[i] = l[i];
+  }
+  outlevel_ = ol;
+  rate_scaling_ = rate_scaling;
+  level_ = 0;
+  down_ = true;
+  advance(0);
+}
+
+int32_t Env::getsample() {
+#ifdef ACCURATE_ENVELOPE
+  if (staticcount_) {
+    staticcount_ -= _N_;
+    if (staticcount_ <= 0) {
+      staticcount_ = 0;
+      advance(ix_ + 1);
+    }
+  }
+#endif
+
+  if (ix_ < 3 || ((ix_ < 4) && !down_)) {
+    if (staticcount_) {
+      ;
+    }
+    else if (rising_) {
+      const int32_t jumptarget = 1716;
+      if (level_ < (jumptarget << 16)) {
+        level_ = jumptarget << 16;
+      }
+      level_ += (((17 << 24) - level_) >> 24) * inc_;
+      // TODO: should probably be more accurate when inc is large
+      if (level_ >= targetlevel_) {
+        level_ = targetlevel_;
+        advance(ix_ + 1);
+      }
+    }
+    else {  // !rising
+      level_ -= inc_;
+      if (level_ <= targetlevel_) {
+        level_ = targetlevel_;
+        advance(ix_ + 1);
+      }
+    }
+  }
+  // TODO: this would be a good place to set level to 0 when under threshold
+  return level_;
+}
+
+void Env::keydown(bool d) {
+  if (down_ != d) {
+    down_ = d;
+    advance(d ? 0 : 3);
+  }
+}
+
+int32_t Env::scaleoutlevel(int32_t outlevel) {
+  return outlevel >= 20 ? 28 + outlevel : levellut[outlevel];
+}
+
+void Env::advance(int newix) {
+  ix_ = newix;
+  if (ix_ < 4) {
+    int32_t newlevel = levels_[ix_];
+    int32_t actuallevel = scaleoutlevel(newlevel) >> 1;
+    actuallevel = (actuallevel << 6) + outlevel_ - 4256;
+    actuallevel = actuallevel < 16 ? 16 : actuallevel;
+    // level here is same as Java impl
+    targetlevel_ = actuallevel << 16;
+    rising_ = (targetlevel_ > level_);
+
+    // rate
+    int32_t qrate = (rates_[ix_] * 41) >> 6;
+    qrate += rate_scaling_;
+    qrate = std::min(int(qrate), 63);
+
+#ifdef ACCURATE_ENVELOPE
+    if (targetlevel_ == level_ || (ix_ == 0 && newlevel == 0)) {
+      // approximate number of samples at 44.100 kHz to achieve the time
+      // empirically gathered using 2 TF1s, could probably use some double-checking
+      // and cleanup, but it's pretty close for now.
+      int32_t staticrate = rates_[ix_];
+      staticrate += rate_scaling_; // needs to be checked, as well, but seems correct
+      staticrate = std::min(int(staticrate), 99);
+      staticcount_ = staticrate < 77 ? statics[staticrate] : 20 * (99 - staticrate);
+      if (staticrate < 77 && (ix_ == 0 && newlevel == 0)) {
+        staticcount_ /= 20; // attack is scaled faster
+      }
+      staticcount_ = (int)(((int64_t)staticcount_ * (int64_t)sr_multiplier) >> 24);
+    }
+    else {
+      staticcount_ = 0;
+    }
+#endif
+    inc_ = (4 + (qrate & 3)) << (2 + LG_N + (qrate >> 2));
+    // meh, this should be fixed elsewhere
+    inc_ = (int)(((int64_t)inc_ * (int64_t)sr_multiplier) >> 24);
+  }
+}
+
+void Env::update(const int32_t r[4], const int32_t l[4], int32_t ol, int32_t rate_scaling) {
+  for (int i = 0; i < 4; i++) {
+    rates_[i] = r[i];
+    levels_[i] = l[i];
+  }
+  outlevel_ = ol;
+  rate_scaling_ = rate_scaling;
+  if ( down_ ) {
+    // for now we simply reset ourselves at level 3
+    int32_t newlevel = levels_[2];
+    int32_t actuallevel = scaleoutlevel(newlevel) >> 1;
+    actuallevel = (actuallevel << 6) - 4256;
+    actuallevel = actuallevel < 16 ? 16 : actuallevel;
+    targetlevel_ = actuallevel << 16;
+    advance(2);
+  }
+}
+
+void Env::getPosition(char *step) {
+  *step = ix_;
+}
+
+void Env::transfer(Env &src) {
+  for (int i = 0; i < 4; i++) {
+    rates_[i] = src.rates_[i];
+    levels_[i] = src.levels_[i];
+  }
+  outlevel_ = src.outlevel_;
+  rate_scaling_ = src.rate_scaling_;
+  level_ = src.level_;
+  targetlevel_ = src.targetlevel_;
+  rising_ = src.rising_;
+  ix_ = src.ix_;
+  down_ = src.down_;
+#ifdef ACCURATE_ENVELOPE
+  staticcount_ = src.staticcount_;
+#endif
+  inc_ = src.inc_;
+}

third-party/Synth_Dexed/env.h

@@ -0,0 +1,81 @@
+/*
+   Copyright 2017 Pascal Gauthier.
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#ifndef __ENV_H
+#define __ENV_H
+
+#include "synth.h"
+
+// DX7 envelope generation
+
+#define ACCURATE_ENVELOPE
+
+class Env {
+  public:
+
+    // The rates and levels arrays are calibrated to match the Dx7 parameters
+    // (ie, value 0..99). The outlevel parameter is calibrated in microsteps
+    // (ie units of approx .023 dB), with 99 * 32 = nominal full scale. The
+    // rate_scaling parameter is in qRate units (ie 0..63).
+    void init(const int32_t rates[4], const int32_t levels[4], int32_t outlevel,
+              int32_t rate_scaling);
+
+    void update(const int32_t rates[4], const int32_t levels[4], int32_t outlevel,
+                int32_t rate_scaling);
+    // Result is in Q24/doubling log format. Also, result is subsampled
+    // for every N samples.
+    // A couple more things need to happen for this to be used as a gain
+    // value. First, the # of outputs scaling needs to be applied. Also,
+    // modulation.
+    // Then, of course, log to linear.
+    int32_t getsample();
+
+    void keydown(bool down);
+    static int32_t scaleoutlevel(int32_t outlevel);
+    void getPosition(char *step);
+
+    static void init_sr(double sample_rate);
+    void transfer(Env &src);
+
+  private:
+
+    // PG: This code is normalized to 44100, need to put a multiplier
+    // if we are not using 44100.
+    static uint32_t sr_multiplier;
+
+    int32_t rates_[4];
+    int32_t levels_[4];
+    int32_t outlevel_;
+    int32_t rate_scaling_;
+    // Level is stored so that 2^24 is one doubling, ie 16 more bits than
+    // the DX7 itself (fraction is stored in level rather than separate
+    // counter)
+    int32_t level_;
+    int32_t targetlevel_;
+    bool rising_;
+    int32_t ix_;
+    int32_t inc_;
+#ifdef ACCURATE_ENVELOPE
+    int32_t staticcount_;
+#endif
+
+    bool down_;
+
+    void advance(int newix);
+};
+
+#endif

third-party/Synth_Dexed/exp2.cpp

@@ -0,0 +1,70 @@
+/*
+ * Copyright 2013 Google Inc.
+ * 
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ * 
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ * 
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define _USE_MATH_DEFINES
+#include <math.h>
+
+#include "synth.h"
+#include "exp2.h"
+
+//#include <stdio.h>
+
+#ifdef _MSC_VER
+#define exp2(arg) pow(2.0, arg)
+#endif
+
+int32_t exp2tab[EXP2_N_SAMPLES << 1];
+
+void Exp2::init() {
+  FRAC_NUM inc = exp2(1.0 / EXP2_N_SAMPLES);
+  FRAC_NUM y = 1 << 30;
+  for (int i = 0; i < EXP2_N_SAMPLES; i++) {
+    exp2tab[(i << 1) + 1] = (int32_t)floor(y + 0.5);
+    y *= inc;
+  }
+  for (int i = 0; i < EXP2_N_SAMPLES - 1; i++) {
+    exp2tab[i << 1] = exp2tab[(i << 1) + 3] - exp2tab[(i << 1) + 1];
+  }
+  exp2tab[(EXP2_N_SAMPLES << 1) - 2] = (1U << 31) - exp2tab[(EXP2_N_SAMPLES << 1) - 1];
+}
+
+int32_t tanhtab[TANH_N_SAMPLES << 1];
+
+static FRAC_NUM dtanh(FRAC_NUM y) {
+  return 1 - y * y;
+}
+
+void Tanh::init() {
+  FRAC_NUM step = 4.0 / TANH_N_SAMPLES;
+  FRAC_NUM y = 0;
+  for (int i = 0; i < TANH_N_SAMPLES; i++) {
+    tanhtab[(i << 1) + 1] = (1 << 24) * y + 0.5;
+    //printf("%d\n", tanhtab[(i << 1) + 1]);
+    // Use a basic 4th order Runge-Kutte to compute tanh from its
+    // differential equation.
+    FRAC_NUM k1 = dtanh(y);
+    FRAC_NUM k2 = dtanh(y + 0.5 * step * k1);
+    FRAC_NUM k3 = dtanh(y + 0.5 * step * k2);
+    FRAC_NUM k4 = dtanh(y + step * k3);
+    FRAC_NUM dy = (step / 6) * (k1 + k4 + 2 * (k2 + k3));
+    y += dy;
+  }
+  for (int i = 0; i < TANH_N_SAMPLES - 1; i++) {
+    tanhtab[i << 1] = tanhtab[(i << 1) + 3] - tanhtab[(i << 1) + 1];
+  }
+  int32_t lasty = (1 << 24) * y + 0.5;
+  tanhtab[(TANH_N_SAMPLES << 1) - 2] = lasty - tanhtab[(TANH_N_SAMPLES << 1) - 1];
+}

third-party/Synth_Dexed/exp2.h

@@ -0,0 +1,80 @@
+/*
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+class Exp2 {
+  public:
+    Exp2();
+
+    static void init();
+
+    // Q24 in, Q24 out
+    static int32_t lookup(int32_t x);
+};
+
+#define EXP2_LG_N_SAMPLES 10
+#define EXP2_N_SAMPLES (1 << EXP2_LG_N_SAMPLES)
+
+#define EXP2_INLINE
+
+extern int32_t exp2tab[EXP2_N_SAMPLES << 1];
+
+#ifdef EXP2_INLINE
+inline
+int32_t Exp2::lookup(int32_t x) {
+  const int32_t SHIFT = 24 - EXP2_LG_N_SAMPLES;
+  int32_t lowbits = x & ((1 << SHIFT) - 1);
+  int32_t x_int = (x >> (SHIFT - 1)) & ((EXP2_N_SAMPLES - 1) << 1);
+  int32_t dy = exp2tab[x_int];
+  int32_t y0 = exp2tab[x_int + 1];
+
+  int32_t y = y0 + (((int64_t)dy * (int64_t)lowbits) >> SHIFT);
+  return y >> (6 - (x >> 24));
+}
+#endif
+
+class Tanh {
+  public:
+    static void init();
+
+    // Q24 in, Q24 out
+    static int32_t lookup(int32_t x);
+};
+
+#define TANH_LG_N_SAMPLES 10
+#define TANH_N_SAMPLES (1 << TANH_LG_N_SAMPLES)
+
+extern int32_t tanhtab[TANH_N_SAMPLES << 1];
+
+inline
+int32_t Tanh::lookup(int32_t x) {
+  int32_t signum = x >> 31;
+  x ^= signum;
+  if (x >= (4 << 24)) {
+    if (x >= (17 << 23)) {
+      return signum ^ (1 << 24);
+    }
+    int32_t sx = ((int64_t) - 48408812 * (int64_t)x) >> 24;
+    return signum ^ ((1 << 24) - 2 * Exp2::lookup(sx));
+  } else {
+    const int32_t SHIFT = 26 - TANH_LG_N_SAMPLES;
+    int32_t lowbits = x & ((1 << SHIFT) - 1);
+    int32_t x_int = (x >> (SHIFT - 1)) & ((TANH_N_SAMPLES - 1) << 1);
+    int32_t dy = tanhtab[x_int];
+    int32_t y0 = tanhtab[x_int + 1];
+    int32_t y = y0 + (((int64_t)dy * (int64_t)lowbits) >> SHIFT);
+    return y ^ signum;
+  }
+}

third-party/Synth_Dexed/fm_core.cpp

@@ -0,0 +1,100 @@
+/*
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+//using namespace std;
+
+#include "synth.h"
+#include "exp2.h"
+#include "fm_op_kernel.h"
+#include "fm_core.h"
+
+const FmAlgorithm FmCore::algorithms[32] = {
+  { { 0xc1, 0x11, 0x11, 0x14, 0x01, 0x14 } }, // 1
+  { { 0x01, 0x11, 0x11, 0x14, 0xc1, 0x14 } }, // 2
+  { { 0xc1, 0x11, 0x14, 0x01, 0x11, 0x14 } }, // 3
+  { { 0xc1, 0x11, 0x94, 0x01, 0x11, 0x14 } }, // 4
+  { { 0xc1, 0x14, 0x01, 0x14, 0x01, 0x14 } }, // 5
+  { { 0xc1, 0x94, 0x01, 0x14, 0x01, 0x14 } }, // 6
+  { { 0xc1, 0x11, 0x05, 0x14, 0x01, 0x14 } }, // 7
+  { { 0x01, 0x11, 0xc5, 0x14, 0x01, 0x14 } }, // 8
+  { { 0x01, 0x11, 0x05, 0x14, 0xc1, 0x14 } }, // 9
+  { { 0x01, 0x05, 0x14, 0xc1, 0x11, 0x14 } }, // 10
+  { { 0xc1, 0x05, 0x14, 0x01, 0x11, 0x14 } }, // 11
+  { { 0x01, 0x05, 0x05, 0x14, 0xc1, 0x14 } }, // 12
+  { { 0xc1, 0x05, 0x05, 0x14, 0x01, 0x14 } }, // 13
+  { { 0xc1, 0x05, 0x11, 0x14, 0x01, 0x14 } }, // 14
+  { { 0x01, 0x05, 0x11, 0x14, 0xc1, 0x14 } }, // 15
+  { { 0xc1, 0x11, 0x02, 0x25, 0x05, 0x14 } }, // 16
+  { { 0x01, 0x11, 0x02, 0x25, 0xc5, 0x14 } }, // 17
+  { { 0x01, 0x11, 0x11, 0xc5, 0x05, 0x14 } }, // 18
+  { { 0xc1, 0x14, 0x14, 0x01, 0x11, 0x14 } }, // 19
+  { { 0x01, 0x05, 0x14, 0xc1, 0x14, 0x14 } }, // 20
+  { { 0x01, 0x14, 0x14, 0xc1, 0x14, 0x14 } }, // 21
+  { { 0xc1, 0x14, 0x14, 0x14, 0x01, 0x14 } }, // 22
+  { { 0xc1, 0x14, 0x14, 0x01, 0x14, 0x04 } }, // 23
+  { { 0xc1, 0x14, 0x14, 0x14, 0x04, 0x04 } }, // 24
+  { { 0xc1, 0x14, 0x14, 0x04, 0x04, 0x04 } }, // 25
+  { { 0xc1, 0x05, 0x14, 0x01, 0x14, 0x04 } }, // 26
+  { { 0x01, 0x05, 0x14, 0xc1, 0x14, 0x04 } }, // 27
+  { { 0x04, 0xc1, 0x11, 0x14, 0x01, 0x14 } }, // 28
+  { { 0xc1, 0x14, 0x01, 0x14, 0x04, 0x04 } }, // 29
+  { { 0x04, 0xc1, 0x11, 0x14, 0x04, 0x04 } }, // 30
+  { { 0xc1, 0x14, 0x04, 0x04, 0x04, 0x04 } }, // 31
+  { { 0xc4, 0x04, 0x04, 0x04, 0x04, 0x04 } }, // 32
+};
+
+int n_out(const FmAlgorithm &alg) {
+  int32_t count = 0;
+  for (int i = 0; i < 6; i++) {
+    if ((alg.ops[i] & 7) == OUT_BUS_ADD) count++;
+  }
+  return count;
+}
+
+uint8_t FmCore::get_carrier_operators(uint8_t algorithm)
+{
+  uint8_t op_out = 0;
+  FmAlgorithm alg = algorithms[algorithm];
+
+  for (uint8_t i = 0; i < 6; i++)
+  {
+    if ((alg.ops[i]&OUT_BUS_ADD) == OUT_BUS_ADD)
+      op_out |= 1 << i;
+  }
+
+  return op_out;
+}
+
+void FmCore::dump() {
+#ifdef VERBOSE
+  for (int i = 0; i < 32; i++) {
+    cout << (i + 1) << ":";
+    const FmAlgorithm &alg = algorithms[i];
+    for (int j = 0; j < 6; j++) {
+      int32_t flags = alg.ops[j];
+      cout << " ";
+      if (flags & FB_IN) cout << "[";
+      cout << (flags & IN_BUS_ONE ? "1" : flags & IN_BUS_TWO ? "2" : "0") << "->";
+      cout << (flags & OUT_BUS_ONE ? "1" : flags & OUT_BUS_TWO ? "2" : "0");
+      if (flags & OUT_BUS_ADD) cout << "+";
+      //cout << alg.ops[j].in << "->" << alg.ops[j].out;
+      if (flags & FB_OUT) cout << "]";
+    }
+    cout << " " << n_out(alg);
+    cout << endl;
+  }
+#endif
+}

third-party/Synth_Dexed/fm_core.h

@@ -0,0 +1,59 @@
+/*
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#ifndef __FM_CORE_H
+#define __FM_CORE_H
+
+#include <stdint.h>
+#include "aligned_buf.h"
+#include "fm_op_kernel.h"
+#include "synth.h"
+#include "controllers.h"
+
+class FmOperatorInfo {
+  public:
+    int32_t in;
+    int32_t out;
+};
+
+enum FmOperatorFlags {
+  OUT_BUS_ONE = 1 << 0,
+  OUT_BUS_TWO = 1 << 1,
+  OUT_BUS_ADD = 1 << 2,
+  IN_BUS_ONE = 1 << 4,
+  IN_BUS_TWO = 1 << 5,
+  FB_IN = 1 << 6,
+  FB_OUT = 1 << 7
+};
+
+class FmAlgorithm {
+  public:
+    int32_t ops[6];
+};
+
+class FmCore {
+  public:
+    FmCore() {};
+    virtual ~FmCore() {};
+    static void dump();
+    static uint8_t get_carrier_operators(uint8_t algorithm);
+    virtual void render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_gain) = 0;
+  protected:
+    AlignedBuf<int32_t, _N_>buf_[2];
+    const static FmAlgorithm algorithms[32];
+};
+
+#endif

third-party/Synth_Dexed/fm_op_kernel.cpp

@@ -0,0 +1,384 @@
+/*
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#include <math.h>
+#include <cstdlib>
+#include <stdint.h>
+#include "synth.h"
+#include "sin.h"
+#include "fm_op_kernel.h"
+
+#ifdef __ARM_NEON
+#include <arm_neon.h>
+extern "C"
+void neon_fm_kernel(const int32_t *in, const int32_t *busin, int32_t *out, int32_t count,
+  int32_t phase0, int32_t freq, int32_t gain1, int32_t dgain);
+
+const int32_t __attribute__ ((aligned(16))) const_0_1_2_3_arg[4] = {0, 1, 2, 3};
+const int32_t __attribute__ ((aligned(16))) mask23_arg = 0x7fffff;
+const float __attribute__ ((aligned(16))) coeffs_arg[4] = {
+        -0.01880853017455781, 0.25215252666796095, -1.2333439964934032, 1.0
+};
+const int32_t __attribute__ ((aligned(16))) zeros[_N_] = {0};
+
+
+void neon_fm_kernel(const int32_t *in, const int32_t *busin, int32_t *out, int32_t count,
+    int32_t phase0, int32_t freq_arg, int32_t gain1_arg, int32_t dgain_arg) {
+  int32x4_t phase = vld1q_dup_s32(&phase0);
+  int32x4_t freq = vld1q_dup_s32(&freq_arg);
+  int32x4_t const_0_1_2_3 = vld1q_s32(const_0_1_2_3_arg);
+  phase = vmlaq_s32(phase, freq, const_0_1_2_3);
+  int32x4_t gain1 = vld1q_dup_s32(&gain1_arg);
+  int32x4_t dgain = vld1q_dup_s32(&dgain_arg);
+  gain1 = vmlaq_s32(gain1, dgain, const_0_1_2_3);
+  int32x4_t mask23 = vld1q_dup_s32(&mask23_arg);
+  float32x4_t coeffs = vld1q_f32(coeffs_arg);
+  float32x4_t gainf = vcvtq_n_f32_s32(gain1, 24);
+  int32x4_t freq4 = vshlq_n_s32(freq, 2);
+  float32x4_t dgainf = vcvtq_n_f32_s32(dgain, 22);
+
+  count -= 4;
+  int32x4_t q15 = vmovq_n_s32(0x800000);
+  int32x4_t q7 = vmovq_n_s32(0x400000);
+  while (true) {
+    int32x4_t phase4 = vaddq_s32(phase, freq4);
+    int32x4_t phase8 = vaddq_s32(phase4, freq4);
+    int32x4_t data1a = vld1q_s32(in);
+    data1a = vaddq_s32(data1a, phase);
+    int32x4_t data1b = vld1q_s32(in + 4);
+    data1b = vaddq_s32(data1b, phase4);
+    int32x4_t data1c = vld1q_s32(in + 8);
+    data1c = vaddq_s32(data1c, phase8);
+    phase = vaddq_s32(phase8, freq4);
+    in += 12;
+    int32x4_t data4a = (int32x4_t)vtstq_s32(data1a, q15);
+    int32x4_t data4b = (int32x4_t)vtstq_s32(data1b, q15);
+    int32x4_t data4c = (int32x4_t)vtstq_s32(data1c, q15);
+    data1a = vandq_s32(data1a, mask23);
+    data1b = vandq_s32(data1b, mask23);
+    data1c = vandq_s32(data1c, mask23);
+    data1a = vsubq_s32(data1a, q7);
+    data1b = vsubq_s32(data1b, q7);
+    data1c = vsubq_s32(data1c, q7);
+    float32x4_t fdata1a = vcvtq_n_f32_s32(data1a, 22);
+    float32x4_t fdata1b = vcvtq_n_f32_s32(data1b, 22);
+    float32x4_t fdata1c = vcvtq_n_f32_s32(data1c, 22);
+    fdata1a = vmulq_f32(fdata1a, fdata1a);
+    fdata1b = vmulq_f32(fdata1b, fdata1b);
+    fdata1c = vmulq_f32(fdata1c, fdata1c);
+    float32x4_t fdata2a = vdupq_lane_f32(vget_low_f32(coeffs), 1);
+    float32x4_t fdata2b = vdupq_lane_f32(vget_low_f32(coeffs), 1);
+    float32x4_t fdata2c = vdupq_lane_f32(vget_low_f32(coeffs), 1);
+    fdata2a = vmlaq_lane_f32(fdata2a, fdata1a, vget_low_f32(coeffs), 0);
+    fdata2b = vmlaq_lane_f32(fdata2b, fdata1b, vget_low_f32(coeffs), 0);
+    fdata2c = vmlaq_lane_f32(fdata2c, fdata1c, vget_low_f32(coeffs), 0);
+    float32x4_t fdata3a = vdupq_lane_f32(vget_high_f32(coeffs), 0);
+    float32x4_t fdata3b = vdupq_lane_f32(vget_high_f32(coeffs), 0);
+    float32x4_t fdata3c = vdupq_lane_f32(vget_high_f32(coeffs), 0);
+    fdata3a = vmlaq_f32(fdata3a, fdata1a, fdata2a);
+    fdata3b = vmlaq_f32(fdata3b, fdata1b, fdata2b);
+    fdata3c = vmlaq_f32(fdata3c, fdata1c, fdata2c);
+    fdata2a = vdupq_lane_f32(vget_high_f32(coeffs), 1);
+    fdata2b = vdupq_lane_f32(vget_high_f32(coeffs), 1);
+    fdata2c = vdupq_lane_f32(vget_high_f32(coeffs), 1);
+    fdata2a = vmlaq_f32(fdata2a, fdata1a, fdata3a);
+    fdata2b = vmlaq_f32(fdata2b, fdata1b, fdata3b);
+    fdata2c = vmlaq_f32(fdata2c, fdata1c, fdata3c);
+    fdata3a = vaddq_f32(gainf, dgainf);
+    fdata3b = vaddq_f32(fdata3a, dgainf);
+    fdata2a = vmulq_f32(fdata2a, gainf);
+    fdata2b = vmulq_f32(fdata2b, fdata3a);
+    fdata2c = vmulq_f32(fdata2c, fdata3b);
+    gainf = vaddq_f32(fdata3b, dgainf);
+    int32x4_t data3a = vcvtq_n_s32_f32(fdata2a, 24);
+    int32x4_t data3b = vcvtq_n_s32_f32(fdata2b, 24);
+    int32x4_t data3c = vcvtq_n_s32_f32(fdata2c, 24);
+    data1a = vld1q_s32(busin);
+    data1b = vld1q_s32(busin + 4);
+    data1c = vld1q_s32(busin + 8);
+    busin += 12;
+    data3a = veorq_s32(data3a, data4a);
+    data3b = veorq_s32(data3b, data4b);
+    data3c = veorq_s32(data3c, data4c);
+    data3a = vaddq_s32(data3a, data1a);
+    data3b = vaddq_s32(data3b, data1b);
+    data3c = vaddq_s32(data3c, data1c);
+    vst1q_s32(out, data3a);
+    vst1q_s32(out + 4, data3b);
+    vst1q_s32(out + 8, data3c);
+    out += 12;
+    count -= 12;
+    if (count <= 0) {
+      if (count == 0) {
+        // finish last chunk of 4
+        data1a = vld1q_s32(in);
+        data1a = vaddq_s32(data1a, phase);
+        data4a = (int32x4_t)vtstq_s32(data1a, q15);
+        data1a = vandq_s32(data1a, mask23);
+        data1a = vsubq_s32(data1a, q7);
+        fdata1a = vcvtq_n_f32_s32(data1a, 22);
+        fdata1a = vmulq_f32(fdata1a, fdata1a);
+        fdata2a = vdupq_lane_f32(vget_low_f32(coeffs), 1);
+        fdata2a = vmlaq_lane_f32(fdata2a, fdata1a, vget_low_f32(coeffs), 0);
+        fdata3a = vdupq_lane_f32(vget_high_f32(coeffs), 0);
+        fdata3a = vmlaq_f32(fdata3a, fdata1a, fdata2a);
+        fdata2a = vdupq_lane_f32(vget_high_f32(coeffs), 1);
+        fdata2a = vmlaq_f32(fdata2a, fdata1a, fdata3a);
+        fdata2a = vmulq_f32(fdata2a, gainf);
+        data3a = vcvtq_n_s32_f32(fdata2a, 24);
+        data1a = vld1q_s32(busin);
+        data3a = veorq_s32(data3a, data4a);
+        data3a = vaddq_s32(data3a, data1a);
+        vst1q_s32(out, data3a);
+      }
+      break;
+    }
+  }
+}
+#endif
+
+void FmOpKernel::compute(int32_t *output, const int32_t *input,
+                         int32_t phase0, int32_t freq,
+                         int32_t gain1, int32_t gain2, bool add) {
+  int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
+  int32_t gain = gain1;
+#ifdef __ARM_NEON
+    neon_fm_kernel(input, add ? output : zeros, output, _N_,
+                   phase0, freq, gain, dgain);
+#else
+  int32_t phase = phase0;
+    if (add) {
+      for (int i = 0; i < _N_; i++) {
+        gain += dgain;
+        int32_t y = Sin::lookup(phase + input[i]);
+        int32_t y1 = ((int64_t)y * (int64_t)gain) >> 24;
+        output[i] += y1;
+        phase += freq;
+      }
+    } else {
+      for (int i = 0; i < _N_; i++) {
+        gain += dgain;
+        int32_t y = Sin::lookup(phase + input[i]);
+        int32_t y1 = ((int64_t)y * (int64_t)gain) >> 24;
+        output[i] = y1;
+        phase += freq;
+      }
+    }
+#endif
+  }
+
+void FmOpKernel::compute_pure(int32_t *output, int32_t phase0, int32_t freq,
+                              int32_t gain1, int32_t gain2, bool add) {
+  int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
+  int32_t gain = gain1;
+#ifdef __ARM_NEON
+    neon_fm_kernel(zeros, add ? output : zeros, output, _N_,
+                   phase0, freq, gain, dgain);
+#else
+  int32_t phase = phase0;
+    if (add) {
+      for (int i = 0; i < _N_; i++) {
+        gain += dgain;
+        int32_t y = Sin::lookup(phase);
+        int32_t y1 = ((int64_t)y * (int64_t)gain) >> 24;
+        output[i] += y1;
+        phase += freq;
+      }
+    } else {
+      for (int i = 0; i < _N_; i++) {
+        gain += dgain;
+        int32_t y = Sin::lookup(phase);
+        int32_t y1 = ((int64_t)y * (int64_t)gain) >> 24;
+        output[i] = y1;
+        phase += freq;
+      }
+    }
+#endif
+}
+
+#define noDOUBLE_ACCURACY
+#define HIGH_ACCURACY
+
+void FmOpKernel::compute_fb(int32_t *output, int32_t phase0, int32_t freq,
+                            int32_t gain1, int32_t gain2,
+                            int32_t *fb_buf, int32_t fb_shift, bool add) {
+  int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
+  int32_t gain = gain1;
+  int32_t phase = phase0;
+  int32_t y0 = fb_buf[0];
+  int32_t y = fb_buf[1];
+  if (add) {
+    for (int i = 0; i < _N_; i++) {
+      gain += dgain;
+      int32_t scaled_fb = (y0 + y) >> (fb_shift + 1);
+      y0 = y;
+      y = Sin::lookup(phase + scaled_fb);
+      y = ((int64_t)y * (int64_t)gain) >> 24;
+      output[i] += y;
+      phase += freq;
+    }
+  } else {
+    for (int i = 0; i < _N_; i++) {
+      gain += dgain;
+      int32_t scaled_fb = (y0 + y) >> (fb_shift + 1);
+      y0 = y;
+      y = Sin::lookup(phase + scaled_fb);
+      y = ((int64_t)y * (int64_t)gain) >> 24;
+      output[i] = y;
+      phase += freq;
+    }
+  }
+  fb_buf[0] = y0;
+  fb_buf[1] = y;
+}
+
+////////////////////////////////////////////////////////////////////////////////////
+////////////////////////////////////////////////////////////////////////////////////
+////////////////////////////////////////////////////////////////////////////////////
+////////////////////////////////////////////////////////////////////////////////////
+
+// Experimental sine wave generators below
+#if 0
+// Results: accuracy 64.3 mean, 170 worst case
+// high accuracy: 5.0 mean, 49 worst case
+void FmOpKernel::compute_pure(int32_t *output, int32_t phase0, int32_t freq,
+                              int32_t gain1, int32_t gain2, bool add) {
+  int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
+  int32_t gain = gain1;
+  int32_t phase = phase0;
+#ifdef HIGH_ACCURACY
+  int32_t u = Sin::compute10(phase << 6);
+  u = ((int64_t)u * gain) >> 30;
+  int32_t v = Sin::compute10((phase << 6) + (1 << 28));  // quarter cycle
+  v = ((int64_t)v * gain) >> 30;
+  int32_t s = Sin::compute10(freq << 6);
+  int32_t c = Sin::compute10((freq << 6) + (1 << 28));
+#else
+  int32_t u = Sin::compute(phase);
+  u = ((int64_t)u * gain) >> 24;
+  int32_t v = Sin::compute(phase + (1 << 22));  // quarter cycle
+  v = ((int64_t)v * gain) >> 24;
+  int32_t s = Sin::compute(freq) << 6;
+  int32_t c = Sin::compute(freq + (1 << 22)) << 6;
+#endif
+  for (int i = 0; i < _N_; i++) {
+    output[i] = u;
+    int32_t t = ((int64_t)v * (int64_t)c - (int64_t)u * (int64_t)s) >> 30;
+    u = ((int64_t)u * (int64_t)c + (int64_t)v * (int64_t)s) >> 30;
+    v = t;
+  }
+}
+#endif
+
+#if 0
+// Results: accuracy 392.3 mean, 15190 worst case (near freq = 0.5)
+// for freq < 0.25, 275.2 mean, 716 worst
+// high accuracy: 57.4 mean, 7559 worst
+//  freq < 0.25: 17.9 mean, 78 worst
+void FmOpKernel::compute_pure(int32_t *output, int32_t phase0, int32_t freq,
+                              int32_t gain1, int32_t gain2, bool add) {
+  int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
+  int32_t gain = gain1;
+  int32_t phase = phase0;
+#ifdef HIGH_ACCURACY
+  int32_t u = floor(gain * sin(phase * (M_PI / (1 << 23))) + 0.5);
+  int32_t v = floor(gain * cos((phase - freq * 0.5) * (M_PI / (1 << 23))) + 0.5);
+  int32_t a = floor((1 << 25) * sin(freq * (M_PI / (1 << 24))) + 0.5);
+#else
+  int32_t u = Sin::compute(phase);
+  u = ((int64_t)u * gain) >> 24;
+  int32_t v = Sin::compute(phase + (1 << 22) - (freq >> 1));
+  v = ((int64_t)v * gain) >> 24;
+  int32_t a = Sin::compute(freq >> 1) << 1;
+#endif
+  for (int i = 0; i < _N_; i++) {
+    output[i] = u;
+    v -= ((int64_t)a * (int64_t)u) >> 24;
+    u += ((int64_t)a * (int64_t)v) >> 24;
+  }
+}
+#endif
+
+#if 0
+// Results: accuracy 370.0 mean, 15480 worst case (near freq = 0.5)
+// with FRAC_NUM accuracy initialization: mean 1.55, worst 58 (near freq = 0)
+// with high accuracy: mean 4.2, worst 292 (near freq = 0.5)
+void FmOpKernel::compute_pure(int32_t *output, int32_t phase0, int32_t freq,
+                              int32_t gain1, int32_t gain2, bool add) {
+  int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
+  int32_t gain = gain1;
+  int32_t phase = phase0;
+#ifdef DOUBLE_ACCURACY
+  int32_t u = floor((1 << 30) * sin(phase * (M_PI / (1 << 23))) + 0.5);
+  FRAC_NUM a_d = sin(freq * (M_PI / (1 << 24)));
+  int32_t v = floor((1LL << 31) * a_d * cos((phase - freq * 0.5) *
+                    (M_PI / (1 << 23))) + 0.5);
+  int32_t aa = floor((1LL << 31) * a_d * a_d + 0.5);
+#else
+#ifdef HIGH_ACCURACY
+  int32_t u = Sin::compute10(phase << 6);
+  int32_t v = Sin::compute10((phase << 6) + (1 << 28) - (freq << 5));
+  int32_t a = Sin::compute10(freq << 5);
+  v = ((int64_t)v * (int64_t)a) >> 29;
+  int32_t aa = ((int64_t)a * (int64_t)a) >> 29;
+#else
+  int32_t u = Sin::compute(phase) << 6;
+  int32_t v = Sin::compute(phase + (1 << 22) - (freq >> 1));
+  int32_t a = Sin::compute(freq >> 1);
+  v = ((int64_t)v * (int64_t)a) >> 17;
+  int32_t aa = ((int64_t)a * (int64_t)a) >> 17;
+#endif
+#endif
+
+  if (aa < 0) aa = (1 << 31) - 1;
+  for (int i = 0; i < _N_; i++) {
+    gain += dgain;
+    output[i] = ((int64_t)u * (int64_t)gain) >> 30;
+    v -= ((int64_t)aa * (int64_t)u) >> 29;
+    u += v;
+  }
+}
+#endif
+
+#if 0
+// Results:: accuracy 112.3 mean, 4262 worst (near freq = 0.5)
+// high accuracy 2.9 mean, 143 worst
+void FmOpKernel::compute_pure(int32_t *output, int32_t phase0, int32_t freq,
+                              int32_t gain1, int32_t gain2, bool add) {
+  int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
+  int32_t gain = gain1;
+  int32_t phase = phase0;
+#ifdef HIGH_ACCURACY
+  int32_t u = Sin::compute10(phase << 6);
+  int32_t lastu = Sin::compute10((phase - freq) << 6);
+  int32_t a = Sin::compute10((freq << 6) + (1 << 28)) << 1;
+#else
+  int32_t u = Sin::compute(phase) << 6;
+  int32_t lastu = Sin::compute(phase - freq) << 6;
+  int32_t a = Sin::compute(freq + (1 << 22)) << 7;
+#endif
+  if (a < 0 && freq < 256) a = (1 << 31) - 1;
+  if (a > 0 && freq > 0x7fff00) a = -(1 << 31);
+  for (int i = 0; i < _N_; i++) {
+    gain += dgain;
+    output[i] = ((int64_t)u * (int64_t)gain) >> 30;
+    //output[i] = u;
+    int32_t newu = (((int64_t)u * (int64_t)a) >> 30) - lastu;
+    lastu = u;
+    u = newu;
+  }
+}
+#endif

third-party/Synth_Dexed/fm_op_kernel.h

@@ -0,0 +1,47 @@
+/*
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#ifndef __FM_OP_KERNEL_H
+#define __FM_OP_KERNEL_H
+
+struct FmOpParams {
+  int32_t level_in;      // value to be computed (from level to gain[0])
+  int32_t gain_out;      // computed value (gain[1] to gain[0])
+  int32_t freq;
+  int32_t phase;
+};
+
+class FmOpKernel {
+  public:
+    // gain1 and gain2 represent linear step: gain for sample i is
+    // gain1 + (1 + i) / 64 * (gain2 - gain1)
+
+    // This is the basic FM operator. No feedback.
+    static void compute(int32_t *output, const int32_t *input,
+                        int32_t phase0, int32_t freq,
+                        int32_t gain1, int32_t gain2, bool add);
+
+    // This is a sine generator, no feedback.
+    static void compute_pure(int32_t *output, int32_t phase0, int32_t freq,
+                             int32_t gain1, int32_t gain2, bool add);
+
+    // One op with feedback, no add.
+    static void compute_fb(int32_t *output, int32_t phase0, int32_t freq,
+                           int32_t gain1, int32_t gain2,
+                           int32_t *fb_buf, int32_t fb_gain, bool add);
+};
+
+#endif

third-party/Synth_Dexed/freqlut.cpp

@@ -0,0 +1,56 @@
+/*
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+// Resolve frequency signal (1.0 in Q24 format = 1 octave) to phase delta.
+
+// The LUT is just a global, and we'll need the init function to be called before
+// use.
+
+#include <stdint.h>
+#include <math.h>
+
+#include "freqlut.h"
+#include "synth.h"
+
+#define LG_N_SAMPLES 10
+#define N_SAMPLES (1 << LG_N_SAMPLES)
+#define SAMPLE_SHIFT (24 - LG_N_SAMPLES)
+
+#define MAX_LOGFREQ_INT 20
+
+int32_t lut[N_SAMPLES + 1];
+
+void Freqlut::init(FRAC_NUM sample_rate) {
+  FRAC_NUM y = (1LL << (24 + MAX_LOGFREQ_INT)) / sample_rate;
+  FRAC_NUM inc = pow(2, 1.0 / N_SAMPLES);
+  for (int i = 0; i < N_SAMPLES + 1; i++) {
+    lut[i] = (int32_t)floor(y + 0.5);
+    y *= inc;
+  }
+}
+
+// Note: if logfreq is more than 20.0, the results will be inaccurate. However,
+// that will be many times the Nyquist rate.
+int32_t Freqlut::lookup(int32_t logfreq) {
+  int32_t ix = (logfreq & 0xffffff) >> SAMPLE_SHIFT;
+
+  int32_t y0 = lut[ix];
+  int32_t y1 = lut[ix + 1];
+  int32_t lowbits = logfreq & ((1 << SAMPLE_SHIFT) - 1);
+  int32_t y = y0 + ((((int64_t)(y1 - y0) * (int64_t)lowbits)) >> SAMPLE_SHIFT);
+  int32_t hibits = logfreq >> 24;
+  return y >> (MAX_LOGFREQ_INT - hibits);
+}

third-party/Synth_Dexed/freqlut.h

@@ -0,0 +1,23 @@
+/*
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#include "synth.h"
+
+class Freqlut {
+  public:
+    static void init(FRAC_NUM sample_rate);
+    static int32_t lookup(int32_t logfreq);
+};

third-party/Synth_Dexed/lfo.cpp

@@ -0,0 +1,99 @@
+/*
+   Copyright 2013 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+// Low frequency oscillator, compatible with DX7
+
+#include <algorithm>
+
+#include "synth.h"
+
+#include "sin.h"
+#include "lfo.h"
+
+uint32_t Lfo::unit_;
+
+void Lfo::init(FRAC_NUM sample_rate) {
+  // constant is 1 << 32 / 15.5s / 11
+  Lfo::unit_ = (int32_t)(_N_ * 25190424 / sample_rate + 0.5);
+}
+
+void Lfo::reset(const uint8_t params[6]) {
+  int32_t rate = params[0];  // 0..99
+  int32_t sr = rate == 0 ? 1 : (165 * rate) >> 6;
+  sr *= sr < 160 ? 11 : (11 + ((sr - 160) >> 4));
+  delta_ = unit_ * sr;
+  int32_t a = 99 - params[1];  // LFO delay
+  if (a == 99) {
+    delayinc_ = ~0u;
+    delayinc2_ = ~0u;
+  } else {
+    a = (16 + (a & 15)) << (1 + (a >> 4));
+    delayinc_ = unit_ * a;
+    a &= 0xff80;
+    a = std::max(0x80, int(a));
+    delayinc2_ = unit_ * a;
+  }
+  waveform_ = params[5];
+  sync_ = params[4] != 0;
+}
+
+int32_t Lfo::getsample() {
+  phase_ += delta_;
+  int32_t x;
+  switch (waveform_) {
+    case 0:  // triangle
+      x = phase_ >> 7;
+      x ^= -(phase_ >> 31);
+      x &= (1 << 24) - 1;
+      return x;
+    case 1:  // sawtooth down
+      return (~phase_ ^ (1U << 31)) >> 8;
+    case 2:  // sawtooth up
+      return (phase_ ^ (1U << 31)) >> 8;
+    case 3:  // square
+      return ((~phase_) >> 7) & (1 << 24);
+    case 4:  // sine
+      return (1 << 23) + (Sin::lookup(phase_ >> 8) >> 1);
+    case 5:  // s&h
+      if (phase_ < delta_) {
+        randstate_ = (randstate_ * 179 + 17) & 0xff;
+      }
+      x = randstate_ ^ 0x80;
+      return (x + 1) << 16;
+  }
+  return 1 << 23;
+}
+
+int32_t Lfo::getdelay() {
+  uint32_t delta = delaystate_ < (1U << 31) ? delayinc_ : delayinc2_;
+  uint64_t d = ((uint64_t)delaystate_) + delta;
+  if (d > ~0u) {
+    return 1 << 24;
+  }
+  delaystate_ = d;
+  if (d < (1U << 31)) {
+    return 0;
+  } else {
+    return (d >> 7) & ((1 << 24) - 1);
+  }
+}
+
+void Lfo::keydown() {
+  if (sync_) {
+    phase_ = (1U << 31) - 1;
+  }
+  delaystate_ = 0;
+}

third-party/Synth_Dexed/lfo.h

@@ -0,0 +1,43 @@
+/*
+   Copyright 2013 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+// Low frequency oscillator, compatible with DX7
+
+class Lfo {
+  public:
+    static void init(FRAC_NUM sample_rate);
+    void reset(const uint8_t params[6]);
+
+    // result is 0..1 in Q24
+    int32_t getsample();
+
+    // result is 0..1 in Q24
+    int32_t getdelay();
+
+    void keydown();
+  private:
+    static uint32_t unit_;
+
+    uint32_t phase_;  // Q32
+    uint32_t delta_;
+    uint8_t waveform_;
+    uint8_t randstate_;
+    bool sync_;
+
+    uint32_t delaystate_;
+    uint32_t delayinc_;
+    uint32_t delayinc2_;
+};

third-party/Synth_Dexed/pitchenv.cpp

@@ -0,0 +1,93 @@
+/*
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#include "synth.h"
+#include "pitchenv.h"
+
+int32_t PitchEnv::unit_;
+
+void PitchEnv::init(FRAC_NUM sample_rate) {
+  unit_ = _N_ * (1 << 24) / (21.3 * sample_rate) + 0.5;
+}
+
+const uint8_t pitchenv_rate[] = {
+  1, 2, 3, 3, 4, 4, 5, 5, 6, 6, 7, 7, 8, 8, 9, 9, 10, 10, 11, 11, 12,
+  12, 13, 13, 14, 14, 15, 16, 16, 17, 18, 18, 19, 20, 21, 22, 23, 24,
+  25, 26, 27, 28, 30, 31, 33, 34, 36, 37, 38, 39, 41, 42, 44, 46, 47,
+  49, 51, 53, 54, 56, 58, 60, 62, 64, 66, 68, 70, 72, 74, 76, 79, 82,
+  85, 88, 91, 94, 98, 102, 106, 110, 115, 120, 125, 130, 135, 141, 147,
+  153, 159, 165, 171, 178, 185, 193, 202, 211, 232, 243, 254, 255
+};
+
+const int8_t pitchenv_tab[] = {
+  -128, -116, -104, -95, -85, -76, -68, -61, -56, -52, -49, -46, -43,
+  -41, -39, -37, -35, -33, -32, -31, -30, -29, -28, -27, -26, -25, -24,
+  -23, -22, -21, -20, -19, -18, -17, -16, -15, -14, -13, -12, -11, -10,
+  -9, -8, -7, -6, -5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10,
+  11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27,
+  28, 29, 30, 31, 32, 33, 34, 35, 38, 40, 43, 46, 49, 53, 58, 65, 73,
+  82, 92, 103, 115, 127
+};
+
+void PitchEnv::set(const int32_t r[4], const int32_t l[4]) {
+  for (int i = 0; i < 4; i++) {
+    rates_[i] = r[i];
+    levels_[i] = l[i];
+  }
+  level_ = pitchenv_tab[l[3]] << 19;
+  down_ = true;
+  advance(0);
+}
+
+int32_t PitchEnv::getsample() {
+  if (ix_ < 3 || ((ix_ < 4) && !down_)) {
+    if (rising_) {
+      level_ += inc_;
+      if (level_ >= targetlevel_) {
+        level_ = targetlevel_;
+        advance(ix_ + 1);
+      }
+    } else {  // !rising
+      level_ -= inc_;
+      if (level_ <= targetlevel_) {
+        level_ = targetlevel_;
+        advance(ix_ + 1);
+      }
+    }
+  }
+  return level_;
+}
+
+void PitchEnv::keydown(bool d) {
+  if (down_ != d) {
+    down_ = d;
+    advance(d ? 0 : 3);
+  }
+}
+
+void PitchEnv::advance(int newix) {
+  ix_ = newix;
+  if (ix_ < 4) {
+    int32_t newlevel = levels_[ix_];
+    targetlevel_ = pitchenv_tab[newlevel] << 19;
+    rising_ = (targetlevel_ > level_);
+    inc_ = pitchenv_rate[rates_[ix_]] * unit_;
+  }
+}
+
+void PitchEnv::getPosition(char *step) {
+  *step = ix_;
+}

third-party/Synth_Dexed/pitchenv.h

@@ -0,0 +1,52 @@
+/*
+   Copyright 2013 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#ifndef __PITCHENV_H
+#define __PITCHENV_H
+
+// Computation of the DX7 pitch envelope
+
+class PitchEnv {
+  public:
+    static void init(FRAC_NUM sample_rate);
+
+    // The rates and levels arrays are calibrated to match the Dx7 parameters
+    // (ie, value 0..99).
+    void set(const int32_t rates[4], const int32_t levels[4]);
+
+    // Result is in Q24/octave
+    int32_t getsample();
+    void keydown(bool down);
+    void getPosition(char *step);
+  private:
+    static int32_t unit_;
+    int32_t rates_[4];
+    int32_t levels_[4];
+    int32_t level_;
+    int32_t targetlevel_;
+    bool rising_;
+    int32_t ix_;
+    int32_t inc_;
+
+    bool down_;
+
+    void advance(int newix);
+};
+
+extern const uint8_t pitchenv_rate[];
+extern const int8_t pitchenv_tab[];
+
+#endif

third-party/Synth_Dexed/porta.cpp

@@ -0,0 +1,35 @@
+/*
+   Copyright 2019 Jean Pierre Cimalando.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#include <math.h>
+#include "porta.h"
+#include "synth.h"
+
+void Porta::init_sr(double sampleRate)
+{
+  // compute portamento for CC 7-bit range
+
+  for (uint8_t i = 0; i < 128; ++i) {
+    // number of semitones travelled
+    double sps = 350.0 * pow(2.0, -0.062 * i);  // per second
+    double spf = sps / sampleRate;              // per frame
+    double spp = spf * _N_;                       // per period
+    const int32_t step = (1 << 24) / 12;
+    rates[i] = (int32_t)(0.5f + step * spp);    // to pitch units
+  }
+}
+
+int32_t Porta::rates[128];

third-party/Synth_Dexed/porta.h

@@ -0,0 +1,28 @@
+/*
+   Copyright 2019 Jean Pierre Cimalando.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#ifndef SYNTH_PORTA_H_
+#define SYNTH_PORTA_H_
+
+#include <stdint.h>
+
+struct Porta {
+  public:
+    static void init_sr(double sampleRate);
+    static int32_t rates[128];
+};
+
+#endif

third-party/Synth_Dexed/sin.cpp

@@ -0,0 +1,144 @@
+/*
+ * Copyright 2012 Google Inc.
+ * 
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ * 
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ * 
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define _USE_MATH_DEFINES
+#include <math.h>
+
+#include "synth.h"
+#include "sin.h"
+
+#define R (1 << 29)
+
+#ifdef SIN_DELTA
+int32_t sintab[SIN_N_SAMPLES << 1];
+#else
+int32_t sintab[SIN_N_SAMPLES + 1];
+#endif
+
+void Sin::init() {
+  FRAC_NUM dphase = 2 * M_PI / SIN_N_SAMPLES;
+  //int32_t c = (int32_t)floor(cos(dphase) * (1 << 30) + 0.5);
+  int32_t c = (int32_t)floor(COS_FUNC(dphase) * (1 << 30) + 0.5);
+  //int32_t s = (int32_t)floor(sin(dphase) * (1 << 30) + 0.5);
+  int32_t s = (int32_t)floor(SIN_FUNC(dphase) * (1 << 30) + 0.5);
+  int32_t u = 1 << 30;
+  int32_t v = 0;
+  for (int i = 0; i < SIN_N_SAMPLES / 2; i++) {
+#ifdef SIN_DELTA
+    sintab[(i << 1) + 1] = (v + 32) >> 6;
+    sintab[((i + SIN_N_SAMPLES / 2) << 1) + 1] = -((v + 32) >> 6);
+#else
+    sintab[i] = (v + 32) >> 6;
+    sintab[i + SIN_N_SAMPLES / 2] = -((v + 32) >> 6);
+#endif
+    int32_t t = ((int64_t)u * (int64_t)s + (int64_t)v * (int64_t)c + R) >> 30;
+    u = ((int64_t)u * (int64_t)c - (int64_t)v * (int64_t)s + R) >> 30;
+    v = t;
+  }
+#ifdef SIN_DELTA
+  for (int i = 0; i < SIN_N_SAMPLES - 1; i++) {
+    sintab[i << 1] = sintab[(i << 1) + 3] - sintab[(i << 1) + 1];
+  }
+  sintab[(SIN_N_SAMPLES << 1) - 2] = -sintab[(SIN_N_SAMPLES << 1) - 1];
+#else
+  sintab[SIN_N_SAMPLES] = 0;
+#endif
+}
+
+#ifndef SIN_INLINE
+int32_t Sin::lookup(int32_t phase) {
+  const int32_t SHIFT = 24 - SIN_LG_N_SAMPLES;
+  int32_t lowbits = phase & ((1 << SHIFT) - 1);
+#ifdef SIN_DELTA
+  int32_t phase_int = (phase >> (SHIFT - 1)) & ((SIN_N_SAMPLES - 1) << 1);
+  int32_t dy = sintab[phase_int];
+  int32_t y0 = sintab[phase_int + 1];
+
+  return y0 + (((int64_t)dy * (int64_t)lowbits) >> SHIFT);
+#else 
+  int32_t phase_int = (phase >> SHIFT) & (SIN_N_SAMPLES - 1);
+  int32_t y0 = sintab[phase_int];
+  int32_t y1 = sintab[phase_int + 1];
+
+  return y0 + (((int64_t)(y1 - y0) * (int64_t)lowbits) >> SHIFT);
+#endif
+}
+#endif
+
+
+#if 0
+// The following is an implementation designed not to use any lookup tables,
+// based on the following implementation by Basile Graf:
+// http://www.rossbencina.com/static/code/sinusoids/even_polynomial_sin_approximation.txt
+
+#define C0 (1 << 24)
+#define C1 (331121857 >> 2)
+#define C2 (1084885537 >> 4)
+#define C3 (1310449902 >> 6)
+
+int32_t Sin::compute(int32_t phase) {
+  int32_t x = (phase & ((1 << 23) - 1)) - (1 << 22);
+  int32_t x2 = ((int64_t)x * (int64_t)x) >> 22;
+  int32_t x4 = ((int64_t)x2 * (int64_t)x2) >> 24;
+  int32_t x6 = ((int64_t)x2 * (int64_t)x4) >> 24;
+  int32_t y = C0 -
+    (((int64_t)C1 * (int64_t)x2) >> 24) +
+    (((int64_t)C2 * (int64_t)x4) >> 24) -
+    (((int64_t)C3 * (int64_t)x6) >> 24);
+  y ^= -((phase >> 23) & 1);
+  return y;
+}
+#endif
+
+#if 1
+// coefficients are Chebyshev polynomial, computed by compute_cos_poly.py
+#define C8_0 16777216
+#define C8_2 -331168742
+#define C8_4 1089453524
+#define C8_6 -1430910663
+#define C8_8 950108533
+
+int32_t Sin::compute(int32_t phase) {
+  int32_t x = (phase & ((1 << 23) - 1)) - (1 << 22);
+  int32_t x2 = ((int64_t)x * (int64_t)x) >> 16;
+  int32_t y = (((((((((((((int64_t)C8_8
+    * (int64_t)x2) >> 32) + C8_6)
+    * (int64_t)x2) >> 32) + C8_4)
+    * (int64_t)x2) >> 32) + C8_2)
+    * (int64_t)x2) >> 32) + C8_0);
+  y ^= -((phase >> 23) & 1);
+  return y;
+}
+#endif
+
+#define C10_0 (1 << 30)
+#define C10_2 -1324675874  // scaled * 4
+#define C10_4 1089501821
+#define C10_6 -1433689867
+#define C10_8 1009356886
+#define C10_10 -421101352
+int32_t Sin::compute10(int32_t phase) {
+  int32_t x = (phase & ((1 << 29) - 1)) - (1 << 28);
+  int32_t x2 = ((int64_t)x * (int64_t)x) >> 26;
+  int32_t y = ((((((((((((((((int64_t)C10_10
+    * (int64_t)x2) >> 34) + C10_8)
+    * (int64_t)x2) >> 34) + C10_6)
+    * (int64_t)x2) >> 34) + C10_4)
+    * (int64_t)x2) >> 32) + C10_2)
+    * (int64_t)x2) >> 30) + C10_0);
+  y ^= -((phase >> 29) & 1);
+  return y;
+}

third-party/Synth_Dexed/sin.h

@@ -0,0 +1,62 @@
+/*
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+class Sin {
+  public:
+    Sin();
+
+    static void init();
+    static int32_t lookup(int32_t phase);
+    static int32_t compute(int32_t phase);
+
+    // A more accurate sine, both input and output Q30
+    static int32_t compute10(int32_t phase);
+};
+
+#define SIN_LG_N_SAMPLES 10
+#define SIN_N_SAMPLES (1 << SIN_LG_N_SAMPLES)
+
+#define SIN_INLINE
+
+// Use twice as much RAM for the LUT but avoid a little computation
+#define SIN_DELTA
+
+#ifdef SIN_DELTA
+extern int32_t sintab[SIN_N_SAMPLES << 1];
+#else
+extern int32_t sintab[SIN_N_SAMPLES + 1];
+#endif
+
+#ifdef SIN_INLINE
+inline
+int32_t Sin::lookup(int32_t phase) {
+  const int32_t SHIFT = 24 - SIN_LG_N_SAMPLES;
+  int32_t lowbits = phase & ((1 << SHIFT) - 1);
+#ifdef SIN_DELTA
+  int32_t phase_int = (phase >> (SHIFT - 1)) & ((SIN_N_SAMPLES - 1) << 1);
+  int32_t dy = sintab[phase_int];
+  int32_t y0 = sintab[phase_int + 1];
+
+  return y0 + (((int64_t)dy * (int64_t)lowbits) >> SHIFT);
+#else
+  int32_t phase_int = (phase >> SHIFT) & (SIN_N_SAMPLES - 1);
+  int32_t y0 = sintab[phase_int];
+  int32_t y1 = sintab[phase_int + 1];
+
+  return y0 + (((int64_t)(y1 - y0) * (int64_t)lowbits) >> SHIFT);
+#endif
+}
+#endif

third-party/Synth_Dexed/src/EngineMkI.cpp

@@ -69,23 +69,23 @@ static inline uint16_t sinLog(uint16_t phi) {
 EngineMkI::EngineMkI() {
     float bitReso = SINLOG_TABLESIZE;
     
-    for(int i=0;i<SINLOG_TABLESIZE;i++) {
+    for(int32_t i=0;i<SINLOG_TABLESIZE;i++) {
         float x1 = sin(((0.5+i)/bitReso) * M_PI/2.0);
         sinLogTable[i] = round(-1024 * log2(x1));
     }
     
     bitReso = SINEXP_TABLESIZE;
-    for(int i=0;i<SINEXP_TABLESIZE;i++) {
+    for(int32_t i=0;i<SINEXP_TABLESIZE;i++) {
         float x1 = (pow(2, float(i)/bitReso)-1) * 4096;
         sinExpTable[i] = round(x1);
     }
     
 #ifdef MKIDEBUG
-    char buffer[4096];
-    int pos = 0;
+    uint8_t buffer[4096];
+    int32_t pos = 0;
     
     TRACE("****************************************");
-    for(int i=0;i<SINLOG_TABLESIZE;i++) {
+    for(int32_t i=0;i<SINLOG_TABLESIZE;i++) {
         pos += sprintf(buffer+pos, "%d ", sinLogTable[i]);
         if ( pos > 90 ) {
             TRACE("SINLOGTABLE: %s" ,buffer);
@@ -97,7 +97,7 @@ EngineMkI::EngineMkI() {
     buffer[0] = 0;
     pos = 0;
     TRACE("----------------------------------------");    
-    for(int i=0;i<SINEXP_TABLESIZE;i++) {
+    for(int32_t i=0;i<SINEXP_TABLESIZE;i++) {
         pos += sprintf(buffer+pos, "%d ", sinExpTable[i]);
         if ( pos > 90 ) {
             TRACE("SINEXTTABLE: %s" ,buffer);
@@ -146,7 +146,7 @@ void EngineMkI::compute(int32_t *output, const int32_t *input,
     int32_t phase = phase0;
     const int32_t *adder = add ? output : zeros;
 
-    for (int i = 0; i < _N_; i++) {
+    for (uint8_t i = 0; i < _N_; i++) {
         gain += dgain;
         int32_t y = mkiSin((phase+input[i]), gain);
         output[i] = y + adder[i];
@@ -154,14 +154,13 @@ void EngineMkI::compute(int32_t *output, const int32_t *input,
     }
 }
 
-void EngineMkI::compute_pure(int32_t *output, int32_t phase0, int32_t freq,
-                             int32_t gain1, int32_t gain2, bool add) {
+void EngineMkI::compute_pure(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, bool add) {
     int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
     int32_t gain = gain1;
     int32_t phase = phase0;
     const int32_t *adder = add ? output : zeros;
     
-    for (int i = 0; i < _N_; i++) {
+    for (uint8_t i = 0; i < _N_; i++) {
         gain += dgain;
         int32_t y = mkiSin(phase , gain);
         output[i] = y + adder[i];
@@ -169,9 +168,7 @@ void EngineMkI::compute_pure(int32_t *output, int32_t phase0, int32_t freq,
     }
 }
 
-void EngineMkI::compute_fb(int32_t *output, int32_t phase0, int32_t freq,
-                           int32_t gain1, int32_t gain2,
-                           int32_t *fb_buf, int fb_shift, bool add) {
+void EngineMkI::compute_fb(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, int32_t *fb_buf, int32_t fb_shift, bool add) {
     int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
     int32_t gain = gain1;
     int32_t phase = phase0;
@@ -179,7 +176,7 @@ void EngineMkI::compute_fb(int32_t *output, int32_t phase0, int32_t freq,
     int32_t y0 = fb_buf[0];
     int32_t y = fb_buf[1];
     
-    for (int i = 0; i < _N_; i++) {
+    for (uint8_t i = 0; i < _N_; i++) {
         gain += dgain;
         int32_t scaled_fb = (y0 + y) >> (fb_shift + 1);
         y0 = y;
@@ -193,7 +190,7 @@ void EngineMkI::compute_fb(int32_t *output, int32_t phase0, int32_t freq,
 }
 
 // exclusively used for ALGO 6 with feedback
-void EngineMkI::compute_fb2(int32_t *output, FmOpParams *parms, int32_t gain01, int32_t gain02, int32_t *fb_buf, int fb_shift) {
+void EngineMkI::compute_fb2(int32_t *output, FmOpParams *parms, int32_t gain01, int32_t gain02, int32_t *fb_buf, int32_t fb_shift) {
     int32_t dgain[2];
     int32_t gain[2];
     int32_t phase[2];
@@ -211,7 +208,7 @@ void EngineMkI::compute_fb2(int32_t *output, FmOpParams *parms, int32_t gain01,
     dgain[0] = (gain02 - gain01 + (_N_ >> 1)) >> LG_N;
     dgain[1] = (parms[1].gain_out - (parms[1].gain_out == 0 ? (ENV_MAX-1) : parms[1].gain_out));
     
-    for (int i = 0; i < _N_; i++) {
+    for (uint8_t i = 0; i < _N_; i++) {
         int32_t scaled_fb = (y0 + y) >> (fb_shift + 1);
         
         // op 0
@@ -232,7 +229,7 @@ void EngineMkI::compute_fb2(int32_t *output, FmOpParams *parms, int32_t gain01,
 }
 
 // exclusively used for ALGO 4 with feedback
-void EngineMkI::compute_fb3(int32_t *output, FmOpParams *parms, int32_t gain01, int32_t gain02, int32_t *fb_buf, int fb_shift) {
+void EngineMkI::compute_fb3(int32_t *output, FmOpParams *parms, int32_t gain01, int32_t gain02, int32_t *fb_buf, int32_t fb_shift) {
     int32_t dgain[3];
     int32_t gain[3];
     int32_t phase[3];
@@ -255,7 +252,7 @@ void EngineMkI::compute_fb3(int32_t *output, FmOpParams *parms, int32_t gain01,
     dgain[2] = (parms[2].gain_out - (parms[2].gain_out == 0 ? (ENV_MAX-1) : parms[2].gain_out));
     
     
-    for (int i = 0; i < _N_; i++) {
+    for (uint8_t i = 0; i < _N_; i++) {
         int32_t scaled_fb = (y0 + y) >> (fb_shift + 1);
         
         // op 0
@@ -281,7 +278,7 @@ void EngineMkI::compute_fb3(int32_t *output, FmOpParams *parms, int32_t gain01,
 }
 
 void EngineMkI::render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_shift) {
-    const int kLevelThresh = ENV_MAX-100;
+    const int32_t kLevelThresh = ENV_MAX-100;
     FmAlgorithm alg = algorithms[algorithm];
     bool has_contents[3] = { true, false, false };
     bool fb_on = feedback_shift < 16;
@@ -292,12 +289,12 @@ void EngineMkI::render(int32_t *output, FmOpParams *params, int32_t algorithm, i
                 alg.ops[0] = 0xc4;
     }
     
-    for (int op = 0; op < 6; op++) {
-        int flags = alg.ops[op];
+    for (int32_t op = 0; op < 6; op++) {
+        int32_t flags = alg.ops[op];
         bool add = (flags & OUT_BUS_ADD) != 0;
         FmOpParams &param = params[op];
-        int inbus = (flags >> 4) & 3;
-        int outbus = flags & 3;
+        int32_t inbus = (flags >> 4) & 3;
+        int32_t outbus = flags & 3;
         int32_t *outptr = (outbus == 0) ? output : buf_[outbus - 1].get();
         int32_t gain1 = param.gain_out == 0 ? (ENV_MAX-1) : param.gain_out;
         int32_t gain2 = ENV_MAX-(param.level_in >> (28-ENV_BITDEPTH));
@@ -315,20 +312,20 @@ void EngineMkI::render(int32_t *output, FmOpParams *params, int32_t algorithm, i
                     switch ( algorithm ) {
                         // three operator feedback, process exception for ALGO 4
                         case 3 :
-                            compute_fb3(outptr, params, gain1, gain2, fb_buf, min((feedback_shift+2), (int32_t)16));
+                            compute_fb3(outptr, params, gain1, gain2, fb_buf, std::min((feedback_shift+2), (int32_t)16));
                             params[1].phase += params[1].freq << LG_N; // hack, we already processed op-5 - op-4
                             params[2].phase += params[2].freq << LG_N; // yuk yuk
                             op += 2; // ignore the 2 other operators
                             break;
                         // two operator feedback, process exception for ALGO 6
                         case 5 :
-                            compute_fb2(outptr, params, gain1, gain2, fb_buf, min((feedback_shift+2), (int32_t)16));
+                            compute_fb2(outptr, params, gain1, gain2, fb_buf, std::min((feedback_shift+2), (int32_t)16));
                             params[1].phase += params[1].freq << LG_N;  // yuk, hack, we already processed op-5
                             op++; // ignore next operator;
                             break;
                         case 31 :
                             // one operator feedback, process exception for ALGO 32
-                            compute_fb(outptr, param.phase, param.freq, gain1, gain2, fb_buf, min((feedback_shift+2), (int32_t)16), add);
+                            compute_fb(outptr, param.phase, param.freq, gain1, gain2, fb_buf, std::min((feedback_shift+2), (int32_t)16), add);
                             break;
                         default:
                             // one operator feedback, normal process

third-party/Synth_Dexed/src/EngineMkI.h

@@ -27,17 +27,17 @@ class EngineMkI : public FmCore {
 public:
     EngineMkI();
     ~EngineMkI() {};
-    void render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_shift);
+    void render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_shift) override;
     
     void compute(int32_t *output, const int32_t *input, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, bool add);
     
     void compute_pure(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, bool add);
     
-    void compute_fb(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, int32_t *fb_buf, int fb_gain, bool add);
+    void compute_fb(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, int32_t *fb_buf, int32_t fb_gain, bool add);
     
-    void compute_fb2(int32_t *output, FmOpParams *params, int32_t gain01, int32_t gain02, int32_t *fb_buf, int fb_shift);
+    void compute_fb2(int32_t *output, FmOpParams *params, int32_t gain01, int32_t gain02, int32_t *fb_buf, int32_t fb_shift);
     
-    void compute_fb3(int32_t *output, FmOpParams *params, int32_t gain01, int32_t gain02, int32_t *fb_buf, int fb_shift);
+    void compute_fb3(int32_t *output, FmOpParams *params, int32_t gain01, int32_t gain02, int32_t *fb_buf, int32_t fb_shift);
 };
 
 

third-party/Synth_Dexed/src/EngineMsfa.cpp

@@ -22,15 +22,15 @@
 #include "EngineMsfa.h"
 
 void EngineMsfa::render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_shift) {
-  const int kLevelThresh = 1120;
+  const int32_t kLevelThresh = 1120;
   const FmAlgorithm alg = algorithms[algorithm];
   bool has_contents[3] = { true, false, false };
-  for (int op = 0; op < 6; op++) {
-    int flags = alg.ops[op];
+  for (uint8_t op = 0; op < 6; op++) {
+    int32_t flags = alg.ops[op];
     bool add = (flags & OUT_BUS_ADD) != 0;
     FmOpParams &param = params[op];
-    int inbus = (flags >> 4) & 3;
-    int outbus = flags & 3;
+    int32_t inbus = (flags >> 4) & 3;
+    int32_t outbus = flags & 3;
     int32_t *outptr = (outbus == 0) ? output : buf_[outbus - 1].get();
     int32_t gain1 = param.gain_out;
     int32_t gain2 = Exp2::lookup(param.level_in - (14 * (1 << 24)));

third-party/Synth_Dexed/src/EngineMsfa.h

@@ -25,5 +25,5 @@ class EngineMsfa : public FmCore {
   public:
     EngineMsfa() {};
     ~EngineMsfa() {};
-    void render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_gain);
+    void render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_gain) override;
 };

third-party/Synth_Dexed/src/EngineOpl.cpp

@@ -118,7 +118,7 @@ void EngineOpl::compute(int32_t *output, const int32_t *input, int32_t phase0, i
     int32_t phase = phase0;
     const int32_t *adder = add ? output : zeros;
 
-    for (int i = 0; i < _N_; i++) {
+    for (uint8_t i = 0; i < _N_; i++) {
         gain += dgain;
         int32_t y = oplSin((phase+input[i]) >> 14, gain);
         output[i] = (y << 14) + adder[i];
@@ -132,7 +132,7 @@ void EngineOpl::compute_pure(int32_t *output, int32_t phase0, int32_t freq, int3
     int32_t phase = phase0;
     const int32_t *adder = add ? output : zeros;
 
-    for (int i = 0; i < _N_; i++) {
+    for (uint8_t i = 0; i < _N_; i++) {
         gain += dgain;
         int32_t y = oplSin(phase >> 14, gain);
         output[i] = (y << 14) + adder[i];
@@ -142,7 +142,7 @@ void EngineOpl::compute_pure(int32_t *output, int32_t phase0, int32_t freq, int3
 
 void EngineOpl::compute_fb(int32_t *output, int32_t phase0, int32_t freq,
                                     int32_t gain1, int32_t gain2,
-                              int32_t *fb_buf, int fb_shift, bool add) {
+                              int32_t *fb_buf, int32_t fb_shift, bool add) {
     int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
     int32_t gain = gain1;
     int32_t phase = phase0;
@@ -150,7 +150,7 @@ void EngineOpl::compute_fb(int32_t *output, int32_t phase0, int32_t freq,
     int32_t y0 = fb_buf[0];
     int32_t y = fb_buf[1];
     
-    for (int i = 0; i < _N_; i++) {
+    for (uint8_t i = 0; i < _N_; i++) {
         gain += dgain;
         int32_t scaled_fb = (y0 + y) >> (fb_shift + 1);
         y0 = y;
@@ -165,15 +165,15 @@ void EngineOpl::compute_fb(int32_t *output, int32_t phase0, int32_t freq,
 
 
 void EngineOpl::render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_shift) {
-    const int kLevelThresh = 507;  // really ????
+    const int32_t kLevelThresh = 507;  // really ????
     const FmAlgorithm alg = algorithms[algorithm];
     bool has_contents[3] = { true, false, false };
-    for (int op = 0; op < 6; op++) {
-        int flags = alg.ops[op];
+    for (uint8_t op = 0; op < 6; op++) {
+        int32_t flags = alg.ops[op];
         bool add = (flags & OUT_BUS_ADD) != 0;
         FmOpParams &param = params[op];
-        int inbus = (flags >> 4) & 3;
-        int outbus = flags & 3;
+        int32_t inbus = (flags >> 4) & 3;
+        int32_t outbus = flags & 3;
         int32_t *outptr = (outbus == 0) ? output : buf_[outbus - 1].get();
         int32_t gain1 = param.gain_out == 0 ? 511 : param.gain_out;
         int32_t gain2 = 512-(param.level_in >> 19);
@@ -207,15 +207,3 @@ void EngineOpl::render(int32_t *output, FmOpParams *params, int32_t algorithm, i
         param.phase += param.freq << LG_N;
     }
 }
-
-
-
-
-
-
-
-
-
-
-
-

third-party/Synth_Dexed/src/EngineOpl.h

@@ -30,10 +30,10 @@ class EngineOpl : public FmCore {
 public:
     EngineOpl() {};
     ~EngineOpl() {};
-    void render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_shift);
+    void render(int32_t *output, FmOpParams *params, int32_t algorithm, int32_t *fb_buf, int32_t feedback_shift) override;
     void compute(int32_t *output, const int32_t *input, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, bool add);
     void compute_pure(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, bool add);
-    void compute_fb(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, int32_t *fb_buf, int fb_gain, bool add);
+    void compute_fb(int32_t *output, int32_t phase0, int32_t freq, int32_t gain1, int32_t gain2, int32_t *fb_buf, int32_t fb_gain, bool add);
 };
 
 #endif  // ENGINEOPL_H_INCLUDED

third-party/Synth_Dexed/src/PluginFx.cpp

@@ -58,7 +58,7 @@ PluginFx::PluginFx() {
   Gain = 1.0;
 }
 
-void PluginFx::init(int sr) {
+void PluginFx::init(uint16_t sr) {
   mm = 0;
   s1 = s2 = s3 = s4 = c = d = 0;
   R24 = 0;
@@ -104,7 +104,7 @@ inline float PluginFx::NR(float sample, float g) {
   return y;
 }
 
-void PluginFx::process(float *work, int sampleSize) {
+void PluginFx::process(float *work, uint16_t sampleSize) {
   // very basic DC filter
   float t_fd = work[0];
   work[0] = work[0] - dc_id + dc_r * dc_od;
@@ -120,12 +120,12 @@ void PluginFx::process(float *work, int sampleSize) {
   // Gain
   if (Gain == 0.0)
   {
-    for (int i = 0; i < sampleSize; i++ )
+    for (uint16_t i = 0; i < sampleSize; i++ )
       work[i] = 0.0;
   }
   else if ( Gain != 1.0)
   {
-    for (int i = 0; i < sampleSize; i++ )
+    for (uint16_t i = 0; i < sampleSize; i++ )
       work[i] *= Gain;
   }
 
@@ -152,7 +152,7 @@ void PluginFx::process(float *work, int sampleSize) {
   float g = rCutoff;
   float lpc = g / (1 + g);
 
-  for (int i = 0; i < sampleSize; i++ ) {
+  for (uint16_t i = 0; i < sampleSize; i++ ) {
     float s = work[i];
     s = s - 0.45 * tptlpupw(c, s, 15, sampleRateInv);
     s = tptpc(d, s, bright);

third-party/Synth_Dexed/src/PluginFx.h

@@ -18,8 +18,9 @@
 
 */
 
-#ifndef PLUGINFX_H_INCLUDED
-#define PLUGINFX_H_INCLUDED
+#pragma once
+
+#include "stdint.h"
 
 class PluginFx {
     float s1, s2, s3, s4;
@@ -33,7 +34,7 @@ class PluginFx {
     // 24 db multimode
     float mm;
     float mmt;
-    int mmch;
+    int32_t mmch;
     inline float NR24(float sample, float g, float lpc);
 
     // preprocess values taken the UI
@@ -51,7 +52,7 @@ class PluginFx {
     inline float NR(float sample, float g);
     bool bandPassSw;
     float rcor, rcorInv;
-    int R;
+    int32_t R;
 
     float dc_id;
     float dc_od;
@@ -65,9 +66,7 @@ class PluginFx {
     float Reso;
     float Gain;
 
-    void init(int sampleRate);
-    void process(float *work, int sampleSize);
+    void init(uint16_t sampleRate);
+    void process(float *work, uint16_t sampleSize);
     float getGain(void);
 };
-
-#endif

third-party/Synth_Dexed/src/compressor.h

@@ -21,7 +21,7 @@
 #include "synth.h"
 
 /*
-static const float32_t zeroblock_f32[] = {
+static const float zeroblock_f32[] = {
     0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
 #if AUDIO_BLOCK_SAMPLES > 16
     0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
@@ -55,13 +55,13 @@ class Compressor
   //GUI: inputs:1, outputs:1  //this line used for automatic generation of GUI node
   public:
     //constructor
-    Compressor(const float32_t sample_rate_Hz) {
+    Compressor(const float sample_rate_Hz) {
 	  //setDefaultValues(AUDIO_SAMPLE_RATE);   resetStates();
 	  setDefaultValues(sample_rate_Hz);
           resetStates();
     };
 	
-    void setDefaultValues(const float32_t sample_rate_Hz) {
+    void setDefaultValues(const float sample_rate_Hz) {
       setThresh_dBFS(-20.0f);     //set the default value for the threshold for compression
       setCompressionRatio(5.0f);  //set the default copression ratio
       setAttack_sec(0.005f, sample_rate_Hz);  //default to this value
@@ -71,7 +71,7 @@ class Compressor
 
 
     //here's the method that does all the work
-    void doCompression(float32_t *audio_block, uint16_t len) {
+    void doCompression(float *audio_block, uint16_t len) {
       //Serial.println("AudioEffectGain_F32: updating.");  //for debugging.
       if (!audio_block) {
         printf("No audio_block available for Compressor!\n");
@@ -87,8 +87,7 @@ class Compressor
         arm_scale_f32(audio_block, pre_gain, audio_block, len); //use ARM DSP for speed!
 
       //calculate the level of the audio (ie, calculate a smoothed version of the signal power)
-      // float32_t* audio_level_dB_block = (float32_t*)malloc(sizeof(float32_t)*len); 
-      float32_t* audio_level_dB_block = new float32_t[len]; 
+      float* audio_level_dB_block = new float[len]; 
       if(!audio_level_dB_block)
       {
         printf("Cannot allocate memory for \"audio_level_dB_block\" - stopping\n");
@@ -101,8 +100,7 @@ class Compressor
       	calcAudioLevel_dB(audio_block, audio_level_dB_block, len); //returns through audio_level_dB_block
 
       //compute the desired gain based on the observed audio level
-      // float32_t* gain_block=(float32_t*)malloc(sizeof(float32_t)*len); 
-      float32_t* gain_block=new float32_t[len]; 
+      float* gain_block=new float[len]; 
       if(!gain_block)
       {
         printf("Cannot allocate memory for \"gain_block\" - stopping\n");
@@ -121,19 +119,18 @@ class Compressor
 
       //release memory
       if(audio_level_dB_block)
-	delete(audio_level_dB_block);
+	delete audio_level_dB_block;
       if(gain_block)
-        delete(gain_block);
+        delete gain_block;
     }
 
     // Here's the method that estimates the level of the audio (in dB)
     // It squares the signal and low-pass filters to get a time-averaged
     // signal power.  It then 
-    void calcAudioLevel_dB(float32_t *wav_block, float32_t *level_dB_block, uint16_t len) { 
+    void calcAudioLevel_dB(float *wav_block, float *level_dB_block, uint16_t len) { 
     	
       // calculate the instantaneous signal power (square the signal)
-      // float32_t* wav_pow_block=(float32_t*)malloc(sizeof(float32_t)*len); 
-      float32_t* wav_pow_block=new float32_t[len]; 
+      float* wav_pow_block=new float[len]; 
       if(!wav_pow_block)
       {
         printf("Cannot allocate memory for \"wav_pow_block\" - stopping\n");
@@ -145,7 +142,7 @@ class Compressor
       arm_mult_f32(wav_block, wav_block, wav_pow_block, len);
 
       // low-pass filter and convert to dB
-      float32_t c1 = level_lp_const, c2 = 1.0f - c1; //prepare constants
+      float c1 = level_lp_const, c2 = 1.0f - c1; //prepare constants
       for (uint16_t i = 0; i < len; i++) {
         // first-order low-pass filter to get a running estimate of the average power
         wav_pow_block[i] = c1*prev_level_lp_pow + c2*wav_pow_block[i];
@@ -165,18 +162,17 @@ class Compressor
 
       //release memory and return
       if(wav_pow_block)
-        delete(wav_pow_block);
+        delete wav_pow_block;
 
       return; //output is passed through level_dB_block
     }
 
     //This method computes the desired gain from the compressor, given an estimate
     //of the signal level (in dB)
-    void calcGain(float32_t *audio_level_dB_block, float32_t *gain_block,uint16_t len) { 
+    void calcGain(float *audio_level_dB_block, float *gain_block,uint16_t len) { 
     
       //first, calculate the instantaneous target gain based on the compression ratio
-      // float32_t* inst_targ_gain_dB_block=(float32_t*)malloc(sizeof(float32_t)*len);
-      float32_t* inst_targ_gain_dB_block=new float32_t[len];
+      float* inst_targ_gain_dB_block=new float[len];
       if(!inst_targ_gain_dB_block)
       {
 	printf("Cannot allocate memory for \"inst_targ_gain_dB_block\" - stopping\n");
@@ -187,8 +183,7 @@ class Compressor
       calcInstantaneousTargetGain(audio_level_dB_block, inst_targ_gain_dB_block,len);
     
       //second, smooth in time (attack and release) by stepping through each sample
-      // float32_t *gain_dB_block = (float32_t*)malloc(sizeof(float32_t)*len); 
-      float32_t *gain_dB_block = new float32_t[len]; 
+      float *gain_dB_block = new float[len]; 
       if(!gain_dB_block)
       {
 	printf("Cannot allocate memory for \"gain_dB_block\" - stopping\n");
@@ -205,20 +200,19 @@ class Compressor
 
       //release memory and return
       if(inst_targ_gain_dB_block)
-        delete(inst_targ_gain_dB_block);
+        delete inst_targ_gain_dB_block;
       if(gain_dB_block)
-        delete(gain_dB_block);
+        delete gain_dB_block;
 
       return;  //output is passed through gain_block
     }
       
     //Compute the instantaneous desired gain, including the compression ratio and
     //threshold for where the comrpession kicks in
-    void calcInstantaneousTargetGain(float32_t *audio_level_dB_block, float32_t *inst_targ_gain_dB_block, uint16_t len) {
+    void calcInstantaneousTargetGain(float *audio_level_dB_block, float *inst_targ_gain_dB_block, uint16_t len) {
       
       // how much are we above the compression threshold?
-      // float32_t* above_thresh_dB_block=(float32_t*)malloc(sizeof(float32_t)*len);
-      float32_t* above_thresh_dB_block=new float32_t[len];
+      float* above_thresh_dB_block=new float[len];
       if(!above_thresh_dB_block)
       {
         printf("Cannot allocate memory for \"above_thresh_dB_block\" - stopping\n");
@@ -251,17 +245,17 @@ class Compressor
 
       // release memory before returning
       if(above_thresh_dB_block)
-        delete(above_thresh_dB_block);
+        delete above_thresh_dB_block;
 
       return;  //output is passed through inst_targ_gain_dB_block
     }
 
     //this method applies the "attack" and "release" constants to smooth the
     //target gain level through time.
-    void calcSmoothedGain_dB(float32_t *inst_targ_gain_dB_block, float32_t *gain_dB_block, uint16_t len) {
-      float32_t gain_dB;
-      float32_t one_minus_attack_const = 1.0f - attack_const;
-      float32_t one_minus_release_const = 1.0f - release_const;
+    void calcSmoothedGain_dB(float *inst_targ_gain_dB_block, float *gain_dB_block, uint16_t len) {
+      float gain_dB;
+      float one_minus_attack_const = 1.0f - attack_const;
+      float one_minus_release_const = 1.0f - release_const;
       for (uint16_t i = 0; i < len; i++) {
         gain_dB = inst_targ_gain_dB_block[i];
 
@@ -288,48 +282,48 @@ class Compressor
       //initialize the HP filter.  (This also resets the filter states,)
       arm_biquad_cascade_df1_init_f32(&hp_filt_struct, hp_nstages, hp_coeff, hp_state);
     }
-    void setPreGain(float32_t g) {  pre_gain = g;  }
-    void setPreGain_dB(float32_t gain_dB) { setPreGain(pow(10.0, gain_dB / 20.0));  }
-    void setCompressionRatio(float32_t cr) {
-      comp_ratio = max(0.001f, cr); //limit to positive values
+    void setPreGain(float g) {  pre_gain = g;  }
+    void setPreGain_dB(float gain_dB) { setPreGain(pow(10.0, gain_dB / 20.0));  }
+    void setCompressionRatio(float cr) {
+      comp_ratio = std::max(0.001f, cr); //limit to positive values
       updateThresholdAndCompRatioConstants();
     }
-    void setAttack_sec(float32_t a, float32_t fs_Hz) {
+    void setAttack_sec(float a, float fs_Hz) {
       attack_sec = a;
       attack_const = expf(-1.0f / (attack_sec * fs_Hz)); //expf() is much faster than exp()
 
       //also update the time constant for the envelope extraction
-      setLevelTimeConst_sec(min(attack_sec,release_sec) / 5.0, fs_Hz);  //make the level time-constant one-fifth the gain time constants
+      setLevelTimeConst_sec(std::min(attack_sec,release_sec) / 5.0, fs_Hz);  //make the level time-constant one-fifth the gain time constants
     } 
-    void setRelease_sec(float32_t r, float32_t fs_Hz) {
+    void setRelease_sec(float r, float fs_Hz) {
       release_sec = r;
       release_const = expf(-1.0f / (release_sec * fs_Hz)); //expf() is much faster than exp()
 
       //also update the time constant for the envelope extraction
-      setLevelTimeConst_sec(min(attack_sec,release_sec) / 5.0, fs_Hz);  //make the level time-constant one-fifth the gain time constants
+      setLevelTimeConst_sec(std::min(attack_sec,release_sec) / 5.0, fs_Hz);  //make the level time-constant one-fifth the gain time constants
     }
-    void setLevelTimeConst_sec(float32_t t_sec, float32_t fs_Hz) {
-      const float32_t min_t_sec = 0.002f;  //this is the minimum allowed value
-      level_lp_sec = max(min_t_sec,t_sec);
+    void setLevelTimeConst_sec(float t_sec, float fs_Hz) {
+      const float min_t_sec = 0.002f;  //this is the minimum allowed value
+      level_lp_sec = std::max(min_t_sec,t_sec);
       level_lp_const = expf(-1.0f / (level_lp_sec * fs_Hz)); //expf() is much faster than exp()
     }
-    void setThresh_dBFS(float32_t val) { 
+    void setThresh_dBFS(float val) { 
       thresh_dBFS = val;
       setThreshPow(pow(10.0, thresh_dBFS / 10.0));
     }
     void enableHPFilter(boolean flag) { use_HP_prefilter = flag; };
 
     //methods to return information about this module
-    float32_t getPreGain_dB(void) { return 20.0 * log10f_approx(pre_gain);  }
-    float32_t getAttack_sec(void) {  return attack_sec; }
-    float32_t getRelease_sec(void) {  return release_sec; }
-    float32_t getLevelTimeConst_sec(void) { return level_lp_sec; }
-    float32_t getThresh_dBFS(void) { return thresh_dBFS; }
-    float32_t getCompressionRatio(void) { return comp_ratio; }
-    float32_t getCurrentLevel_dBFS(void) { return 10.0* log10f_approx(prev_level_lp_pow); }
-    float32_t getCurrentGain_dB(void) { return prev_gain_dB; }
-
-    void setHPFilterCoeff_N2IIR_Matlab(float32_t b[], float32_t a[]){
+    float getPreGain_dB(void) { return 20.0 * log10f_approx(pre_gain);  }
+    float getAttack_sec(void) {  return attack_sec; }
+    float getRelease_sec(void) {  return release_sec; }
+    float getLevelTimeConst_sec(void) { return level_lp_sec; }
+    float getThresh_dBFS(void) { return thresh_dBFS; }
+    float getCompressionRatio(void) { return comp_ratio; }
+    float getCurrentLevel_dBFS(void) { return 10.0* log10f_approx(prev_level_lp_pow); }
+    float getCurrentGain_dB(void) { return prev_gain_dB; }
+
+    void setHPFilterCoeff_N2IIR_Matlab(float b[], float a[]){
       //https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
       //Use matlab to compute the coeff for HP at 20Hz: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
       hp_coeff[0] = b[0];   hp_coeff[1] = b[1];  hp_coeff[2] = b[2]; //here are the matlab "b" coefficients
@@ -338,20 +332,20 @@ class Compressor
     
   private:
     //state-related variables
-    float32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module
-    float32_t prev_level_lp_pow = 1.0;
-    float32_t prev_gain_dB = 0.0; //last gain^2 used
+    float *inputQueueArray_f32[1]; //memory pointer for the input to this module
+    float prev_level_lp_pow = 1.0;
+    float prev_gain_dB = 0.0; //last gain^2 used
 
     //HP filter state-related variables
     arm_biquad_casd_df1_inst_f32 hp_filt_struct;
     static const uint8_t hp_nstages = 1;
-    float32_t hp_coeff[5 * hp_nstages] = {1.0, 0.0, 0.0, 0.0, 0.0}; //no filtering. actual filter coeff set later
-    float32_t hp_state[4 * hp_nstages];
+    float hp_coeff[5 * hp_nstages] = {1.0, 0.0, 0.0, 0.0, 0.0}; //no filtering. actual filter coeff set later
+    float hp_state[4 * hp_nstages];
     void setHPFilterCoeff(void) {
       //https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
       //Use matlab to compute the coeff for HP at 20Hz: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
-      float32_t b[] = {9.979871156751189e-01,    -1.995974231350238e+00, 9.979871156751189e-01};  //from Matlab
-      float32_t a[] = { 1.000000000000000e+00,    -1.995970179642828e+00,    9.959782830576472e-01};  //from Matlab
+      float b[] = {9.979871156751189e-01,    -1.995974231350238e+00, 9.979871156751189e-01};  //from Matlab
+      float a[] = { 1.000000000000000e+00,    -1.995970179642828e+00,    9.959782830576472e-01};  //from Matlab
       setHPFilterCoeff_N2IIR_Matlab(b, a);
       //hp_coeff[0] = b[0];   hp_coeff[1] = b[1];  hp_coeff[2] = b[2]; //here are the matlab "b" coefficients
       //hp_coeff[3] = -a[1];  hp_coeff[4] = -a[2];  //the DSP needs the "a" terms to have opposite sign vs Matlab
@@ -359,34 +353,34 @@ class Compressor
 
 
     //private parameters related to gain calculation
-    float32_t attack_const, release_const, level_lp_const; //used in calcGain().  set by setAttack_sec() and setRelease_sec();
-    float32_t comp_ratio_const, thresh_pow_FS_wCR;  //used in calcGain();  set in updateThresholdAndCompRatioConstants()
+    float attack_const, release_const, level_lp_const; //used in calcGain().  set by setAttack_sec() and setRelease_sec();
+    float comp_ratio_const, thresh_pow_FS_wCR;  //used in calcGain();  set in updateThresholdAndCompRatioConstants()
     void updateThresholdAndCompRatioConstants(void) {
       comp_ratio_const = 1.0f-(1.0f / comp_ratio);
       thresh_pow_FS_wCR = powf(thresh_pow_FS, comp_ratio_const);    
     }
 
     //settings
-    float32_t attack_sec, release_sec, level_lp_sec; 
-    float32_t thresh_dBFS = 0.0;  //threshold for compression, relative to digital full scale
-    float32_t thresh_pow_FS = 1.0f;  //same as above, but not in dB
-    void setThreshPow(float32_t t_pow) { 
+    float attack_sec, release_sec, level_lp_sec; 
+    float thresh_dBFS = 0.0;  //threshold for compression, relative to digital full scale
+    float thresh_pow_FS = 1.0f;  //same as above, but not in dB
+    void setThreshPow(float t_pow) { 
       thresh_pow_FS = t_pow;
       updateThresholdAndCompRatioConstants();
     }
-    float32_t comp_ratio = 1.0;  //compression ratio
-    float32_t pre_gain = -1.0;  //gain to apply before the compression.  negative value disables
+    float comp_ratio = 1.0;  //compression ratio
+    float pre_gain = -1.0;  //gain to apply before the compression.  negative value disables
     boolean use_HP_prefilter;
     
     
     // Accelerate the powf(10.0,x) function
-    static float32_t pow10f(float32_t x) {
+    static float pow10f(float x) {
       //return powf(10.0f,x)   //standard, but slower
       return expf(2.302585092994f*x);  //faster:  exp(log(10.0f)*x)
     }
 
     // Accelerate the log10f(x)  function?
-    static float32_t log10f_approx(float32_t x) {
+    static float log10f_approx(float x) {
       //return log10f(x);   //standard, but slower
       return log2f_approx(x)*0.3010299956639812f; //faster:  log2(x)/log2(10)
     }
@@ -400,12 +394,12 @@ class Compressor
     ** when computing db20() is accurate to 7.984884e-003 dB.
     ** ------------------------------------------------------------------- */
     //https://community.arm.com/tools/f/discussions/4292/cmsis-dsp-new-functionality-proposal/22621#22621
-    //float32_t log2f_approx_coeff[4] = {1.23149591368684f, -4.11852516267426f, 6.02197014179219f, -3.13396450166353f};
-    static float32_t log2f_approx(float32_t X) {
-      //float32_t *C = &log2f_approx_coeff[0];
-      float32_t Y;
-      float32_t F;
-      int E;
+    //float log2f_approx_coeff[4] = {1.23149591368684f, -4.11852516267426f, 6.02197014179219f, -3.13396450166353f};
+    static float log2f_approx(float X) {
+      //float *C = &log2f_approx_coeff[0];
+      float Y;
+      float F;
+      int32_t E;
     
       // This is the approximation to log2()
       F = frexpf(fabsf(X), &E);

third-party/Synth_Dexed/src/controllers.h

@@ -25,10 +25,10 @@
 #include "fm_core.h"
 
 // State of MIDI controllers
-const int kControllerPitch = 0;
-const int kControllerPitchRange = 1;
-const int kControllerPitchStep = 2;
-const int kControllerPortamentoGlissando = 3;
+const uint8_t kControllerPitch = 0;
+const uint8_t kControllerPitchRange = 1;
+const uint8_t kControllerPitchStep = 2;
+const uint8_t kControllerPortamentoGlissando = 3;
 
 class FmMod {
   public:
@@ -96,13 +96,13 @@ class Controllers {
       }
 
       if (mod.amp)
-        amp_mod = max(amp_mod, total);
+        amp_mod = std::max(amp_mod, total);
 
       if (mod.pitch)
-        pitch_mod = max(pitch_mod, total);
+        pitch_mod = std::max(pitch_mod, total);
 
       if (mod.eg)
-        eg_mod = max(eg_mod, total);
+        eg_mod = std::max(eg_mod, total);
     }
 
   public:
@@ -117,9 +117,9 @@ class Controllers {
     uint8_t foot_cc;
     uint8_t modwheel_cc;
     bool portamento_enable_cc;
-    int portamento_cc;
+    int32_t portamento_cc;
     bool portamento_gliss_cc;
-    int masterTune;
+    int32_t masterTune;
 
     uint8_t opSwitch;
 

third-party/Synth_Dexed/src/dexed.cpp

@@ -38,7 +38,7 @@
 #include "porta.h"
 #include "compressor.h"
 
-Dexed::Dexed(uint8_t maxnotes, int rate)
+Dexed::Dexed(uint8_t maxnotes, uint16_t rate)
 {
   samplerate = float32_t(rate);
 
@@ -53,7 +53,6 @@ Dexed::Dexed(uint8_t maxnotes, int rate)
   Porta::init_sr(rate);
   fx.init(rate);
 
-  max_notes = maxnotes;
   currentNote = 0;
   resetControllers();
   controllers.masterTune = 0;
@@ -64,7 +63,23 @@ Dexed::Dexed(uint8_t maxnotes, int rate)
   sustain = false;
   voices = NULL;
 
-  setMaxNotes(max_notes);
+  max_notes=maxnotes;
+  if (max_notes > 0)
+  {
+    voices = new ProcessorVoice[max_notes]; // sizeof(ProcessorVoice) = 20
+    for (uint8_t i = 0; i < max_notes; i++)
+    {
+      voices[i].dx7_note = new Dx7Note; // sizeof(Dx7Note) = 692
+      voices[i].keydown = false;
+      voices[i].sustained = false;
+      voices[i].live = false;
+      voices[i].key_pressed_timer = 0;
+    }
+  }
+  else
+    voices = NULL;
+
+  used_notes=max_notes;
   setMonoMode(false);
   loadInitVoice();
 
@@ -91,9 +106,7 @@ Dexed::~Dexed()
 
   for (uint8_t note = 0; note < max_notes; note++)
     delete voices[note].dx7_note;
-
-  for (uint8_t note = 0; note < max_notes; note++)
-    delete &voices[note];
+  delete[] voices;
 }
 
 void Dexed::setEngineType(uint8_t engine)
@@ -103,19 +116,19 @@ void Dexed::setEngineType(uint8_t engine)
   switch(engine)
   {
     case MSFA:
-      controllers.core = (EngineMkI*)engineMsfa;
+      controllers.core = (FmCore*)engineMsfa;
       engineType=MSFA;
       break;
     case MKI:
-      controllers.core = (EngineMkI*)engineMkI;
+      controllers.core = (FmCore*)engineMkI;
       engineType=MKI;
       break;
     case OPL:
-      controllers.core = (EngineMkI*)engineOpl;
+      controllers.core = (FmCore*)engineOpl;
       engineType=OPL;
       break;
     default:
-      controllers.core = (EngineMkI*)engineMsfa;
+      controllers.core = (FmCore*)engineMsfa;
       engineType=MSFA;
       break;
   }
@@ -135,44 +148,8 @@ FmCore* Dexed::getEngineAddress(void)
 
 void Dexed::setMaxNotes(uint8_t new_max_notes)
 {
-  uint8_t i = 0;
-
-  max_notes = constrain(max_notes, 0, _MAX_NOTES);
-
-#if defined(MICRODEXED_VERSION) && defined(DEBUG)
-  Serial.print("Allocating memory for ");
-  Serial.print(max_notes, DEC);
-  Serial.println(" notes.");
-  Serial.println();
-#endif
-
-  if (voices)
-  {
   panic();
-    for (i = 0; i < max_notes; i++)
-    {
-      if (voices[i].dx7_note)
-        delete voices[i].dx7_note;
-    }
-    delete(voices);
-  }
-
-  max_notes = constrain(new_max_notes, 0, _MAX_NOTES);
-
-  if (max_notes > 0)
-  {
-    voices = new ProcessorVoice[max_notes]; // sizeof(ProcessorVoice) = 20
-    for (i = 0; i < max_notes; i++)
-    {
-      voices[i].dx7_note = new Dx7Note; // sizeof(Dx7Note) = 692
-      voices[i].keydown = false;
-      voices[i].sustained = false;
-      voices[i].live = false;
-      voices[i].key_pressed_timer = 0;
-    }
-  }
-  else
-    voices = NULL;
+  used_notes = constrain(new_max_notes, 0, max_notes);
 }
 
 void Dexed::activate(void)
@@ -186,11 +163,11 @@ void Dexed::deactivate(void)
   panic();
 }
 
-void Dexed::getSamples(float32_t* buffer, uint16_t n_samples)
+void Dexed::getSamples(float* buffer, uint16_t n_samples)
 {
   if (refreshVoice)
   {
-    for (uint8_t i = 0; i < max_notes; i++)
+    for (uint8_t i = 0; i < used_notes; i++)
     {
       if ( voices[i].live )
         voices[i].dx7_note->update(data, voices[i].midi_note, voices[i].velocity, voices[i].porta, &controllers);
@@ -213,7 +190,7 @@ void Dexed::getSamples(float32_t* buffer, uint16_t n_samples)
     int32_t lfovalue = lfo.getsample();
     int32_t lfodelay = lfo.getdelay();
 
-    for (uint8_t note = 0; note < max_notes; note++)
+    for (uint8_t note = 0; note < used_notes; note++)
     {
       if (voices[note].live)
       {
@@ -238,13 +215,13 @@ void Dexed::getSamples(float32_t* buffer, uint16_t n_samples)
 
 void Dexed::getSamples(int16_t* buffer, uint16_t n_samples)
 {
-  float32_t tmp[n_samples];
+  float tmp[n_samples];
 
   getSamples(tmp, n_samples);
   arm_float_to_q15(tmp, (q15_t*)buffer, n_samples);
 }
 
-void Dexed::keydown(int16_t pitch, uint8_t velo) {
+void Dexed::keydown(uint8_t pitch, uint8_t velo) {
   if ( velo == 0 ) {
     keyup(pitch);
     return;
@@ -254,10 +231,10 @@ void Dexed::keydown(int16_t pitch, uint8_t velo) {
 
   pitch += data[144] - TRANSPOSE_FIX;
 
-  int previousKeyDown = lastKeyDown;
+  int32_t previousKeyDown = lastKeyDown;
   lastKeyDown = pitch;
 
-  int porta = -1;
+  int32_t porta = -1;
   if ( controllers.portamento_enable_cc && previousKeyDown >= 0 )
     porta = controllers.portamento_cc;
 
@@ -266,7 +243,7 @@ void Dexed::keydown(int16_t pitch, uint8_t velo) {
 
   if (!monoMode && noteRefreshMode)
   {
-    for (uint8_t i = 0; i < max_notes; i++)
+    for (uint8_t i = 0; i < used_notes; i++)
     {
       if (voices[i].midi_note == pitch && voices[i].keydown == false && voices[i].live && voices[i].sustained == true)
       {
@@ -284,17 +261,17 @@ void Dexed::keydown(int16_t pitch, uint8_t velo) {
     }
   }
 
-  for (uint8_t i = 0; i <= max_notes; i++)
+  for (uint8_t i = 0; i <= used_notes; i++)
   {
-    if (i == max_notes)
+    if (i == used_notes)
     {
-      uint32_t min_timer = 0xffff;
+      uint32_t min_timer = 0xffffffff;
 
       if (monoMode)
         break;
 
       // no free sound slot found, so use the oldest note slot
-      for (uint8_t n = 0; n < max_notes; n++)
+      for (uint8_t n = 0; n < used_notes; n++)
       {
         if (voices[n].key_pressed_timer < min_timer)
         {
@@ -311,14 +288,14 @@ void Dexed::keydown(int16_t pitch, uint8_t velo) {
 
     if (!voices[note].keydown)
     {
-      currentNote = (note + 1) % max_notes;
+      currentNote = (note + 1) % used_notes;
       //if (keydown_counter == 0) // Original comment: TODO: should only do this if # keys down was 0
         lfo.keydown();
       voices[note].midi_note = pitch;
       voices[note].velocity = velo;
       voices[note].sustained = sustain;
       voices[note].keydown = true;
-      int srcnote = (previousKeyDown >= 0) ? previousKeyDown : pitch;
+      int32_t srcnote = (previousKeyDown >= 0) ? previousKeyDown : pitch;
       voices[note].dx7_note->init(data, pitch, velo, srcnote, porta, &controllers);
       if ( data[136] )
         voices[note].dx7_note->oscSync();
@@ -330,11 +307,11 @@ void Dexed::keydown(int16_t pitch, uint8_t velo) {
     {
       keydown_counter++;
     }
-    note = (note + 1) % max_notes;
+    note = (note + 1) % used_notes;
   }
 
   if ( monoMode ) {
-    for (uint8_t i = 0; i < max_notes; i++) {
+    for (uint8_t i = 0; i < used_notes; i++) {
       if ( voices[i].live ) {
         // all keys are up, only transfer signal
         if ( ! voices[i].keydown ) {
@@ -355,14 +332,14 @@ void Dexed::keydown(int16_t pitch, uint8_t velo) {
   voices[note].live = true;
 }
 
-void Dexed::keyup(int16_t pitch) {
+void Dexed::keyup(uint8_t pitch) {
   uint8_t note;
 
   pitch = constrain(pitch, 0, 127);
 
   pitch += data[144] - TRANSPOSE_FIX;
 
-  for (note = 0; note < max_notes; note++) {
+  for (note = 0; note < used_notes; note++) {
     if ( voices[note].midi_note == pitch && voices[note].keydown ) {
       voices[note].keydown = false;
       voices[note].key_pressed_timer = 0;
@@ -372,14 +349,14 @@ void Dexed::keyup(int16_t pitch) {
   }
 
   // note not found ?
-  if ( note >= max_notes ) {
+  if ( note >= used_notes ) {
     return;
   }
 
   if ( monoMode ) {
-    int16_t highNote = -1;
+    int8_t highNote = -1;
     uint8_t target = 0;
-    for (int8_t i = 0; i < max_notes; i++) {
+    for (int8_t i = 0; i < used_notes; i++) {
       if ( voices[i].keydown && voices[i].midi_note > highNote ) {
         target = i;
         highNote = voices[i].midi_note;
@@ -496,7 +473,7 @@ void Dexed::notesOff(void) {
 
 uint8_t Dexed::getMaxNotes(void)
 {
-  return max_notes;
+  return used_notes;
 }
 
 uint8_t Dexed::getNumNotesPlaying(void)
@@ -505,7 +482,7 @@ uint8_t Dexed::getNumNotesPlaying(void)
   uint8_t i;
   uint8_t count_playing_voices = 0;
 
-  for (i = 0; i < max_notes; i++)
+  for (i = 0; i < used_notes; i++)
   {
     if (voices[i].live == true)
     {
@@ -613,7 +590,7 @@ bool Dexed::decodeVoice(uint8_t* new_data, uint8_t* encoded_data)
   panic();
   doRefreshVoice();
 
-  strncpy(dexed_voice_name, (char *)&encoded_data[118], sizeof(dexed_voice_name) - 1);
+  strlcpy(dexed_voice_name, (char *)&encoded_data[118], sizeof(dexed_voice_name) - 1);
   dexed_voice_name[10] = '\0';
 #if defined(MICRODEXED_VERSION) && defined(DEBUG)
   Serial.print(F("Voice ["));
@@ -703,7 +680,7 @@ void Dexed::loadVoiceParameters(uint8_t* new_data)
   memcpy(&data, new_data, 155);
   doRefreshVoice();
 #if defined(MICRODEXED_VERSION) && defined(DEBUG)
-  strncpy(dexed_voice_name, (char *)&new_data[145], sizeof(dexed_voice_name) - 1);
+  strlcpy(dexed_voice_name, (char *)&new_data[145], sizeof(dexed_voice_name) - 1);
   dexed_voice_name[10] = '\0';
 
   Serial.print(F("Voice ["));
@@ -1707,12 +1684,12 @@ uint8_t Dexed::getTranspose(void)
 
 void Dexed::setName(char* name)
 {
-  strncpy(name, (char*)&data[DEXED_VOICE_OFFSET + DEXED_NAME], 10);
+  strlcpy(name, (char*)&data[DEXED_VOICE_OFFSET + DEXED_NAME], 10);
 }
 
 void Dexed::getName(char* buffer)
 {
-  strncpy((char*)&data[DEXED_VOICE_OFFSET + DEXED_NAME], buffer, 10);
+  strlcpy((char*)&data[DEXED_VOICE_OFFSET + DEXED_NAME], buffer, 10);
   buffer[10] = '\0';
 }
 
@@ -1764,52 +1741,52 @@ bool Dexed::getCompressor(void)
   return (use_compressor);
 }
 
-void Dexed::setCompressorPreGain_dB(float32_t pre_gain)
+void Dexed::setCompressorPreGain_dB(float pre_gain)
 {
   compressor->setPreGain_dB(pre_gain);
 }
 
-void Dexed::setCompressorAttack_sec(float32_t attack_sec)
+void Dexed::setCompressorAttack_sec(float attack_sec)
 {
   compressor->setAttack_sec(attack_sec, samplerate);
 }
 
-void Dexed::setCompressorRelease_sec(float32_t release_sec)
+void Dexed::setCompressorRelease_sec(float release_sec)
 {
   compressor->setRelease_sec(release_sec, samplerate);
 }
 
-void Dexed::setCompressorThresh_dBFS(float32_t thresh_dBFS)
+void Dexed::setCompressorThresh_dBFS(float thresh_dBFS)
 {
   compressor->setThresh_dBFS(thresh_dBFS);
 }
 
-void Dexed::setCompressionRatio(float32_t comp_ratio)
+void Dexed::setCompressionRatio(float comp_ratio)
 {
   compressor->setCompressionRatio(comp_ratio);
 }
 
-float32_t Dexed::getCompressorPreGain_dB(void)
+float Dexed::getCompressorPreGain_dB(void)
 {
   return (compressor->getPreGain_dB());
 }
 
-float32_t Dexed::getCompressorAttack_sec(void)
+float Dexed::getCompressorAttack_sec(void)
 {
   return (compressor->getAttack_sec());
 }
 
-float32_t Dexed::getCompressorRelease_sec(void)
+float Dexed::getCompressorRelease_sec(void)
 {
   return (compressor->getRelease_sec());
 }
 
-float32_t Dexed::getCompressorThresh_dBFS(void)
+float Dexed::getCompressorThresh_dBFS(void)
 {
   return (compressor->getThresh_dBFS());
 }
 
-float32_t Dexed::getCompressionRatio(void)
+float Dexed::getCompressionRatio(void)
 {
   return (compressor->getCompressionRatio());
 }

third-party/Synth_Dexed/src/dexed.h

@@ -51,7 +51,7 @@
 #define NUM_VOICE_PARAMETERS 156
 
 struct ProcessorVoice {
-  int16_t midi_note;
+  uint8_t midi_note;
   uint8_t velocity;
   int16_t porta;
   bool keydown;
@@ -165,7 +165,7 @@ enum ENGINES {
 class Dexed
 {
   public:
-    Dexed(uint8_t maxnotes, int rate);
+    Dexed(uint8_t maxnotes, uint16_t rate);
     ~Dexed();
 
     // Global methods
@@ -174,7 +174,6 @@ class Dexed
     bool getMonoMode(void);
     void setMonoMode(bool mode);
     void setNoteRefreshMode(bool mode);
-    void setMaxNotes(uint8_t n);
     uint8_t getMaxNotes(void);
     void doRefreshVoice(void);
     void setOPAll(uint8_t ops);
@@ -193,28 +192,29 @@ class Dexed
     void setVelocityScale(uint8_t offset, uint8_t max);
     void getVelocityScale(uint8_t* offset, uint8_t* max);
     void setVelocityScale(uint8_t setup);
+    void setMaxNotes(uint8_t n);
     void setEngineType(uint8_t engine);
     uint8_t getEngineType(void);
     FmCore* getEngineAddress(void);
 #ifndef TEENSYDUINO
     void setCompressor(bool comp);
     bool getCompressor(void);
-    void setCompressorPreGain_dB(float32_t pre_gain);
-    void setCompressorAttack_sec(float32_t attack_sec);
-    void setCompressorRelease_sec(float32_t release_sec);
-    void setCompressorThresh_dBFS(float32_t thresh_dBFS);
-    void setCompressionRatio(float32_t comp_ratio);
-    float32_t getCompressorPreGain_dB(void);
-    float32_t getCompressorAttack_sec(void);
-    float32_t getCompressorRelease_sec(void);
-    float32_t getCompressorThresh_dBFS(void);
-    float32_t getCompressionRatio(void);
+    void setCompressorPreGain_dB(float pre_gain);
+    void setCompressorAttack_sec(float attack_sec);
+    void setCompressorRelease_sec(float release_sec);
+    void setCompressorThresh_dBFS(float thresh_dBFS);
+    void setCompressionRatio(float comp_ratio);
+    float getCompressorPreGain_dB(void);
+    float getCompressorAttack_sec(void);
+    float getCompressorRelease_sec(void);
+    float getCompressorThresh_dBFS(void);
+    float getCompressionRatio(void);
 #endif
     int16_t checkSystemExclusive(const uint8_t* sysex, const uint16_t len);
 
     // Sound methods
-    void keyup(int16_t pitch);
-    void keydown(int16_t pitch, uint8_t velo);
+    void keyup(uint8_t pitch);
+    void keydown(uint8_t pitch, uint8_t velo);
     void setSustain(bool sustain);
     bool getSustain(void);
     void panic(void);
@@ -355,12 +355,13 @@ class Dexed
       03, 48,                                                                             // pitch_mod_sensitivity, transpose
       73, 78, 73, 84, 32, 86, 79, 73, 67, 69                                              // 10 * char for name ("INIT VOICE")
     };
-    float32_t samplerate;
+    float samplerate;
     uint8_t data[NUM_VOICE_PARAMETERS];
     uint8_t max_notes;
+    uint8_t used_notes;
     PluginFx fx;
     Controllers controllers;
-    int lastKeyDown;
+    int32_t lastKeyDown;
     uint32_t xrun;
     uint16_t render_time_max;
     int16_t currentNote;
@@ -375,9 +376,9 @@ class Dexed
     EngineMsfa* engineMsfa;
     EngineMkI* engineMkI;
     EngineOpl* engineOpl;
-    void getSamples(float32_t* buffer, uint16_t n_samples);
+    void getSamples(float* buffer, uint16_t n_samples);
     void getSamples(int16_t* buffer, uint16_t n_samples);
-    void compress(float32_t* wav_in, float32_t* wav_out, uint16_t n, float32_t threshold, float32_t slope, uint16_t sr,  float32_t tla, float32_t twnd, float32_t tatt, float32_t trel);
+    void compress(float* wav_in, float* wav_out, uint16_t n, float threshold, float slope, uint16_t sr,  float tla, float twnd, float tatt, float trel);
     bool use_compressor;
     uint8_t velocity_offset;
     uint8_t velocity_max;

third-party/Synth_Dexed/src/dx7note.cpp

@@ -87,14 +87,14 @@ const uint8_t velocity_data[64] = {
 
 // See "velocity" section of notes. Returns velocity delta in microsteps.
 int ScaleVelocity(int velocity, int sensitivity) {
-  int clamped_vel = max(0, min(127, velocity));
+  int clamped_vel = std::max(0, std::min(127, velocity));
   int vel_value = velocity_data[clamped_vel >> 1] - 239;
   int scaled_vel = ((sensitivity * vel_value + 7) >> 3) << 4;
   return scaled_vel;
 }
 
 int ScaleRate(int midinote, int sensitivity) {
-  int x = min(31, max(0, midinote / 3 - 7));
+  int x = std::min(31, std::max(0, midinote / 3 - 7));
   int qratedelta = (sensitivity * x) >> 3;
 #ifdef SUPER_PRECISE
   int rem = x & 7;
@@ -120,7 +120,7 @@ int ScaleCurve(int group, int depth, int curve) {
   } else {
     // exponential
     int n_scale_data = sizeof(exp_scale_data);
-    int raw_exp = exp_scale_data[min(group, n_scale_data - 1)];
+    int raw_exp = exp_scale_data[std::min(group, n_scale_data - 1)];
     scale = (raw_exp * depth * 329) >> 15;
   }
   if (curve < 2) {
@@ -170,12 +170,12 @@ void Dx7Note::init(const uint8_t patch[156], int midinote, int velocity, int src
     int level_scaling = ScaleLevel(midinote, patch[off + 8], patch[off + 9],
                                    patch[off + 10], patch[off + 11], patch[off + 12]);
     outlevel += level_scaling;
-    outlevel = min(127, outlevel);
+    outlevel = std::min(127, outlevel);
     outlevel = outlevel << 5;
     outlevel += ScaleVelocity(velocity, patch[off + 15]);
-    outlevel = max(0, outlevel);
+    outlevel = std::max(0, outlevel);
     int rate_scaling = ScaleRate(midinote, patch[off + 13]);
-    env_[op].init(rates, levels, outlevel, rate_scaling);
+    env_[op].init((const int32_t*)rates, (const int32_t*)levels, outlevel, rate_scaling);
 
     int mode = patch[off + 17];
     int coarse = patch[off + 18];
@@ -194,7 +194,7 @@ void Dx7Note::init(const uint8_t patch[156], int midinote, int velocity, int src
     rates[i] = patch[126 + i];
     levels[i] = patch[130 + i];
   }
-  pitchenv_.set(rates, levels);
+  pitchenv_.set((const int32_t *)rates, (const int32_t*)levels);
   algorithm_ = patch[134];
   int feedback = patch[135];
   fb_shift_ = feedback != 0 ? FEEDBACK_BITDEPTH - feedback : 16;
@@ -213,7 +213,7 @@ void Dx7Note::compute(int32_t *buf, int32_t lfo_val, int32_t lfo_delay, const Co
   pmod_1 = abs(pmod_1);
   int32_t pmod_2 = (int32_t)(((int64_t)ctrls->pitch_mod * (int64_t)senslfo) >> 14);
   pmod_2 = abs(pmod_2);
-  int32_t pitch_mod = max(pmod_1, pmod_2);
+  int32_t pitch_mod = std::max(pmod_1, pmod_2);
   pitch_mod = pitchenv_.getsample() + (pitch_mod * (senslfo < 0 ? -1 : 1));
 
   // ---- PITCH BEND ----
@@ -236,11 +236,11 @@ void Dx7Note::compute(int32_t *buf, int32_t lfo_val, int32_t lfo_delay, const Co
   uint32_t amod_1 = (uint32_t)(((int64_t) ampmoddepth_ * (int64_t) lfo_delay) >> 8); // Q24 :D
   amod_1 = (uint32_t)(((int64_t) amod_1 * (int64_t) lfo_val) >> 24);
   uint32_t amod_2 = (uint32_t)(((int64_t) ctrls->amp_mod * (int64_t) lfo_val) >> 7); // Q?? :|
-  uint32_t amd_mod = max(amod_1, amod_2);
+  uint32_t amd_mod = std::max(amod_1, amod_2);
 
   // ==== EG AMP MOD ====
   uint32_t amod_3 = (ctrls->eg_mod + 1) << 17;
-  amd_mod = max((1 << 24) - amod_3, amd_mod);
+  amd_mod = std::max((1 << 24) - amod_3, amd_mod);
 
   // ==== OP RENDER ====
   for (int op = 0; op < 6; op++) {
@@ -330,12 +330,12 @@ void Dx7Note::update(const uint8_t patch[156], int midinote, int velocity, int p
     int level_scaling = ScaleLevel(midinote, patch[off + 8], patch[off + 9],
                                    patch[off + 10], patch[off + 11], patch[off + 12]);
     outlevel += level_scaling;
-    outlevel = min(127, outlevel);
+    outlevel = std::min(127, outlevel);
     outlevel = outlevel << 5;
     outlevel += ScaleVelocity(velocity, patch[off + 15]);
-    outlevel = max(0, outlevel);
+    outlevel = std::max(0, outlevel);
     int rate_scaling = ScaleRate(midinote, patch[off + 13]);
-    env_[op].update(rates, levels, outlevel, rate_scaling);
+    env_[op].update((const int32_t*)rates, (const int32_t*)levels, (int32_t)outlevel, rate_scaling);
   }
   algorithm_ = patch[134];
   int feedback = patch[135];

third-party/Synth_Dexed/src/dx7note.h

@@ -64,7 +64,7 @@ class Dx7Note {
     FmOpParams params_[6];
     PitchEnv pitchenv_;
     int32_t basepitch_[6];
-    int32_t fb_buf_[2];
+    int32_t fb_buf_[2]={0 ,0};
     int32_t fb_shift_;
     int32_t ampmodsens_[6];
     int32_t opMode[6];

third-party/Synth_Dexed/src/env.cpp

@@ -24,12 +24,12 @@
 
 uint32_t Env::sr_multiplier = (1 << 24);
 
-const int levellut[] = {
+const int32_t levellut[] = {
   0, 5, 9, 13, 17, 20, 23, 25, 27, 29, 31, 33, 35, 37, 39, 41, 42, 43, 45, 46
 };
 
 #ifdef ACCURATE_ENVELOPE
-const int statics[] = {
+const int32_t statics[] = {
   1764000, 1764000, 1411200, 1411200, 1190700, 1014300, 992250,
   882000, 705600, 705600, 584325, 507150, 502740, 441000, 418950,
   352800, 308700, 286650, 253575, 220500, 220500, 176400, 145530,
@@ -47,7 +47,7 @@ void Env::init_sr(double sampleRate) {
   sr_multiplier = (44100.0 / sampleRate) * (1 << 24);
 }
 
-void Env::init(const int r[4], const int l[4], int ol, int rate_scaling) {
+void Env::init(const int32_t r[4], const int32_t l[4], int32_t ol, int32_t rate_scaling) {
   for (int i = 0; i < 4; i++) {
     rates_[i] = r[i];
     levels_[i] = l[i];
@@ -75,7 +75,7 @@ int32_t Env::getsample() {
       ;
     }
     else if (rising_) {
-      const int jumptarget = 1716;
+      const int32_t jumptarget = 1716;
       if (level_ < (jumptarget << 16)) {
         level_ = jumptarget << 16;
       }
@@ -105,15 +105,15 @@ void Env::keydown(bool d) {
   }
 }
 
-int Env::scaleoutlevel(int outlevel) {
+int32_t Env::scaleoutlevel(int32_t outlevel) {
   return outlevel >= 20 ? 28 + outlevel : levellut[outlevel];
 }
 
 void Env::advance(int newix) {
   ix_ = newix;
   if (ix_ < 4) {
-    int newlevel = levels_[ix_];
-    int actuallevel = scaleoutlevel(newlevel) >> 1;
+    int32_t newlevel = levels_[ix_];
+    int32_t actuallevel = scaleoutlevel(newlevel) >> 1;
     actuallevel = (actuallevel << 6) + outlevel_ - 4256;
     actuallevel = actuallevel < 16 ? 16 : actuallevel;
     // level here is same as Java impl
@@ -121,18 +121,18 @@ void Env::advance(int newix) {
     rising_ = (targetlevel_ > level_);
 
     // rate
-    int qrate = (rates_[ix_] * 41) >> 6;
+    int32_t qrate = (rates_[ix_] * 41) >> 6;
     qrate += rate_scaling_;
-    qrate = min(qrate, 63);
+    qrate = std::min(int(qrate), 63);
 
 #ifdef ACCURATE_ENVELOPE
     if (targetlevel_ == level_ || (ix_ == 0 && newlevel == 0)) {
       // approximate number of samples at 44.100 kHz to achieve the time
       // empirically gathered using 2 TF1s, could probably use some double-checking
       // and cleanup, but it's pretty close for now.
-      int staticrate = rates_[ix_];
+      int32_t staticrate = rates_[ix_];
       staticrate += rate_scaling_; // needs to be checked, as well, but seems correct
-      staticrate = min(staticrate, 99);
+      staticrate = std::min(int(staticrate), 99);
       staticcount_ = staticrate < 77 ? statics[staticrate] : 20 * (99 - staticrate);
       if (staticrate < 77 && (ix_ == 0 && newlevel == 0)) {
         staticcount_ /= 20; // attack is scaled faster
@@ -149,7 +149,7 @@ void Env::advance(int newix) {
   }
 }
 
-void Env::update(const int r[4], const int l[4], int ol, int rate_scaling) {
+void Env::update(const int32_t r[4], const int32_t l[4], int32_t ol, int32_t rate_scaling) {
   for (int i = 0; i < 4; i++) {
     rates_[i] = r[i];
     levels_[i] = l[i];
@@ -158,8 +158,8 @@ void Env::update(const int r[4], const int l[4], int ol, int rate_scaling) {
   rate_scaling_ = rate_scaling;
   if ( down_ ) {
     // for now we simply reset ourselves at level 3
-    int newlevel = levels_[2];
-    int actuallevel = scaleoutlevel(newlevel) >> 1;
+    int32_t newlevel = levels_[2];
+    int32_t actuallevel = scaleoutlevel(newlevel) >> 1;
     actuallevel = (actuallevel << 6) - 4256;
     actuallevel = actuallevel < 16 ? 16 : actuallevel;
     targetlevel_ = actuallevel << 16;

third-party/Synth_Dexed/src/env.h

@@ -31,11 +31,11 @@ class Env {
     // (ie, value 0..99). The outlevel parameter is calibrated in microsteps
     // (ie units of approx .023 dB), with 99 * 32 = nominal full scale. The
     // rate_scaling parameter is in qRate units (ie 0..63).
-    void init(const int rates[4], const int levels[4], int outlevel,
-              int rate_scaling);
+    void init(const int32_t rates[4], const int32_t levels[4], int32_t outlevel,
+              int32_t rate_scaling);
 
-    void update(const int rates[4], const int levels[4], int outlevel,
-                int rate_scaling);
+    void update(const int32_t rates[4], const int32_t levels[4], int32_t outlevel,
+                int32_t rate_scaling);
     // Result is in Q24/doubling log format. Also, result is subsampled
     // for every N samples.
     // A couple more things need to happen for this to be used as a gain
@@ -45,7 +45,7 @@ class Env {
     int32_t getsample();
 
     void keydown(bool down);
-    static int scaleoutlevel(int outlevel);
+    static int32_t scaleoutlevel(int32_t outlevel);
     void getPosition(char *step);
 
     static void init_sr(double sample_rate);
@@ -57,20 +57,20 @@ class Env {
     // if we are not using 44100.
     static uint32_t sr_multiplier;
 
-    int rates_[4];
-    int levels_[4];
-    int outlevel_;
-    int rate_scaling_;
+    int32_t rates_[4];
+    int32_t levels_[4];
+    int32_t outlevel_;
+    int32_t rate_scaling_;
     // Level is stored so that 2^24 is one doubling, ie 16 more bits than
     // the DX7 itself (fraction is stored in level rather than separate
     // counter)
     int32_t level_;
-    int targetlevel_;
+    int32_t targetlevel_;
     bool rising_;
-    int ix_;
-    int inc_;
+    int32_t ix_;
+    int32_t inc_;
 #ifdef ACCURATE_ENVELOPE
-    int staticcount_;
+    int32_t staticcount_;
 #endif
 
     bool down_;

third-party/Synth_Dexed/src/exp2.h

@@ -34,13 +34,13 @@ extern int32_t exp2tab[EXP2_N_SAMPLES << 1];
 #ifdef EXP2_INLINE
 inline
 int32_t Exp2::lookup(int32_t x) {
-  const int SHIFT = 24 - EXP2_LG_N_SAMPLES;
-  int lowbits = x & ((1 << SHIFT) - 1);
-  int x_int = (x >> (SHIFT - 1)) & ((EXP2_N_SAMPLES - 1) << 1);
-  int dy = exp2tab[x_int];
-  int y0 = exp2tab[x_int + 1];
+  const int32_t SHIFT = 24 - EXP2_LG_N_SAMPLES;
+  int32_t lowbits = x & ((1 << SHIFT) - 1);
+  int32_t x_int = (x >> (SHIFT - 1)) & ((EXP2_N_SAMPLES - 1) << 1);
+  int32_t dy = exp2tab[x_int];
+  int32_t y0 = exp2tab[x_int + 1];
 
-  int y = y0 + (((int64_t)dy * (int64_t)lowbits) >> SHIFT);
+  int32_t y = y0 + (((int64_t)dy * (int64_t)lowbits) >> SHIFT);
   return y >> (6 - (x >> 24));
 }
 #endif
@@ -69,12 +69,12 @@ int32_t Tanh::lookup(int32_t x) {
     int32_t sx = ((int64_t) - 48408812 * (int64_t)x) >> 24;
     return signum ^ ((1 << 24) - 2 * Exp2::lookup(sx));
   } else {
-    const int SHIFT = 26 - TANH_LG_N_SAMPLES;
-    int lowbits = x & ((1 << SHIFT) - 1);
-    int x_int = (x >> (SHIFT - 1)) & ((TANH_N_SAMPLES - 1) << 1);
-    int dy = tanhtab[x_int];
-    int y0 = tanhtab[x_int + 1];
-    int y = y0 + (((int64_t)dy * (int64_t)lowbits) >> SHIFT);
+    const int32_t SHIFT = 26 - TANH_LG_N_SAMPLES;
+    int32_t lowbits = x & ((1 << SHIFT) - 1);
+    int32_t x_int = (x >> (SHIFT - 1)) & ((TANH_N_SAMPLES - 1) << 1);
+    int32_t dy = tanhtab[x_int];
+    int32_t y0 = tanhtab[x_int + 1];
+    int32_t y = y0 + (((int64_t)dy * (int64_t)lowbits) >> SHIFT);
     return y ^ signum;
   }
 }

third-party/Synth_Dexed/src/fm_core.cpp

@@ -57,7 +57,7 @@ const FmAlgorithm FmCore::algorithms[32] = {
 };
 
 int n_out(const FmAlgorithm &alg) {
-  int count = 0;
+  int32_t count = 0;
   for (int i = 0; i < 6; i++) {
     if ((alg.ops[i] & 7) == OUT_BUS_ADD) count++;
   }
@@ -84,7 +84,7 @@ void FmCore::dump() {
     cout << (i + 1) << ":";
     const FmAlgorithm &alg = algorithms[i];
     for (int j = 0; j < 6; j++) {
-      int flags = alg.ops[j];
+      int32_t flags = alg.ops[j];
       cout << " ";
       if (flags & FB_IN) cout << "[";
       cout << (flags & IN_BUS_ONE ? "1" : flags & IN_BUS_TWO ? "2" : "0") << "->";

third-party/Synth_Dexed/src/fm_core.h

@@ -25,8 +25,8 @@
 
 class FmOperatorInfo {
   public:
-    int in;
-    int out;
+    int32_t in;
+    int32_t out;
 };
 
 enum FmOperatorFlags {
@@ -41,7 +41,7 @@ enum FmOperatorFlags {
 
 class FmAlgorithm {
   public:
-    int ops[6];
+    int32_t ops[6];
 };
 
 class FmCore {

third-party/Synth_Dexed/src/fm_op_kernel.cpp

@@ -29,7 +29,7 @@ void neon_fm_kernel(const int32_t *in, const int32_t *busin, int32_t *out, int32
 
 const int32_t __attribute__ ((aligned(16))) const_0_1_2_3_arg[4] = {0, 1, 2, 3};
 const int32_t __attribute__ ((aligned(16))) mask23_arg = 0x7fffff;
-const float32_t __attribute__ ((aligned(16))) coeffs_arg[4] = {
+const float __attribute__ ((aligned(16))) coeffs_arg[4] = {
         -0.01880853017455781, 0.25215252666796095, -1.2333439964934032, 1.0
 };
 const int32_t __attribute__ ((aligned(16))) zeros[_N_] = {0};
@@ -214,7 +214,7 @@ void FmOpKernel::compute_pure(int32_t *output, int32_t phase0, int32_t freq,
 
 void FmOpKernel::compute_fb(int32_t *output, int32_t phase0, int32_t freq,
                             int32_t gain1, int32_t gain2,
-                            int32_t *fb_buf, int fb_shift, bool add) {
+                            int32_t *fb_buf, int32_t fb_shift, bool add) {
   int32_t dgain = (gain2 - gain1 + (_N_ >> 1)) >> LG_N;
   int32_t gain = gain1;
   int32_t phase = phase0;

third-party/Synth_Dexed/src/fm_op_kernel.h

@@ -41,7 +41,7 @@ class FmOpKernel {
     // One op with feedback, no add.
     static void compute_fb(int32_t *output, int32_t phase0, int32_t freq,
                            int32_t gain1, int32_t gain2,
-                           int32_t *fb_buf, int fb_gain, bool add);
+                           int32_t *fb_buf, int32_t fb_gain, bool add);
 };
 
 #endif

third-party/Synth_Dexed/src/freqlut.cpp

@@ -45,12 +45,12 @@ void Freqlut::init(FRAC_NUM sample_rate) {
 // Note: if logfreq is more than 20.0, the results will be inaccurate. However,
 // that will be many times the Nyquist rate.
 int32_t Freqlut::lookup(int32_t logfreq) {
-  int ix = (logfreq & 0xffffff) >> SAMPLE_SHIFT;
+  int32_t ix = (logfreq & 0xffffff) >> SAMPLE_SHIFT;
 
   int32_t y0 = lut[ix];
   int32_t y1 = lut[ix + 1];
-  int lowbits = logfreq & ((1 << SAMPLE_SHIFT) - 1);
+  int32_t lowbits = logfreq & ((1 << SAMPLE_SHIFT) - 1);
   int32_t y = y0 + ((((int64_t)(y1 - y0) * (int64_t)lowbits)) >> SAMPLE_SHIFT);
-  int hibits = logfreq >> 24;
+  int32_t hibits = logfreq >> 24;
   return y >> (MAX_LOGFREQ_INT - hibits);
 }

third-party/Synth_Dexed/src/lfo.cpp

@@ -16,6 +16,8 @@
 
 // Low frequency oscillator, compatible with DX7
 
+#include <algorithm>
+
 #include "synth.h"
 
 #include "sin.h"
@@ -29,11 +31,11 @@ void Lfo::init(FRAC_NUM sample_rate) {
 }
 
 void Lfo::reset(const uint8_t params[6]) {
-  int rate = params[0];  // 0..99
-  int sr = rate == 0 ? 1 : (165 * rate) >> 6;
+  int32_t rate = params[0];  // 0..99
+  int32_t sr = rate == 0 ? 1 : (165 * rate) >> 6;
   sr *= sr < 160 ? 11 : (11 + ((sr - 160) >> 4));
   delta_ = unit_ * sr;
-  int a = 99 - params[1];  // LFO delay
+  int32_t a = 99 - params[1];  // LFO delay
   if (a == 99) {
     delayinc_ = ~0u;
     delayinc2_ = ~0u;
@@ -41,7 +43,7 @@ void Lfo::reset(const uint8_t params[6]) {
     a = (16 + (a & 15)) << (1 + (a >> 4));
     delayinc_ = unit_ * a;
     a &= 0xff80;
-    a = max(0x80, a);
+    a = std::max(0x80, int(a));
     delayinc2_ = unit_ * a;
   }
   waveform_ = params[5];

third-party/Synth_Dexed/src/pitchenv.cpp

@@ -17,7 +17,7 @@
 #include "synth.h"
 #include "pitchenv.h"
 
-int PitchEnv::unit_;
+int32_t PitchEnv::unit_;
 
 void PitchEnv::init(FRAC_NUM sample_rate) {
   unit_ = _N_ * (1 << 24) / (21.3 * sample_rate) + 0.5;
@@ -42,7 +42,7 @@ const int8_t pitchenv_tab[] = {
   82, 92, 103, 115, 127
 };
 
-void PitchEnv::set(const int r[4], const int l[4]) {
+void PitchEnv::set(const int32_t r[4], const int32_t l[4]) {
   for (int i = 0; i < 4; i++) {
     rates_[i] = r[i];
     levels_[i] = l[i];
@@ -81,7 +81,7 @@ void PitchEnv::keydown(bool d) {
 void PitchEnv::advance(int newix) {
   ix_ = newix;
   if (ix_ < 4) {
-    int newlevel = levels_[ix_];
+    int32_t newlevel = levels_[ix_];
     targetlevel_ = pitchenv_tab[newlevel] << 19;
     rising_ = (targetlevel_ > level_);
     inc_ = pitchenv_rate[rates_[ix_]] * unit_;

third-party/Synth_Dexed/src/pitchenv.h

@@ -25,21 +25,21 @@ class PitchEnv {
 
     // The rates and levels arrays are calibrated to match the Dx7 parameters
     // (ie, value 0..99).
-    void set(const int rates[4], const int levels[4]);
+    void set(const int32_t rates[4], const int32_t levels[4]);
 
     // Result is in Q24/octave
     int32_t getsample();
     void keydown(bool down);
     void getPosition(char *step);
   private:
-    static int unit_;
-    int rates_[4];
-    int levels_[4];
+    static int32_t unit_;
+    int32_t rates_[4];
+    int32_t levels_[4];
     int32_t level_;
-    int targetlevel_;
+    int32_t targetlevel_;
     bool rising_;
-    int ix_;
-    int inc_;
+    int32_t ix_;
+    int32_t inc_;
 
     bool down_;
 

third-party/Synth_Dexed/src/porta.cpp

@@ -22,12 +22,12 @@ void Porta::init_sr(double sampleRate)
 {
   // compute portamento for CC 7-bit range
 
-  for (unsigned int i = 0; i < 128; ++i) {
+  for (uint8_t i = 0; i < 128; ++i) {
     // number of semitones travelled
     double sps = 350.0 * pow(2.0, -0.062 * i);  // per second
     double spf = sps / sampleRate;              // per frame
     double spp = spf * _N_;                       // per period
-    const int step = (1 << 24) / 12;
+    const int32_t step = (1 << 24) / 12;
     rates[i] = (int32_t)(0.5f + step * spp);    // to pitch units
   }
 }

third-party/Synth_Dexed/src/sin.cpp

@@ -60,18 +60,18 @@ void Sin::init() {
 
 #ifndef SIN_INLINE
 int32_t Sin::lookup(int32_t phase) {
-  const int SHIFT = 24 - SIN_LG_N_SAMPLES;
-  int lowbits = phase & ((1 << SHIFT) - 1);
+  const int32_t SHIFT = 24 - SIN_LG_N_SAMPLES;
+  int32_t lowbits = phase & ((1 << SHIFT) - 1);
 #ifdef SIN_DELTA
-  int phase_int = (phase >> (SHIFT - 1)) & ((SIN_N_SAMPLES - 1) << 1);
-  int dy = sintab[phase_int];
-  int y0 = sintab[phase_int + 1];
+  int32_t phase_int = (phase >> (SHIFT - 1)) & ((SIN_N_SAMPLES - 1) << 1);
+  int32_t dy = sintab[phase_int];
+  int32_t y0 = sintab[phase_int + 1];
 
   return y0 + (((int64_t)dy * (int64_t)lowbits) >> SHIFT);
 #else 
-  int phase_int = (phase >> SHIFT) & (SIN_N_SAMPLES - 1);
-  int y0 = sintab[phase_int];
-  int y1 = sintab[phase_int + 1];
+  int32_t phase_int = (phase >> SHIFT) & (SIN_N_SAMPLES - 1);
+  int32_t y0 = sintab[phase_int];
+  int32_t y1 = sintab[phase_int + 1];
 
   return y0 + (((int64_t)(y1 - y0) * (int64_t)lowbits) >> SHIFT);
 #endif

third-party/Synth_Dexed/src/sin.h

@@ -43,18 +43,18 @@ extern int32_t sintab[SIN_N_SAMPLES + 1];
 #ifdef SIN_INLINE
 inline
 int32_t Sin::lookup(int32_t phase) {
-  const int SHIFT = 24 - SIN_LG_N_SAMPLES;
-  int lowbits = phase & ((1 << SHIFT) - 1);
+  const int32_t SHIFT = 24 - SIN_LG_N_SAMPLES;
+  int32_t lowbits = phase & ((1 << SHIFT) - 1);
 #ifdef SIN_DELTA
-  int phase_int = (phase >> (SHIFT - 1)) & ((SIN_N_SAMPLES - 1) << 1);
-  int dy = sintab[phase_int];
-  int y0 = sintab[phase_int + 1];
+  int32_t phase_int = (phase >> (SHIFT - 1)) & ((SIN_N_SAMPLES - 1) << 1);
+  int32_t dy = sintab[phase_int];
+  int32_t y0 = sintab[phase_int + 1];
 
   return y0 + (((int64_t)dy * (int64_t)lowbits) >> SHIFT);
 #else
-  int phase_int = (phase >> SHIFT) & (SIN_N_SAMPLES - 1);
-  int y0 = sintab[phase_int];
-  int y1 = sintab[phase_int + 1];
+  int32_t phase_int = (phase >> SHIFT) & (SIN_N_SAMPLES - 1);
+  int32_t y0 = sintab[phase_int];
+  int32_t y1 = sintab[phase_int + 1];
 
   return y0 + (((int64_t)(y1 - y0) * (int64_t)lowbits) >> SHIFT);
 #endif

third-party/Synth_Dexed/src/synth.h

@@ -25,12 +25,10 @@
 #define TRANSPOSE_FIX 24
 #define VOICE_SILENCE_LEVEL 1100
 
-#define _MAX_NOTES 32
-
 #define LG_N 6
 #define _N_ (1 << LG_N)
 
-template<typename T>
+/*template<typename T>
 inline static T min(const T& a, const T& b) {
   return a < b ? a : b;
 }
@@ -38,7 +36,7 @@ inline static T min(const T& a, const T& b) {
 template<typename T>
 inline static T max(const T& a, const T& b) {
   return a > b ? a : b;
-}
+}*/
 
 #define QER(n,b) ( ((float)n)/(1<<b) )
 
@@ -67,7 +65,7 @@ inline static T max(const T& a, const T& b) {
   (_amt < _low) ? _low : ((_amt > _high) ? _high : _amt); \
 })
 
-static inline int32_t signed_saturate_rshift(int32_t val, int bits, int rshift)
+static inline int32_t signed_saturate_rshift(int32_t val, int32_t bits, int32_t rshift)
 {
   int32_t out, max;
 

third-party/Synth_Dexed/src/synth_dexed.h

@@ -37,8 +37,6 @@
 #define TRANSPOSE_FIX 24
 #define VOICE_SILENCE_LEVEL 1100
 
-#define _MAX_NOTES 32
-
 #define PB_RANGE_DEFAULT 1
 #define PB_STEP_DEFAULT 0
 #define MW_RANGE_DEFAULT 50

third-party/Synth_Dexed/synth.h

@@ -0,0 +1,91 @@
+/*
+   Copyright 2012 Google Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+        http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+*/
+
+#ifndef SYNTH_H
+#define SYNTH_H
+
+//#define SUPER_PRECISE
+
+#include <stdint.h>
+
+#define MIDI_CONTROLLER_MODE_MAX 2
+#define TRANSPOSE_FIX 24
+#define VOICE_SILENCE_LEVEL 1100
+
+#define LG_N 6
+#define _N_ (1 << LG_N)
+
+/*template<typename T>
+inline static T min(const T& a, const T& b) {
+  return a < b ? a : b;
+}
+
+template<typename T>
+inline static T max(const T& a, const T& b) {
+  return a > b ? a : b;
+}*/
+
+#define QER(n,b) ( ((float)n)/(1<<b) )
+
+#define FRAC_NUM float
+#define SIN_FUNC sinf
+// #define SIN_FUNC arm_sin_f32  // very fast but not as accurate
+#define COS_FUNC cosf
+// #define COS_FUNC arm_cos_f32  // very fast but not as accurate
+#define LOG_FUNC logf
+#define EXP_FUNC expf
+#define SQRT_FUNC sqrtf
+// #define ARM_SQRT_FUNC arm_sqrt_f32 // fast but not as accurate
+
+#if defined(__circle__)
+
+#include <circle/timer.h>
+
+#ifndef M_PI
+  #define M_PI 3.14159265358979323846
+#endif
+
+#define constrain(amt, low, high) ({ \
+  __typeof__(amt) _amt = (amt); \
+  __typeof__(low) _low = (low); \
+  __typeof__(high) _high = (high); \
+  (_amt < _low) ? _low : ((_amt > _high) ? _high : _amt); \
+})
+
+static inline int32_t signed_saturate_rshift(int32_t val, int32_t bits, int32_t rshift)
+{
+  int32_t out, max;
+
+  out = val >> rshift;
+  max = 1 << (bits - 1);
+  if (out >= 0)
+  {
+    if (out > max - 1) out = max - 1;
+  }
+  else
+  {
+    if (out < -max) out = -max;
+  }
+  return out;
+}
+
+static inline uint32_t millis (void)
+{
+        return uint32_t(CTimer::Get ()->GetClockTicks () / (CLOCKHZ / 1000));
+}
+
+#endif
+#endif

third-party/Synth_Dexed/synth_dexed.cpp

@@ -0,0 +1,63 @@
+/*
+   synth_dexed
+
+   synth_dexed is a port of the Dexed sound engine (https://github.com/asb2m10/dexed)
+   as library for the Teensy-3.5/3.6/4.x with an audio shield.
+   Dexed ist heavily based on https://github.com/google/music-synthesizer-for-android
+
+   (c)2018-2021 H. Wirtz <wirtz@parasitstudio.de>
+
+   This program is free software; you can redistribute it and/or modify
+   it under the terms of the GNU General Public License as published by
+   the Free Software Foundation; either version 3 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 General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software Foundation,
+   Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
+
+*/
+
+#include "synth_dexed.h"
+
+#if defined(TEENSYDUINO)
+void AudioSynthDexed::update(void)
+{
+  if (in_update == true)
+  {
+    xrun++;
+    return;
+  }
+  else
+    in_update = true;
+
+  elapsedMicros render_time;
+  audio_block_t *block;
+
+  block = allocate();
+
+  if (!block)
+  {
+    in_update = false;
+    return;
+  }
+
+  getSamples(block->data, AUDIO_BLOCK_SAMPLES);
+
+  if (render_time > audio_block_time_us) // everything greater audio_block_time_us (2.9ms for buffer size of 128) is a buffer underrun!
+    xrun++;
+
+  if (render_time > render_time_max)
+    render_time_max = render_time;
+
+  transmit(block, 0);
+  release(block);
+
+  in_update = false;
+};
+#endif

third-party/Synth_Dexed/synth_dexed.h

@@ -0,0 +1,72 @@
+/*
+   synth_dexed
+
+   synth_dexed is a port of the Dexed sound engine (https://github.com/asb2m10/dexed)
+   as library for the Teensy-3.5/3.6/4.x with an audio shield.
+   Dexed ist heavily based on https://github.com/google/music-synthesizer-for-android
+
+   (c)2018-2021 H. Wirtz <wirtz@parasitstudio.de>
+
+   This program is free software; you can redistribute it and/or modify
+   it under the terms of the GNU General Public License as published by
+   the Free Software Foundation; either version 3 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 General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software Foundation,
+   Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
+
+*/
+
+#pragma once
+#include "dexed.h"
+#if defined(TEENSYDUINO)
+#include <AudioStream.h>
+#endif
+#include <stdint.h>
+
+#define SYNTH_DEXED_VERSION "1.0.1"
+//#define DEBUG 1
+#define SAMPLE_RATE 44100
+
+#define TRANSPOSE_FIX 24
+#define VOICE_SILENCE_LEVEL 1100
+
+#define PB_RANGE_DEFAULT 1
+#define PB_STEP_DEFAULT 0
+#define MW_RANGE_DEFAULT 50
+#define MW_ASSIGN_DEFAULT 0 // Bitmapped: 0: Pitch, 1: Amp, 2: Bias
+#define MW_MODE_DEFAULT 0
+#define FC_RANGE_DEFAULT 50
+#define FC_ASSIGN_DEFAULT 0 // Bitmapped: 0: Pitch, 1: Amp, 2: Bias
+#define FC_MODE_DEFAULT 0
+#define BC_RANGE_DEFAULT 50
+#define BC_ASSIGN_DEFAULT 0 // Bitmapped: 0: Pitch, 1: Amp, 2: Bias
+#define BC_MODE_DEFAULT 0
+#define AT_RANGE_DEFAULT 50
+#define AT_ASSIGN_DEFAULT 0 // Bitmapped: 0: Pitch, 1: Amp, 2: Bias
+#define AT_MODE_DEFAULT 0
+#define PORTAMENTO_MODE_DEFAULT 0 // 0: Retain, 1: Follow
+#define PORTAMENTO_GLISSANDO_DEFAULT 0
+#define PORTAMENTO_TIME_DEFAULT 0
+
+//#define USE_SIMPLE_COMPRESSOR 1
+
+#if defined(TEENSYDUINO)
+class AudioSynthDexed : public AudioStream, public Dexed
+{
+  public:
+
+    AudioSynthDexed(uint8_t max_notes, uint16_t sample_rate) : AudioStream(0, NULL), Dexed(max_notes,sample_rate) { };
+
+  protected:
+    const uint16_t audio_block_time_us = 1000000 / (SAMPLE_RATE / AUDIO_BLOCK_SAMPLES);
+    volatile bool in_update = false;
+    void update(void);
+};
+#endif