1 /* reasonable simple synth
3 * Copyright (C) 2013 Robin Gareus <robin@gareus.org>
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; either version 2, or (at your option)
10 * This program is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 * GNU General Public License for more details.
15 * You should have received a copy of the GNU General Public License
16 * along with this program; if not, write to the Free Software Foundation,
17 * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
21 #define _GNU_SOURCE // needed for M_PI
30 #ifndef BUFFER_SIZE_SAMPLES
31 #define BUFFER_SIZE_SAMPLES 64
35 #define MIN(A, B) ( (A) < (B) ? (A) : (B) )
38 /* internal MIDI event abstraction */
47 struct rmidi_event_t {
48 enum RMIDI_EV_TYPE type;
49 uint8_t channel; /**< the MIDI channel number 0-15 */
63 uint32_t tme[3]; // attack, decay, release times [settings:ms || internal:samples]
64 float vol[2]; // attack, sustain volume [0..1]
65 uint32_t off[3]; // internal use (added attack,decay,release times)
68 typedef struct _RSSynthChannel {
70 uint32_t adsr_cnt[128];
72 float phase[128]; // various use, zero'ed on note-on
73 int8_t miditable[128]; // internal, note-on/off velocity
75 void (*synthesize) (struct _RSSynthChannel* sc,
76 const uint8_t note, const float vol, const float pc,
77 const size_t n_samples, float* left, float* right);
80 typedef void (*SynthFunction) (RSSynthChannel* sc,
81 const uint8_t note, const float vol, const float pc,
82 const size_t n_samples, float* left, float* right);
86 float buf [2][BUFFER_SIZE_SAMPLES];
87 RSSynthChannel sc[16];
97 /* initialize ADSR values
99 * @param rate sample-rate
100 * @param a attack time in seconds
101 * @param d decay time in seconds
102 * @param r release time in seconds
103 * @param avol attack gain [0..1]
104 * @param svol sustain volume level [0..1]
106 static void init_adsr(ADSRcfg *adsr, const double rate,
107 const uint32_t a, const uint32_t d, const uint32_t r,
108 const float avol, const float svol) {
112 adsr->tme[0] = a * rate / 1000.0;
113 adsr->tme[1] = d * rate / 1000.0;
114 adsr->tme[2] = r * rate / 1000.0;
116 assert(adsr->tme[0] > 32);
117 assert(adsr->tme[1] > 32);
118 assert(adsr->tme[2] > 32);
119 assert(adsr->vol[0] >=0 && adsr->vol[1] <= 1.0);
120 assert(adsr->vol[1] >=0 && adsr->vol[1] <= 1.0);
122 adsr->off[0] = adsr->tme[0];
123 adsr->off[1] = adsr->tme[1] + adsr->off[0];
124 adsr->off[2] = adsr->tme[2] + adsr->off[1];
127 /* calculate per-sample, per-key envelope */
128 static inline float adsr_env(RSSynthChannel *sc, const uint8_t note) {
130 if (sc->adsr_cnt[note] < sc->adsr.off[0]) {
132 const uint32_t p = ++sc->adsr_cnt[note];
133 if (p == sc->adsr.tme[0]) {
134 sc->adsr_amp[note] = sc->adsr.vol[0];
135 return sc->adsr.vol[0];
137 const float d = sc->adsr.vol[0] - sc->adsr_amp[note];
138 return sc->adsr_amp[note] + (p / (float) sc->adsr.tme[0]) * d;
141 else if (sc->adsr_cnt[note] < sc->adsr.off[1]) {
143 const uint32_t p = ++sc->adsr_cnt[note] - sc->adsr.off[0];
144 if (p == sc->adsr.tme[1]) {
145 sc->adsr_amp[note] = sc->adsr.vol[1];
146 return sc->adsr.vol[1];
148 const float d = sc->adsr.vol[1] - sc->adsr_amp[note];
149 return sc->adsr_amp[note] + (p / (float) sc->adsr.tme[1]) * d;
152 else if (sc->adsr_cnt[note] == sc->adsr.off[1]) {
154 return sc->adsr.vol[1];
156 else if (sc->adsr_cnt[note] < sc->adsr.off[2]) {
158 const uint32_t p = ++sc->adsr_cnt[note] - sc->adsr.off[1];
159 if (p == sc->adsr.tme[2]) {
160 sc->adsr_amp[note] = 0;
163 const float d = 0 - sc->adsr_amp[note];
164 return sc->adsr_amp[note] + (p / (float) sc->adsr.tme[2]) * d;
168 sc->adsr_cnt[note] = 0;
174 /*****************************************************************************/
175 /* piano like sound w/slight stereo phase */
176 static void synthesize_sineP (RSSynthChannel* sc,
177 const uint8_t note, const float vol, const float fq,
178 const size_t n_samples, float* left, float* right) {
181 float phase = sc->phase[note];
183 for (i=0; i < n_samples; ++i) {
184 float env = adsr_env(sc, note);
185 if (sc->adsr_cnt[note] == 0) break;
186 const float amp = vol * env;
188 left[i] += amp * sinf(2.0 * M_PI * phase);
189 left[i] += .300 * amp * sinf(2.0 * M_PI * phase * 2.0);
190 left[i] += .150 * amp * sinf(2.0 * M_PI * phase * 3.0);
191 left[i] += .080 * amp * sinf(2.0 * M_PI * phase * 4.0);
192 //left[i] -= .007 * amp * sinf(2.0 * M_PI * phase * 5.0);
193 //left[i] += .010 * amp * sinf(2.0 * M_PI * phase * 6.0);
194 left[i] += .020 * amp * sinf(2.0 * M_PI * phase * 7.0);
196 right[i] += amp * sinf(2.0 * M_PI * phase);
197 right[i] += .300 * amp * sinf(2.0 * M_PI * phase * 2.0);
198 right[i] += .150 * amp * sinf(2.0 * M_PI * phase * 3.0);
199 right[i] -= .080 * amp * sinf(2.0 * M_PI * phase * 4.0);
200 //right[i] += .007 * amp * sinf(2.0 * M_PI * phase * 5.0);
201 //right[i] += .010 * amp * sinf(2.0 * M_PI * phase * 6.0);
202 right[i] -= .020 * amp * sinf(2.0 * M_PI * phase * 7.0);
203 if (phase > 1.0) phase -= 2.0;
205 sc->phase[note] = phase;
208 static const ADSRcfg piano_adsr = {{ 5, 800, 100}, { 1.0, 0.0}, {0,0,0}};
210 /*****************************************************************************/
213 /* process note - move through ADSR states, count active keys,.. */
214 static void process_key (void *synth,
215 const uint8_t chn, const uint8_t note,
216 const size_t n_samples, float *left, float *right)
218 RSSynthesizer* rs = (RSSynthesizer*)synth;
219 RSSynthChannel* sc = &rs->sc[chn];
220 const int8_t vel = sc->miditable[note];
221 const float vol = /* master_volume */ 0.25 * fabsf(vel) / 127.0;
222 const float phase = sc->phase[note];
224 if (phase == -10 && vel > 0) {
226 assert(sc->adsr_cnt[note] == 0);
227 sc->adsr_amp[note] = 0;
228 sc->adsr_cnt[note] = 0;
231 //printf("[On] Now %d keys active on chn %d\n", sc->keycomp, chn);
233 else if (phase >= -1.0 && phase <= 1.0 && vel > 0) {
234 // sustain note or re-start note while adsr in progress:
235 if (sc->adsr_cnt[note] > sc->adsr.off[1]) {
237 sc->adsr_amp[note] = adsr_env(sc, note);
238 sc->adsr_cnt[note] = 0;
241 else if (phase >= -1.0 && phase <= 1.0 && vel < 0) {
243 if (sc->adsr_cnt[note] <= sc->adsr.off[1]) {
244 if (sc->adsr_cnt[note] != sc->adsr.off[1]) {
246 sc->adsr_amp[note] = adsr_env(sc, note);
248 sc->adsr_cnt[note] = sc->adsr.off[1] + 1;
252 /* note-on + off in same cycle */
253 sc->miditable[note] = 0;
254 sc->adsr_cnt[note] = 0;
255 sc->phase[note] = -10;
259 // synthesize actual sound
260 sc->synthesize(sc, note, vol, rs->freqs[note], n_samples, left, right);
262 if (sc->adsr_cnt[note] == 0) {
263 //printf("Note %d,%d released\n", chn, note);
264 sc->miditable[note] = 0;
265 sc->adsr_amp[note] = 0;
266 sc->phase[note] = -10;
268 //printf("[off] Now %d keys active on chn %d\n", sc->keycomp, chn);
272 /* synthesize a BUFFER_SIZE_SAMPLES's of audio-data */
273 static void synth_fragment (void *synth, const size_t n_samples, float *left, float *right) {
274 RSSynthesizer* rs = (RSSynthesizer*)synth;
275 memset (left, 0, n_samples * sizeof(float));
276 memset (right, 0, n_samples * sizeof(float));
281 for (c=0; c < 16; ++c) {
282 for (k=0; k < 128; ++k) {
283 if (rs->sc[c].miditable[k] == 0) continue;
284 process_key(synth, c, k, n_samples, left, right);
286 keycomp += rs->sc[c].keycomp;
289 #if 1 // key-compression
290 float kctgt = 8.0 / (float)(keycomp + 7.0);
291 if (kctgt < .5) kctgt = .5;
292 if (kctgt > 1.0) kctgt = 1.0;
293 const float _w = rs->kcfilt;
294 for (i=0; i < n_samples; ++i) {
295 rs->kcgain += _w * (kctgt - rs->kcgain);
296 left[i] *= rs->kcgain;
297 right[i] *= rs->kcgain;
303 static void synth_reset_channel(RSSynthChannel* sc) {
305 for (k=0; k < 128; ++k) {
309 sc->miditable[k] = 0;
314 static void synth_reset(void *synth) {
315 RSSynthesizer* rs = (RSSynthesizer*)synth;
317 for (c=0; c < 16; ++c) {
318 synth_reset_channel(&(rs->sc[c]));
323 static void synth_load(RSSynthChannel *sc, const double rate,
324 SynthFunction synthesize,
325 ADSRcfg const * const adsr) {
326 synth_reset_channel(sc);
327 init_adsr(&sc->adsr, rate,
328 adsr->tme[0], adsr->tme[1], adsr->tme[2],
329 adsr->vol[0], adsr->vol[1]);
330 sc->synthesize = synthesize;
335 * internal abstraction of MIDI data handling
337 static void synth_process_midi_event(void *synth, struct rmidi_event_t *ev) {
338 RSSynthesizer* rs = (RSSynthesizer*)synth;
341 if (rs->sc[ev->channel].miditable[ev->d.tone.note] <= 0)
342 rs->sc[ev->channel].miditable[ev->d.tone.note] = ev->d.tone.velocity;
345 if (rs->sc[ev->channel].miditable[ev->d.tone.note] > 0)
346 rs->sc[ev->channel].miditable[ev->d.tone.note] *= -1.0;
351 if (ev->d.control.param == 0x00 || ev->d.control.param == 0x20) {
352 /* 0x00 and 0x20 are used for BANK select */
355 if (ev->d.control.param == 121) {
356 /* reset all controllers */
359 if (ev->d.control.param == 120 || ev->d.control.param == 123) {
360 /* Midi panic: 120: all sound off, 123: all notes off*/
361 synth_reset_channel(&(rs->sc[ev->channel]));
364 if (ev->d.control.param >= 120) {
365 /* params 122-127 are reserved - skip them. */
374 /******************************************************************************
375 * PUBLIC API (used by lv2.c)
379 * align LV2 and internal synth buffers
380 * call synth_fragment as often as needed for the given LV2 buffer size
382 * @param synth synth-handle
383 * @param written samples written so far (offset in \ref out)
384 * @param nframes total samples to synthesize and write to the \out buffer
385 * @param out pointer to stereo output buffers
386 * @return end of buffer (written + nframes)
388 static uint32_t synth_sound (void *synth, uint32_t written, const uint32_t nframes, float **out) {
389 RSSynthesizer* rs = (RSSynthesizer*)synth;
391 while (written < nframes) {
392 uint32_t nremain = nframes - written;
394 if (rs->boffset >= BUFFER_SIZE_SAMPLES) {
396 synth_fragment(rs, BUFFER_SIZE_SAMPLES, rs->buf[0], rs->buf[1]);
399 uint32_t nread = MIN(nremain, (BUFFER_SIZE_SAMPLES - rs->boffset));
401 memcpy(&out[0][written], &rs->buf[0][rs->boffset], nread*sizeof(float));
402 memcpy(&out[1][written], &rs->buf[1][rs->boffset], nread*sizeof(float));
405 rs->boffset += nread;
411 * parse raw midi-data.
413 * @param synth synth-handle
414 * @param data 8bit midi message
415 * @param size number of bytes in the midi-message
417 static void synth_parse_midi(void *synth, const uint8_t *data, const size_t size) {
418 if (size < 2 || size > 3) return;
419 // All messages need to be 3 bytes; except program-changes: 2bytes.
420 if (size == 2 && (data[0] & 0xf0) != 0xC0) return;
422 struct rmidi_event_t ev;
424 ev.channel = data[0]&0x0f;
425 switch (data[0] & 0xf0) {
428 ev.d.tone.note=data[1]&0x7f;
429 ev.d.tone.velocity=data[2]&0x7f;
433 ev.d.tone.note=data[1]&0x7f;
434 ev.d.tone.velocity=data[2]&0x7f;
437 ev.type=CONTROL_CHANGE;
438 ev.d.control.param=data[1]&0x7f;
439 ev.d.control.value=data[2]&0x7f;
442 ev.type=PROGRAM_CHANGE;
443 ev.d.control.value=data[1]&0x7f;
448 synth_process_midi_event(synth, &ev);
451 static const uint8_t jingle[] = { 71 ,71 ,71 ,71 ,71 ,71 ,71 ,74 ,67 ,69 ,71 ,72 ,72 ,72 ,72 ,72 ,71 ,71 ,71 ,71 ,71 ,69 ,69 ,71 ,69 ,74 ,71 ,71 ,71 ,71 ,71 ,71 ,71 ,74 ,67 ,69 ,71 ,72 ,72 ,72 ,72 ,72 ,71 ,71 ,71 ,71 ,74 ,74 ,72 ,69 ,67 ,62 ,62 ,71 ,69 ,67 ,62 ,62 ,62 ,62 ,71 ,69 ,67 ,64 ,64 ,64 ,72 ,71 ,69 ,66 ,74 ,76 ,74 ,72 ,69 ,71 ,62 ,62 ,71 ,69 ,67 ,62 ,62 ,62 ,62 ,71 ,69 ,67 ,64 ,64 ,64 ,72 ,71 ,69 ,74 ,74 ,74 ,74 ,76 ,74 ,72 ,69 ,67 ,74 ,71 ,71 ,71 ,71 ,71 ,71 ,71 ,74 ,67 ,69 ,71 ,72 ,72 ,72 ,72 ,72 ,71 ,71 ,71 ,71 ,71 ,69 ,69 ,71 ,69 ,74 ,71 ,71 ,71 ,71 ,71 ,71 ,71 ,74 ,67 ,69 ,71 ,72 ,72 ,72 ,72 ,72 ,71 ,71 ,71 ,71 ,74 ,74 ,72 ,69 ,67 };
453 static void synth_parse_xmas(void *synth, const uint8_t *data, const size_t size) {
454 RSSynthesizer* rs = (RSSynthesizer*)synth;
455 if (size < 2 || size > 3) return;
456 // All messages need to be 3 bytes; except program-changes: 2bytes.
457 if (size == 2 && (data[0] & 0xf0) != 0xC0) return;
459 struct rmidi_event_t ev;
461 ev.channel = data[0]&0x0f;
462 switch (data[0] & 0xf0) {
465 ev.d.tone.note=jingle[rs->xmas_off++];
466 ev.d.tone.velocity=data[2]&0x7f;
467 if (rs->xmas_off >= sizeof(jingle)) rs->xmas_off = 0;
471 ev.d.tone.note=jingle[rs->xmas_on++];
472 ev.d.tone.velocity=data[2]&0x7f;
473 if (rs->xmas_on >= sizeof(jingle)) rs->xmas_on = 0;
476 ev.type=CONTROL_CHANGE;
477 ev.d.control.param=data[1]&0x7f;
478 ev.d.control.value=data[2]&0x7f;
481 ev.type=PROGRAM_CHANGE;
482 ev.d.control.value=data[1]&0x7f;
487 synth_process_midi_event(synth, &ev);
490 * initialize the synth
491 * This should be called after synth_alloc()
492 * as soon as the sample-rate is known
494 * @param synth synth-handle
495 * @param rate sample-rate
497 static void synth_init(void *synth, double rate) {
498 RSSynthesizer* rs = (RSSynthesizer*)synth;
500 rs->boffset = BUFFER_SIZE_SAMPLES;
501 const float tuning = 440;
503 for (k=0; k < 128; k++) {
504 rs->freqs[k] = (2.0 * tuning / 32.0f) * powf(2, (k - 9.0) / 12.0) / rate;
505 assert(rs->freqs[k] < M_PI/2); // otherwise spatialization may phase out..
507 rs->kcfilt = 12.0 / rate;
510 for (c=0; c < 16; c++) {
511 synth_load(&rs->sc[c], rate, &synthesize_sineP, &piano_adsr);
518 * Allocate data-structure, create a handle for all other synth_* functions.
520 * This data should be freeded with \ref synth_free when the synth is no
523 * The synth can only be used after calling \rev synth_init as well.
525 * @return synth-handle
527 static void * synth_alloc(void) {
528 return calloc(1, sizeof(RSSynthesizer));
532 * release synth data structure
533 * @param synth synth-handle
535 static void synth_free(void *synth) {
538 /* vi:set ts=8 sts=2 sw=2 et: */