2 Calf Box, an open source musical instrument.
3 Copyright (C) 2010-2013 Krzysztof Foltman
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 3 of the License, or
8 (at your option) any later version.
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, see <http://www.gnu.org/licenses/>.
20 #include "config-api.h"
27 #include "sampler_impl.h"
28 #include "sfzloader.h"
40 static void lfo_init(struct sampler_lfo
*lfo
, struct sampler_lfo_params
*lfop
, int srate
, double srate_inv
)
44 lfo
->delta
= (uint32_t)(lfop
->freq
* 65536.0 * 65536.0 * CBOX_BLOCK_SIZE
* srate_inv
);
45 lfo
->delay
= (uint32_t)(lfop
->delay
* srate
);
46 lfo
->fade
= (uint32_t)(lfop
->fade
* srate
);
49 static inline float lfo_run(struct sampler_lfo
*lfo
)
51 if (lfo
->age
< lfo
->delay
)
53 lfo
->age
+= CBOX_BLOCK_SIZE
;
57 const int FRAC_BITS
= 32 - 11;
58 lfo
->phase
+= lfo
->delta
;
59 uint32_t iphase
= lfo
->phase
>> FRAC_BITS
;
60 float frac
= (lfo
->phase
& ((1 << FRAC_BITS
) - 1)) * (1.0 / (1 << FRAC_BITS
));
62 float v
= sampler_sine_wave
[iphase
] + (sampler_sine_wave
[iphase
+ 1] - sampler_sine_wave
[iphase
]) * frac
;
63 if (lfo
->fade
&& lfo
->age
< lfo
->delay
+ lfo
->fade
)
65 v
*= (lfo
->age
- lfo
->delay
) * 1.0 / lfo
->fade
;
66 lfo
->age
+= CBOX_BLOCK_SIZE
;
72 static gboolean
is_tail_finished(struct sampler_voice
*v
)
74 if (v
->layer
->cutoff
== -1)
76 double eps
= 1.0 / 65536.0;
77 if (cbox_biquadf_is_audible(&v
->filter_left
, eps
))
79 if (cbox_biquadf_is_audible(&v
->filter_right
, eps
))
81 if (sampler_layer_data_is_4pole(v
->layer
))
83 if (cbox_biquadf_is_audible(&v
->filter_left2
, eps
))
85 if (cbox_biquadf_is_audible(&v
->filter_right2
, eps
))
96 static inline void mix_block_into_with_gain(cbox_sample_t
**outputs
, int oofs
, float *src_left
, float *src_right
, float gain
)
98 float *dst_left
= outputs
[oofs
];
99 float *dst_right
= outputs
[oofs
+ 1];
100 float32x2_t gain2
= {gain
, gain
};
101 for (size_t i
= 0; i
< CBOX_BLOCK_SIZE
; i
+= 4)
103 float32x2_t l1
= vld1_f32(&src_left
[i
]);
104 float32x2_t l2
= vld1_f32(&src_left
[i
+ 2]);
105 float32x2_t r1
= vld1_f32(&src_right
[i
]);
106 float32x2_t r2
= vld1_f32(&src_right
[i
+ 2]);
107 float32x2_t dl1
= vld1_f32(&dst_left
[i
]);
108 float32x2_t dl2
= vld1_f32(&dst_left
[i
+ 2]);
109 float32x2_t dr1
= vld1_f32(&dst_right
[i
]);
110 float32x2_t dr2
= vld1_f32(&dst_right
[i
+ 2]);
112 l1
= vmla_f32(dl1
, l1
, gain2
);
113 l2
= vmla_f32(dl2
, l2
, gain2
);
114 vst1_f32(&dst_left
[i
], l1
);
115 vst1_f32(&dst_left
[i
+ 2], l2
);
116 r1
= vmla_f32(dr1
, r1
, gain2
);
117 r2
= vmla_f32(dr2
, r2
, gain2
);
118 vst1_f32(&dst_right
[i
], r1
);
119 vst1_f32(&dst_right
[i
+ 2], r2
);
123 static inline void mix_block_into(cbox_sample_t
**outputs
, int oofs
, float *src_left
, float *src_right
)
125 float *dst_left
= outputs
[oofs
];
126 float *dst_right
= outputs
[oofs
+ 1];
127 for (size_t i
= 0; i
< CBOX_BLOCK_SIZE
; i
+= 4)
129 float32x2_t l1
= vld1_f32(&src_left
[i
]);
130 float32x2_t l2
= vld1_f32(&src_left
[i
+ 2]);
131 float32x2_t r1
= vld1_f32(&src_right
[i
]);
132 float32x2_t r2
= vld1_f32(&src_right
[i
+ 2]);
133 float32x2_t dl1
= vld1_f32(&dst_left
[i
]);
134 float32x2_t dl2
= vld1_f32(&dst_left
[i
+ 2]);
135 float32x2_t dr1
= vld1_f32(&dst_right
[i
]);
136 float32x2_t dr2
= vld1_f32(&dst_right
[i
+ 2]);
138 l1
= vadd_f32(dl1
, l1
);
139 l2
= vadd_f32(dl2
, l2
);
140 vst1_f32(&dst_left
[i
], l1
);
141 vst1_f32(&dst_left
[i
+ 2], l2
);
142 r1
= vadd_f32(dr1
, r1
);
143 r2
= vadd_f32(dr2
, r2
);
144 vst1_f32(&dst_right
[i
], r1
);
145 vst1_f32(&dst_right
[i
+ 2], r2
);
151 static inline void mix_block_into_with_gain(cbox_sample_t
**outputs
, int oofs
, float *src_left
, float *src_right
, float gain
)
153 cbox_sample_t
*dst_left
= outputs
[oofs
];
154 cbox_sample_t
*dst_right
= outputs
[oofs
+ 1];
155 for (size_t i
= 0; i
< CBOX_BLOCK_SIZE
; i
++)
157 dst_left
[i
] += gain
* src_left
[i
];
158 dst_right
[i
] += gain
* src_right
[i
];
162 static inline void mix_block_into(cbox_sample_t
**outputs
, int oofs
, float *src_left
, float *src_right
)
164 cbox_sample_t
*dst_left
= outputs
[oofs
];
165 cbox_sample_t
*dst_right
= outputs
[oofs
+ 1];
166 for (size_t i
= 0; i
< CBOX_BLOCK_SIZE
; i
++)
168 dst_left
[i
] += src_left
[i
];
169 dst_right
[i
] += src_right
[i
];
175 ////////////////////////////////////////////////////////////////////////////////
177 void sampler_voice_activate(struct sampler_voice
*v
, enum sampler_player_type mode
)
179 assert(v
->gen
.mode
== spt_inactive
);
180 sampler_voice_unlink(&v
->program
->module
->voices_free
, v
);
181 assert(mode
!= spt_inactive
);
184 sampler_voice_link(&v
->channel
->voices_running
, v
);
187 void sampler_voice_start(struct sampler_voice
*v
, struct sampler_channel
*c
, struct sampler_layer_data
*l
, int note
, int vel
, int *exgroups
, int *pexgroupcount
)
189 struct sampler_module
*m
= c
->module
;
190 sampler_gen_reset(&v
->gen
);
193 if (l
->trigger
== stm_release
)
195 // time since last 'note on' for that note
196 v
->age
= m
->current_time
- c
->prev_note_start_time
[note
];
197 double age
= v
->age
* m
->module
.srate_inv
;
198 // if attenuation is more than 84dB, ignore the release trigger
199 if (age
* l
->rt_decay
> 84)
202 uint32_t end
= l
->eff_waveform
->info
.frames
;
204 end
= (l
->end
== -1) ? 0 : l
->end
;
205 v
->last_waveform
= l
->eff_waveform
;
206 v
->gen
.cur_sample_end
= end
;
207 if (end
> l
->eff_waveform
->info
.frames
)
208 end
= l
->eff_waveform
->info
.frames
;
210 assert(!v
->current_pipe
);
211 if (end
> l
->eff_waveform
->preloaded_frames
)
213 if (l
->loop_mode
== slm_loop_continuous
&& l
->loop_end
< l
->eff_waveform
->preloaded_frames
)
215 // Everything fits in prefetch, because loop ends in prefetch and post-loop part is not being played
219 uint32_t loop_start
= -1, loop_end
= end
;
220 // If in loop mode, set the loop over the looped part... unless we're doing sustain-only loop on prefetch area only. Then
221 // streaming will only cover the release part, and it shouldn't be looped.
222 if (l
->loop_mode
== slm_loop_continuous
|| (l
->loop_mode
== slm_loop_sustain
&& l
->loop_end
>= l
->eff_waveform
->preloaded_frames
))
224 loop_start
= l
->loop_start
;
225 loop_end
= l
->loop_end
;
227 // Those are initial values only, they will be adjusted in process function
228 v
->current_pipe
= cbox_prefetch_stack_pop(m
->pipe_stack
, l
->eff_waveform
, loop_start
, loop_end
, l
->count
);
229 if (!v
->current_pipe
)
230 g_warning("Prefetch pipe pool exhausted, no streaming playback will be possible");
234 v
->output_pair_no
= l
->output
% m
->output_pairs
;
235 v
->serial_no
= m
->serial_no
;
237 uint32_t pos
= l
->offset
;
239 if (l
->offset_random
)
240 pos
+= ((uint32_t)(rand() + (rand() << 16))) % l
->offset_random
;
243 v
->gen
.bigpos
= ((uint64_t)pos
) << 32;
245 float delay
= l
->delay
;
247 delay
+= rand() * (1.0 / RAND_MAX
) * l
->delay_random
;
249 v
->delay
= (int)(delay
* m
->module
.srate
);
252 v
->gen
.loop_overlap
= l
->loop_overlap
;
253 v
->gen
.loop_overlap_step
= l
->loop_overlap
> 0 ? 1.0 / l
->loop_overlap
: 0;
254 v
->gain_fromvel
= 1.0 + (l
->eff_velcurve
[vel
] - 1.0) * l
->amp_veltrack
* 0.01;
260 v
->released_with_sustain
= 0;
261 v
->released_with_sostenuto
= 0;
262 v
->captured_sostenuto
= 0;
265 v
->program
= c
->program
;
266 v
->amp_env
.shape
= &l
->amp_env_shape
;
267 v
->filter_env
.shape
= &l
->filter_env_shape
;
268 v
->pitch_env
.shape
= &l
->pitch_env_shape
;
270 v
->cutoff_shift
= vel
* l
->fil_veltrack
/ 127.0 + (note
- l
->fil_keycenter
) * l
->fil_keytrack
;
271 v
->loop_mode
= l
->loop_mode
;
272 v
->off_by
= l
->off_by
;
273 int auxes
= (m
->module
.outputs
- m
->module
.aux_offset
) / 2;
274 if (l
->effect1bus
>= 1 && l
->effect1bus
< 1 + auxes
)
275 v
->send1bus
= l
->effect1bus
;
278 if (l
->effect2bus
>= 1 && l
->effect2bus
< 1 + auxes
)
279 v
->send2bus
= l
->effect2bus
;
282 v
->send1gain
= l
->effect1
* 0.01;
283 v
->send2gain
= l
->effect2
* 0.01;
284 if (l
->group
>= 1 && *pexgroupcount
< MAX_RELEASED_GROUPS
)
286 gboolean found
= FALSE
;
287 for (int j
= 0; j
< *pexgroupcount
; j
++)
289 if (exgroups
[j
] == l
->group
)
297 exgroups
[(*pexgroupcount
)++] = l
->group
;
300 lfo_init(&v
->amp_lfo
, &l
->amp_lfo
, m
->module
.srate
, m
->module
.srate_inv
);
301 lfo_init(&v
->filter_lfo
, &l
->filter_lfo
, m
->module
.srate
, m
->module
.srate_inv
);
302 lfo_init(&v
->pitch_lfo
, &l
->pitch_lfo
, m
->module
.srate
, m
->module
.srate_inv
);
304 cbox_biquadf_reset(&v
->filter_left
);
305 cbox_biquadf_reset(&v
->filter_right
);
306 cbox_biquadf_reset(&v
->filter_left2
);
307 cbox_biquadf_reset(&v
->filter_right2
);
309 GSList
*nif
= v
->layer
->nifs
;
312 struct sampler_noteinitfunc
*p
= nif
->data
;
317 cbox_envelope_reset(&v
->amp_env
);
318 cbox_envelope_reset(&v
->filter_env
);
319 cbox_envelope_reset(&v
->pitch_env
);
321 v
->last_eq_bitmask
= 0;
323 sampler_voice_activate(v
, l
->eff_waveform
->info
.channels
== 2 ? spt_stereo16
: spt_mono16
);
325 if (v
->current_pipe
&& v
->gen
.bigpos
)
326 cbox_prefetch_pipe_consumed(v
->current_pipe
, v
->gen
.bigpos
>> 32);
327 v
->layer_changed
= TRUE
;
330 void sampler_voice_link(struct sampler_voice
**pv
, struct sampler_voice
*v
)
339 void sampler_voice_unlink(struct sampler_voice
**pv
, struct sampler_voice
*v
)
344 v
->prev
->next
= v
->next
;
346 v
->next
->prev
= v
->prev
;
351 void sampler_voice_inactivate(struct sampler_voice
*v
, gboolean expect_active
)
353 assert((v
->gen
.mode
!= spt_inactive
) == expect_active
);
354 sampler_voice_unlink(&v
->channel
->voices_running
, v
);
355 v
->gen
.mode
= spt_inactive
;
358 cbox_prefetch_stack_push(v
->program
->module
->pipe_stack
, v
->current_pipe
);
359 v
->current_pipe
= NULL
;
362 sampler_voice_link(&v
->program
->module
->voices_free
, v
);
365 void sampler_voice_release(struct sampler_voice
*v
, gboolean is_polyaft
)
367 if ((v
->loop_mode
== slm_one_shot_chokeable
) != is_polyaft
)
369 if (v
->delay
>= v
->age
+ CBOX_BLOCK_SIZE
)
372 sampler_voice_inactivate(v
, TRUE
);
376 if (v
->loop_mode
!= slm_one_shot
&& !v
->layer
->count
)
379 if (v
->loop_mode
== slm_loop_sustain
&& v
->current_pipe
)
382 v
->current_pipe
->file_loop_end
= v
->gen
.cur_sample_end
;
383 v
->current_pipe
->file_loop_start
= -1;
389 void sampler_voice_process(struct sampler_voice
*v
, struct sampler_module
*m
, cbox_sample_t
**outputs
)
391 struct sampler_layer_data
*l
= v
->layer
;
392 assert(v
->gen
.mode
!= spt_inactive
);
394 // if it's a DAHD envelope without sustain, consider the note finished
395 if (v
->amp_env
.cur_stage
== 4 && v
->amp_env
.shape
->stages
[3].end_value
<= 0.f
)
396 cbox_envelope_go_to(&v
->amp_env
, 15);
398 struct sampler_channel
*c
= v
->channel
;
399 v
->age
+= CBOX_BLOCK_SIZE
;
401 if (v
->age
< v
->delay
)
404 // XXXKF I'm sacrificing sample accuracy for delays for now
406 const float velscl
= v
->vel
* (1.f
/ 127.f
);
407 if (v
->layer_changed
)
410 if (v
->last_waveform
!= v
->layer
->eff_waveform
)
412 v
->last_waveform
= v
->layer
->eff_waveform
;
413 if (v
->layer
->eff_waveform
)
415 v
->gen
.mode
= v
->layer
->eff_waveform
->info
.channels
== 2 ? spt_stereo16
: spt_mono16
;
416 v
->gen
.cur_sample_end
= v
->layer
->eff_waveform
->info
.frames
;
420 sampler_voice_inactivate(v
, TRUE
);
424 #define RECALC_EQ_IF(index) \
425 if (l->eq_bitmask & (1 << (index - 1))) \
427 cbox_biquadf_set_peakeq_rbj_scaled(&v->eq_coeffs[index - 1], l->eq##index.effective_freq + velscl * l->eq##index.vel2freq, 1.0 / l->eq##index.bw, dB2gain(0.5 * (l->eq##index.gain + velscl * l->eq##index.vel2gain)), m->module.srate); \
428 if (!(v->last_eq_bitmask & (1 << (index - 1)))) \
430 cbox_biquadf_reset(&v->eq_left[index-1]); \
431 cbox_biquadf_reset(&v->eq_right[index-1]); \
438 v
->last_eq_bitmask
= l
->eq_bitmask
;
439 v
->layer_changed
= FALSE
;
442 float pitch
= (v
->note
- l
->pitch_keycenter
) * l
->pitch_keytrack
+ l
->tune
+ l
->transpose
* 100 + v
->pitch_shift
;
443 float modsrcs
[smsrc_pernote_count
];
444 modsrcs
[smsrc_vel
- smsrc_pernote_offset
] = v
->vel
* velscl
;
445 modsrcs
[smsrc_pitch
- smsrc_pernote_offset
] = pitch
* (1.f
/ 100.f
);
446 modsrcs
[smsrc_polyaft
- smsrc_pernote_offset
] = 0.f
; // XXXKF not supported yet
447 modsrcs
[smsrc_pitchenv
- smsrc_pernote_offset
] = cbox_envelope_get_next(&v
->pitch_env
, v
->released
) * 0.01f
;
448 modsrcs
[smsrc_filenv
- smsrc_pernote_offset
] = cbox_envelope_get_next(&v
->filter_env
, v
->released
) * 0.01f
;
449 modsrcs
[smsrc_ampenv
- smsrc_pernote_offset
] = cbox_envelope_get_next(&v
->amp_env
, v
->released
) * 0.01f
;
451 modsrcs
[smsrc_amplfo
- smsrc_pernote_offset
] = lfo_run(&v
->amp_lfo
);
452 modsrcs
[smsrc_fillfo
- smsrc_pernote_offset
] = lfo_run(&v
->filter_lfo
);
453 modsrcs
[smsrc_pitchlfo
- smsrc_pernote_offset
] = lfo_run(&v
->pitch_lfo
);
455 if (v
->amp_env
.cur_stage
< 0)
457 if (is_tail_finished(v
))
459 sampler_voice_inactivate(v
, TRUE
);
464 float moddests
[smdestcount
];
465 moddests
[smdest_gain
] = 0;
466 moddests
[smdest_pitch
] = pitch
;
467 moddests
[smdest_cutoff
] = v
->cutoff_shift
;
468 moddests
[smdest_resonance
] = 0;
469 GSList
*mod
= l
->modulations
;
470 if (l
->trigger
== stm_release
)
471 moddests
[smdest_gain
] -= v
->age
* l
->rt_decay
* m
->module
.srate_inv
;
474 moddests
[smdest_pitch
] += c
->pitchwheel
* (c
->pitchwheel
> 0 ? l
->bend_up
: l
->bend_down
) >> 13;
476 static const int modoffset
[4] = {0, -1, -1, 1 };
477 static const int modscale
[4] = {1, 1, 2, -2 };
480 struct sampler_modulation
*sm
= mod
->data
;
481 float value
= 0.f
, value2
= 1.f
;
482 if (sm
->src
< smsrc_pernote_offset
)
483 value
= c
->cc
[sm
->src
] * (1.f
/ 127.f
);
485 value
= modsrcs
[sm
->src
- smsrc_pernote_offset
];
486 value
= modoffset
[sm
->flags
& 3] + value
* modscale
[sm
->flags
& 3];
488 if (sm
->src2
!= smsrc_none
)
490 if (sm
->src2
< smsrc_pernote_offset
)
491 value2
= c
->cc
[sm
->src2
] * (1.f
/ 127.f
);
493 value2
= modsrcs
[sm
->src2
- smsrc_pernote_offset
];
495 value2
= modoffset
[(sm
->flags
& 12) >> 2] + value2
* modscale
[(sm
->flags
& 12) >> 2];
498 moddests
[sm
->dest
] += value
* sm
->amount
;
500 mod
= g_slist_next(mod
);
503 double maxv
= 127 << 7;
504 double freq
= l
->eff_freq
* cent2factor(moddests
[smdest_pitch
]) ;
505 uint64_t freq64
= (uint64_t)(freq
* 65536.0 * 65536.0 * m
->module
.srate_inv
);
507 gboolean playing_sustain_loop
= !v
->released
&& v
->loop_mode
== slm_loop_sustain
;
508 uint32_t loop_start
, loop_end
;
509 gboolean bandlimited
= FALSE
;
511 if (!v
->current_pipe
)
513 v
->gen
.sample_data
= v
->last_waveform
->data
;
514 if (v
->last_waveform
->levels
)
516 if (v
->last_level
> 0 && v
->last_level
< v
->last_waveform
->level_count
517 && freq64
> v
->last_level_min_rate
&& freq64
<= v
->last_waveform
->levels
[v
->last_level
].max_rate
)
519 v
->gen
.sample_data
= v
->last_waveform
->levels
[v
->last_level
].data
;
524 for (int i
= 0; i
< v
->last_waveform
->level_count
; i
++)
526 if (freq64
<= v
->last_waveform
->levels
[i
].max_rate
)
529 v
->gen
.sample_data
= v
->last_waveform
->levels
[i
].data
;
534 v
->last_level_min_rate
= v
->last_waveform
->levels
[i
].max_rate
;
540 gboolean play_loop
= v
->layer
->loop_end
&& (v
->loop_mode
== slm_loop_continuous
|| playing_sustain_loop
) && v
->layer
->on_cc_number
== -1;
541 loop_start
= play_loop
? v
->layer
->loop_start
: (v
->layer
->count
? 0 : (uint32_t)-1);
542 loop_end
= play_loop
? v
->layer
->loop_end
: v
->gen
.cur_sample_end
;
546 v
->gen
.sample_data
= v
->gen
.loop_count
? v
->current_pipe
->data
: v
->last_waveform
->data
;
547 v
->gen
.streaming_buffer
= v
->current_pipe
->data
;
549 v
->gen
.prefetch_only_loop
= (loop_end
< v
->last_waveform
->preloaded_frames
);
550 v
->gen
.loop_overlap
= 0;
551 if (v
->gen
.prefetch_only_loop
)
553 assert(!v
->gen
.in_streaming_buffer
); // XXXKF this won't hold true when loops are edited while sound is being played (but that's not supported yet anyway)
554 v
->gen
.loop_start
= loop_start
;
555 v
->gen
.loop_end
= loop_end
;
556 v
->gen
.streaming_buffer_frames
= 0;
560 v
->gen
.loop_start
= 0;
561 v
->gen
.loop_end
= v
->last_waveform
->preloaded_frames
;
562 v
->gen
.streaming_buffer_frames
= v
->current_pipe
->buffer_loop_end
;
567 v
->gen
.loop_count
= v
->layer
->count
;
568 v
->gen
.loop_start
= loop_start
;
569 v
->gen
.loop_end
= loop_end
;
573 // Use pre-calculated join
574 v
->gen
.scratch
= loop_start
== (uint32_t)-1 ? v
->layer
->scratch_end
: v
->layer
->scratch_loop
;
578 // The standard waveforms have extra MAX_INTERPOLATION_ORDER of samples from the loop start added past loop_end,
579 // to avoid wasting time generating the joins in all the practical cases. The slow path covers custom loops
580 // (i.e. partial loop or no loop) over bandlimited versions of the standard waveforms, and those are probably
581 // not very useful anyway, as changing the loop removes the guarantee of the waveform being bandlimited and
582 // may cause looping artifacts or introduce DC offset (e.g. if only a positive part of a sine wave is looped).
583 if (loop_start
== 0 && loop_end
== l
->eff_waveform
->info
.frames
)
584 v
->gen
.scratch
= v
->gen
.sample_data
+ l
->eff_waveform
->info
.frames
- MAX_INTERPOLATION_ORDER
;
587 // Generate the join for the current wave level
588 // XXXKF this could be optimised further, by checking if waveform and loops are the same as the last
589 // time. However, this code is not likely to be used... ever, so optimising it is not the priority.
590 int shift
= l
->eff_waveform
->info
.channels
== 2 ? 1 : 0;
591 uint32_t halfscratch
= MAX_INTERPOLATION_ORDER
<< shift
;
593 v
->gen
.scratch
= v
->gen
.scratch_bandlimited
;
594 memcpy(&v
->gen
.scratch_bandlimited
[0], &v
->gen
.sample_data
[(loop_end
- MAX_INTERPOLATION_ORDER
) << shift
], halfscratch
* sizeof(int16_t) );
595 if (loop_start
!= (uint32_t)-1)
596 memcpy(v
->gen
.scratch_bandlimited
+ halfscratch
, &v
->gen
.sample_data
[loop_start
<< shift
], halfscratch
* sizeof(int16_t));
598 memset(v
->gen
.scratch_bandlimited
+ halfscratch
, 0, halfscratch
* sizeof(int16_t));
604 v
->gen
.bigdelta
= freq64
;
605 float gain
= modsrcs
[smsrc_ampenv
- smsrc_pernote_offset
] * l
->volume_linearized
* v
->gain_fromvel
* sampler_channel_addcc(c
, 7) * sampler_channel_addcc(c
, 11) / (maxv
* maxv
);
606 if (moddests
[smdest_gain
] != 0.f
)
607 gain
*= dB2gain(moddests
[smdest_gain
]);
608 // http://drealm.info/sfz/plj-sfz.xhtml#amp "The overall gain must remain in the range -144 to 6 decibels."
611 float pan
= (l
->pan
+ 100.f
) * (1.f
/ 200.f
) + (sampler_channel_addcc(c
, 10) * 1.f
/ maxv
- 0.5f
) * 2.f
;
616 v
->gen
.lgain
= gain
* (1.f
- pan
) / 32768.f
;
617 v
->gen
.rgain
= gain
* pan
/ 32768.f
;
618 struct cbox_biquadf_coeffs
*second_filter
= &v
->filter_coeffs
;
619 gboolean is4p
= sampler_layer_data_is_4pole(v
->layer
);
620 if (l
->cutoff
!= -1.f
)
622 float logcutoff
= l
->logcutoff
+ moddests
[smdest_cutoff
];
625 if (logcutoff
> 12798)
627 //float resonance = v->resonance*pow(32.0,c->cc[71]/maxv);
628 float resonance
= l
->resonance_linearized
* dB2gain((is4p
? 0.5 : 1) * moddests
[smdest_resonance
]);
629 if (resonance
< 0.7f
)
631 if (resonance
> 32.f
)
636 cbox_biquadf_set_lp_rbj_lookup(&v
->filter_coeffs
, &m
->sincos
[(int)logcutoff
], resonance
* resonance
);
637 cbox_biquadf_set_1plp_lookup(&v
->filter_coeffs_extra
, &m
->sincos
[(int)logcutoff
], 1);
638 second_filter
= &v
->filter_coeffs_extra
;
643 cbox_biquadf_set_lp_rbj_lookup(&v
->filter_coeffs
, &m
->sincos
[(int)logcutoff
], resonance
);
647 cbox_biquadf_set_hp_rbj_lookup(&v
->filter_coeffs
, &m
->sincos
[(int)logcutoff
], resonance
);
651 cbox_biquadf_set_bp_rbj_lookup(&v
->filter_coeffs
, &m
->sincos
[(int)logcutoff
], resonance
);
656 cbox_biquadf_set_1plp_lookup(&v
->filter_coeffs
, &m
->sincos
[(int)logcutoff
], l
->fil_type
!= sft_lp6
);
661 cbox_biquadf_set_1php_lookup(&v
->filter_coeffs
, &m
->sincos
[(int)logcutoff
], l
->fil_type
!= sft_hp6
);
668 float left
[CBOX_BLOCK_SIZE
], right
[CBOX_BLOCK_SIZE
];
670 uint32_t samples
= 0;
675 uint32_t limit
= cbox_prefetch_pipe_get_remaining(v
->current_pipe
) - 4;
677 v
->gen
.mode
= spt_inactive
;
680 samples
= sampler_gen_sample_playback(&v
->gen
, left
, right
, limit
);
681 cbox_prefetch_pipe_consumed(v
->current_pipe
, v
->gen
.consumed
);
687 samples
= sampler_gen_sample_playback(&v
->gen
, left
, right
, (uint32_t)-1);
690 for (int i
= samples
; i
< CBOX_BLOCK_SIZE
; i
++)
691 left
[i
] = right
[i
] = 0.f
;
694 cbox_biquadf_process(&v
->filter_left
, &v
->filter_coeffs
, left
);
696 cbox_biquadf_process(&v
->filter_left2
, second_filter
, left
);
697 cbox_biquadf_process(&v
->filter_right
, &v
->filter_coeffs
, right
);
699 cbox_biquadf_process(&v
->filter_right2
, second_filter
, right
);
701 for (int eq
= 0; eq
< 3; eq
++)
703 if (l
->eq_bitmask
& (1 << eq
))
705 cbox_biquadf_process(&v
->eq_left
[eq
], &v
->eq_coeffs
[eq
], left
);
706 cbox_biquadf_process(&v
->eq_right
[eq
], &v
->eq_coeffs
[eq
], right
);
710 mix_block_into(outputs
, v
->output_pair_no
* 2, left
, right
);
711 if ((v
->send1bus
> 0 && v
->send1gain
!= 0) || (v
->send2bus
> 0 && v
->send2gain
!= 0))
713 if (v
->send1bus
> 0 && v
->send1gain
!= 0)
715 int oofs
= m
->module
.aux_offset
+ (v
->send1bus
- 1) * 2;
716 mix_block_into_with_gain(outputs
, oofs
, left
, right
, v
->send1gain
);
718 if (v
->send2bus
> 0 && v
->send2gain
!= 0)
720 int oofs
= m
->module
.aux_offset
+ (v
->send2bus
- 1) * 2;
721 mix_block_into_with_gain(outputs
, oofs
, left
, right
, v
->send2gain
);
724 if (v
->gen
.mode
== spt_inactive
)
725 sampler_voice_inactivate(v
, FALSE
);